2013-10-21  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.24.0 Released.

2013-10-18  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.24.0-rc2 Released.

	* Properly copy/remove the device state cache flag over a masquerade.

	  In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that
	  tells the devstate system to not cache states for non-real devices.
	  However, when optimizing away channels (ast_do_masquerade), that flag
	  wasn't copied.

	  In my case, using Local devices as queue members created a situation
	  where the endpoint was considered in use, but the state change of the
	  device being available again was ignored (not cached). The endpoint
	  channel was optimized into the (previously) Local channel, but kept
	  the do-not-cache flag. The end result being that the queue member
	  apparently stayed in use forever.

2013-09-19  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.24.0-rc1 Released.

2013-09-18 19:54 +0000 [r399402]  Kinsey Moore <kmoore@digium.com>

	* main/abstract_jb.c: Fix jitter buffer log file creation This
	  adjusts '/'-to-'#' replacement to replace all instances of '/'
	  instead of just the first to ensure that the jitter buffer log
	  file gets the correct name as per Richard Kenner's suggestion.
	  (closes issue ASTERISK-21036) Reported by: Richard Kenner

2013-09-18 17:15 +0000 [r399351]  Matthew Jordan <mjordan@digium.com>

	* build_tools/prep_tarball: Update prep_tarball with new
	  documentation files on the Asterisk wiki This will now pull both
	  a command reference for the version being prepared, as well as an
	  Admin Guide that applies to all versions of Asterisk. (issue
	  ASTERISK-22439) Reported by: Olle Johansson

2013-09-18 01:32 +0000 [r399304]  Michael L. Young <elgueromexicano@gmail.com>

	* main/features.c: Fix Segfault When Syntax Of A Line Under
	  [applicationmap] Is Invalid When processing the lines under the
	  [applicationmap] context in features.conf, a segfault occurs from
	  attempting to process a line with an invalid syntax (basically
	  missing most of the arguments). Example: [applicationmap]
	  automon=*6 * This patch moves the checking for empty arguments to
	  before they are accessed. * Also, checked the "todo" comment and
	  removed it. Some applications do not require arguments. (closes
	  issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
	  Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2803

2013-09-16 16:37 +0000 [r399158]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c: chan_iax2: Fix saving the wrong expiry time
	  in astdb. When a new IAX2 client registers, the astdb database is
	  updated with the value of minregexpire defined in iax.conf
	  instead of using the expiry time that is provided by the client.
	  The provided expiry time of the client is updated after inserting
	  the astdb entry. As a consequence, restarting or reloading
	  asterisk creates clients whose registration may expire before
	  they reregister. The clients are therefore unavailable after
	  minregexpire seconds until they reregister. * Move updating of
	  the expiry time to before inserting into the astdb. (closes issue
	  ASTERISK-22504) Reported by: Stefan Wachtler Patches:
	  chan_iax2.c.patch (license #6533) patch uploaded by Stefan
	  Wachtler

2013-09-13 20:47 +0000 [r399098]  David M. Lee <dlee@digium.com>

	* main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
	  defined. If MALLOC_DEBUG is enabled, then the debug destructor
	  for the container is used, which would erroneously write to
	  /tmp/refs. This patch only uses the debug destructor if ref_debug
	  is used. (closes issue ASTERISK-22536)

2013-09-13 13:31 +0000 [r399033]  Kinsey Moore <kmoore@digium.com>

	* apps/app_meetme.c: Fix several crashes in MeetMeAdmin This change
	  ensures that MeetMeAdmin commands requiring a user actually get a
	  user and fixes another issue where an extra dereference could
	  occur for a last-entered user being ejected if a user identifier
	  was also provided. (closes issue ASTERISK-21907) Reported by:
	  Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/

2013-09-12 20:09 +0000 [r398937-398977]  Jonathan Rose <jrose@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Revert
	  r398835 due to failing tests involving originate (issue
	  ASTERISK-22424) Reported by: Jonathan Rose

	* res/res_musiconhold.c: res_musiconhold: Fix reference leaks
	  caused when reloading with REF_DEBUG set Due to a faulty function
	  for debugging reference decrementing, it was possible to reduce
	  the refcount on the wrong object if two moh classes of the same
	  name were in the moh class container. (closes issue
	  ASTERISK-22252) Reported by: Walter Doekes Patches:
	  18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
	  6182)

2013-09-12 00:00 +0000 [r398880-398884]  Rusty Newton <rnewton@digium.com>

	* apps/app_queue.c: 'queue add member' help text correction You are
	  adding dial strings to the queue, not channels. An aribitrary
	  string could be used, but you are typically referencing a
	  channel. Correcting the command help text. (issue ASTERISK-22263)
	  (closes issue ASTERISK-22263) Reported By: Rusty Newton

	* configs/chan_dahdi.conf.sample: Documentation fix -
	  waitfordialtone is not boolean, it's time in milliseconds
	  Changing text in chan_dahdi.conf sample to be accurate. (issue
	  ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
	  Malcolm Davenport

2013-09-11 19:39 +0000 [r398835]  Jonathan Rose <jrose@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
	  calls without prior SDP on 200 OK If we receive a 200 OK without
	  SDP, we will now check to see if the remote address has been
	  established for that channel's RTP session and if the to tag for
	  that channel has changed from the most recent to tag in a
	  response less than 200. If either a change has been made since
	  the last to-tag was received or the remote address is unset, then
	  we will drop the call. (closes issue ASTERISK-22424) Reported by:
	  Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/2827/diff/#index_header

2013-09-10 17:53 +0000 [r398757]  Richard Mudgett <rmudgett@digium.com>

	* main/xmldoc.c, main/cli.c, funcs/func_dialgroup.c, main/heap.c,
	  main/event.c, res/res_musiconhold.c, main/indications.c,
	  main/asterisk.c: Fix incorrect usages of ast_realloc(). There are
	  several locations in the code base where this is done: buf =
	  ast_realloc(buf, new_size); This is going to leak the original
	  buf contents if the realloc fails. Review:
	  https://reviewboard.asterisk.org/r/2832/

2013-09-10 17:47 +0000 [r398748-398752]  David M. Lee <dlee@digium.com>

	* utils/check_expr.c: Fixed utils directory breakage from r398748,
	  this time with extra hate.

	* utils/check_expr.c, utils/ael_main.c, utils/conf2ael.c: Fixed
	  utils directory breakage from r398648

2013-09-09 23:15 +0000 [r398703]  Richard Mudgett <rmudgett@digium.com>

	* main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
	  completely different from the freed magic number. Race conditions
	  between freeing a nul terminated string and ast_strdup()'ing it
	  are more likely to be detected if the fence and freed magic
	  numbers are completely different.

2013-09-09 19:56 +0000 [r398648]  David M. Lee <dlee@digium.com>

	* main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
	  DEBUG_THREADS when lock is acquired in __constructor__ This patch
	  fixes some long-standing bugs in debug threads that were
	  exacerbated with recent Optional API work in Asterisk 12. With
	  debug threads enabled, on some systems, there's a lock ordering
	  problem between our mutex and glibc's mutex protecting its module
	  list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
	  thread, the module list will be locked before acquiring our
	  mutex. In another thread, our mutex will be locked before locking
	  the module list (which happens in the depths of calling
	  backtrace()). This patch fixes this issue by moving backtrace()
	  calls outside of critical sections that have the mutex acquired.
	  The bigger change was to reentrancy tracking for
	  ast_cond_{timed,}wait, which wrongly assumed that waiting on the
	  mutex was equivalent to a single unlock (it actually suspends all
	  recursive locks on the mutex). (closes issue ASTERISK-22455)
	  Review: https://reviewboard.asterisk.org/r/2824/

2013-09-06 20:56 +0000 [r398523-398576]  Kinsey Moore <kmoore@digium.com>

	* res/res_jabber.c: Commit the remainder of r398523 This is a
	  missing part of the commit in revision 398523 that corrects the
	  name of a variable. (issue ASTERISK-22435)

	* res/res_jabber.c: Fix Jabber/XMPP distributed MWI The mailbox and
	  context are swapped on the receiving end for all users of Jabber
	  and XMPP distributed MWI in Asterisk 1.8 and all more recent
	  versions. This swaps those values to be correct when publishing
	  to the internal event system from Jabber/XMPP distributed MWI
	  state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested
	  by: Michael Keuter Patches:
	  asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
	  abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
	  uploaded by abelbeck

2013-09-05 19:00 +0000 [r398235-398456]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c: chan_iax2: Reduce indentation in
	  __attempt_transmit(). * Reduce indentation in
	  __attempt_transmit(). * Don't update the static last error time
	  variable every time in __schedule_action() and socket_read().

	* channels/chan_iax2.c: chan_iax2: Fix stray reference to worker
	  thread idle_list. * Fix stray reference to idle_list in
	  cleanup_thread_list(). This may be the reason for the note in
	  iax2_process_thread() about threads not being removed from the
	  task lists. * Move cleanup_thread_list(&idle_list) to after the
	  other lists are cleaned up.

	* channels/chan_iax2.c: chan_iax2: Fix bridgecallno deadlock
	  avoidance. * Fix bridgecallno deadlock avoidance. When doing
	  deadlock avoidance, you need to retest the status of values for
	  each loop to see if you still need the lock for bridgecallno. *
	  As a safety check, after acquiring the bridgecallno lock you
	  should check if iaxs[bridgecallno] is NULL just like the current
	  callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
	  to after processing any deferred frames to ensure that the
	  iostate is IDLE when it is placed back into the idle list.
	  defer_full_frame() tries to ensure iax2_process_thread() wakes up
	  to process the frame.

	* channels/iax2-parser.c: chan_iax2: Add missing control frame
	  names to debug frame decode output. (Part 2)

	* channels/iax2-parser.c: chan_iax2: Add missing control frame
	  names to debug frame decode output.

	* channels/chan_misdn.c: chan_misdn: Fix misdn debug output printed
	  with arbitrary verbose levels. Fix the misdn debug output to
	  remote consoles. chan_misdn uses ast_console_puts() which doesn't
	  know about verbose levels. Better to use ast_verbose() instead.
	  Without this patch the misdn debug messages are appended to the
	  verbose level which ever was set by the message sent to the
	  console before, i.e. any undefined level. (closes issue AST-1218)
	  Reported by: Guenther Kelleter Patches: misdnlog.patch (license
	  #6372) patch uploaded by Guenther Kelleter

2013-09-02 07:24 +0000 [r398167]  Walter Doekes <walter+asterisk@wjd.nu>

	* cel/cel_custom.c: Be a little more verbose when loading
	  cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/

2013-08-30 18:55 +0000 [r398021-398102]  Kevin Harwell <kharwell@digium.com>

	* channels/chan_sip.c, main/config.c, res/res_security_log.c: Fix
	  various memory leaks main/config.c - cleanup cache fie includes
	  res/res_security_log.c - unregister logger level
	  channesl/chan_sip.c - cleanup io context and notify_types (closes
	  issues ASTERISK-22378) Reported by: Corey Farrell Patches:
	  config_shutdown.patch uploaded by coreyfarrell (license 5909)
	  res_security_log.patch uploaded by coreyfarrell (license 5909)
	  chan_sip-1.8.patch uploaded by coreyfarrell (license 5909)

	* main/manager.c, res/res_agi.c: Memory leak fix
	  ast_xmldoc_printable returns an allocated block that must be
	  freed by the caller. Fixed manager.c and res_agi.c to stop
	  leaking these results. (closes issue ASTERISK-22395) Reported by:
	  Corey Farrell Patches: manager-leaks-1.8.patch uploaded by
	  coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
	  by coreyfarrell (license 5909)

	* main/features.c: Fix memory leak Fixed a features.c test that
	  leaked a reference to a parked call. This caused chancount to
	  never reach 0, so graceful shutdown stops. Also added an
	  unregister test. (closes issue ASTERISK-22413) Reported by: Corey
	  Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
	  coreyfarrell (license 5909)

2013-08-30 16:46 +0000 [r398018]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_substitution.c: test_substituition: Fix failed test
	  reporting to actually report failure. You cannot put the "Testing
	  <blah> pass/fail" on a single line before actually performing the
	  test. Now any additional failure information is logged before the
	  test pass/fail announcement. * Added an additional CDR(answer,u)
	  test.

2013-08-27 17:55 +0000 [r397710-397756]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
	  SDP If the SIP channel driver processes an invalid SDP that
	  defines media descriptions before connection information, it may
	  attempt to reference the socket address information even though
	  that information has not yet been set. This will cause a crash.
	  This patch adds checks when handling the various media
	  descriptions that ensures the media descriptions are handled only
	  if we have connection information suitable for that media. Thanks
	  to Walter Doekes, OSSO B.V., for reporting, testing, and
	  providing the solution to this problem. (closes issue
	  ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
	  issueA22007_sdp_without_c_death.patch uploaded by wdoekes
	  (License 5674)

	* channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK on
	  dialog that has no channel A remote exploitable crash
	  vulnerability exists in the SIP channel driver if an ACK with SDP
	  is received after the channel has been terminated. The handling
	  code incorrectly assumed that the channel would always be
	  present. This patch adds a check such that the SDP will only be
	  parsed and applied if Asterisk has a channel present that is
	  associated with the dialog. Note that the patch being applied was
	  modified only slightly from the patch provided by Walter Doekes
	  of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
	  Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
	  issueA21064_fix.patch uploaded by wdoekes (License 5674)

2013-08-23 15:34 +0000 [r397525]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/astmm.h, main/logger.c, main/utils.c,
	  include/asterisk/lock.h, main/astmm.c, channels/sig_pri.c,
	  main/astobj2.c, include/asterisk/logger.h, main/lock.c,
	  include/asterisk/utils.h: Fix memory corruption when trying to
	  get "core show locks". Review
	  https://reviewboard.asterisk.org/r/2580/ tried to fix the
	  mismatch in memory pools but had a math error determining the
	  buffer size and didn't address other similar memory pool
	  mismatches. * Effectively reverted the previous patch to go in
	  the same direction as trunk for the returned memory pool of
	  ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols()
	  when BETTER_BACKTRACES is defined. * Fixed some formatting in
	  ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated
	  by libpri when MALLOC_DEBUG is enabled. * Fixed
	  __dump_backtrace() freeing memory from ast_bt_get_symbols() when
	  MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of
	  compile issues with the utils directory. (closes issue
	  ASTERISK-22221) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2778/

2013-08-22 08:19 +0000 [r397377]  Walter Doekes <walter+asterisk@wjd.nu>

	* default.exports, main/asterisk.exports.in: Add _IO_stdin_used in
	  version-script to fix SIGBUSes on Sparc. The
	  --version-script,asterisk.exports linker flag (and the module
	  exports) didn't provide _IO_stdin_used in the list of exported
	  symbols. That causes some kind of libc compatibility mode to kick
	  in, where stdio file structures (stdout/stderr) land somewhere
	  else. In the case of the Sparc, they landed on misaligned memory.
	  This became apparent first after r376428 (Reorder startup
	  sequence) when a lot of ast_log's were replaced with fprintf's.
	  Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
	  architectures, the Sparc is very picky about memory alignment.)
	  (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
	  Kister Review: https://reviewboard.asterisk.org/r/2760/

2013-08-21 17:00 +0000 [r397308]  David M. Lee <dlee@digium.com>

	* main/http.c: Complete http_shutdown. This patch frees up some
	  resources allocated in http.c. * tcp listeners stopped * tls
	  settings freed * uri redirects freed * unregister internal http.c
	  uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
	  Patches: http.patch uploaded by Corey Farrell (license 5909)

2013-08-21 14:56 +0000 [r397256]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/frame.h: Set 14400 as the default max bit rate
	  if T38MaxBitRate is not specified If an endpoint fails to include
	  the T38MaxBitRate attribute during negotiation, Asterisk will
	  negotiate a bit rate of 2400 instead of the ITU recommended bit
	  rate of 14400. This patch fixes this by making AST_T38_RATE_14400
	  the 'default' value of the enum by assigning it a value of 0,
	  such that if an endpoint fails to include the attribute, the
	  default will be 14400. Note that Walter Doekes included the nice
	  comment in frame.h about why we are purposefully assigning
	  AST_T38_RATE_14400 a value of 0. (closes issue ASTERISK-22275)
	  Reported by: Andreas Steinmetz patches: fax-fix.patch uploaded by
	  anstein (License 6523)

2013-08-21 02:09 +0000 [r397204]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix Not Storing Current Incoming Recv
	  Address In 1.8, r384779 introduced a regression by retrieving an
	  old dialog and keeping the old recv address since recv was
	  already set. This has caused a problem when a proxy is involved
	  since responses to incoming requests from the proxy server, after
	  an outbound call is established, are never sent to the correct
	  recv address. In 11, r382322 introduced this regression. The fix
	  is to revert that change and always store the recv address on
	  incoming requests. Thank you Walter Doekes for helping to point
	  out this error and Mark Michelson for your input/review of the
	  fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
	  Tested by: Alex Zarubin, Karsten Wemheuer Patches:
	  asterisk-22071-store-recvd-address.diff by Michael L. Young
	  (license 5026)

2013-08-20 17:40 +0000 [r397112-397156]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove REF_DEBUG definition.

	* channels/sip/dialplan_functions.c, channels/chan_sip.c: Fix
	  refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
	  the list of pvts. (closes issue ASTERISK-22248) reported by Corey
	  Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
	  (license #5909)

2013-08-20 15:26 +0000 [r397033-397106]  Kinsey Moore <kmoore@digium.com>

	* main/threadstorage.c, main/astfd.c: Unregister CLI commands on
	  exit This patch ensures that CLI commands enabled by
	  DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
	  exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
	  Tested by: Corey Farrell Patches: debug_cli_unregister.patch
	  uploaded by Corey Farrell

	* main/xmldoc.c: Fix xmldoc memory leak This fixes a
	  single-attribute memory leak that was occurring when the
	  "required" attribute was not true. (closes issue ASTERISK-22249)
	  Reported by: Corey Farrell Tested by: Corey Farrell Patches:
	  xmldoc-free_attr_required.patch uploaded by Corey Farrell

	* main/cel.c: Protect CEL from an invalid config on reload This
	  patch fixes CEL to properly handle an invalid config on reload.
	  (closes issue ASTERISK-22259) Reported by: Corey Farrell Tested
	  by: Corey Farrell Patches: cel-config.patch uploaded by Corey
	  Farrell

2013-08-20 11:46 +0000 [r396994]  Walter Doekes <walter+asterisk@wjd.nu>

	* configs/h323.conf.sample, configs/sip.conf.sample: Add
	  "autoframing" option to sip.conf.sample and h323.conf.sample. The
	  autoframing option was added to chan_sip.c in r43243 (mogorman,
	  2006-09-19 01:32:57), but never made its way into the sample
	  configs. Review: https://reviewboard.asterisk.org/r/2768/

2013-08-20 01:17 +0000 [r396958]  Matthew Jordan <mjordan@digium.com>

	* main/data.c: Fix invalid access to disposed memory in main/data
	  unit test It is not safe to iterate over a macro'd list of ao2
	  objects, deref them such that the item's destructor is called,
	  and leave them in the list. The list macro to iterate over items
	  requires the item to be a valid allocated object in order to
	  proceed to the next item; with MALLOC_DEBUG on the corruption of
	  the linked list is caught in the crash. This patch fixes the
	  invalid access to free'd memory by removing the ao2 item from the
	  list before de-refing it. Note that this is a backport of r396915
	  from Asterisk trunk.

2013-08-15 16:21 +0000 [r396745]  Kinsey Moore <kmoore@digium.com>

	* main/cli.c, main/asterisk.c: Remove leading spaces from the CLI
	  command before parsing If you've mistakenly put a space before
	  typing in a command, the leading space will be included as part
	  of the command, and the command parser will not find the
	  corresponding command. This patch rectifies that situation by
	  stripping the leading spaces on commands. Review:
	  https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
	  Lesher

2013-08-14 19:05 +0000 [r396619-396656]  Joshua Colp <jcolp@digium.com>

	* tests/test_hashtab_thrash.c: Tweak comment for why usleep is
	  used.

	* tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
	  allow the critical threads to execute. Depending on certain
	  conditions it was possible for the hashtab counting thread to
	  starve other threads, preventing them from executing in the
	  expected fashion. This change adds a sleep to allow the others to
	  do what they need to do. While this doesn't thrash the hashtab as
	  much as previously, it at least works. (closes issue
	  ASTERISK-22276) Reported by: Matt Jordan

2013-08-13 18:44 +0000 [r396579-396582]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: chan_sip: Convert 'just did sched_add
	  waitid...' from warning to debug message. Patches:
	  reviewboard-2377.patch uploaded by Paul Belanger Review:
	  https://reviewboard.asterisk.org/r/2377/

	* channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
	  rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
	  by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/

2013-08-08 20:14 +0000 [r396427]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/logger.c, main/utils.c, main/astobj2.c,
	  include/asterisk/logger.h: Consistent memory allocation by
	  ast_bt_get_symbols. Always use ast_alloc/ast_free. This is
	  handled differently in trunk (r391012). Review:
	  https://reviewboard.asterisk.org/r/2580/

2013-08-06 08:14 +0000 [r396279]  Walter Doekes <walter+asterisk@wjd.nu>

	* pbx/pbx_dundi.c, utils/extconf.c, apps/app_stack.c,
	  apps/app_playback.c, funcs/func_global.c, main/cdr.c,
	  pbx/pbx_loopback.c, main/pbx.c, funcs/func_strings.c: Check
	  result of ast_var_assign() calls for memory allocation failure.
	  We try to keep the system running even when all available memory
	  is spent. Review: https://reviewboard.asterisk.org/r/2734/

2013-08-05 20:17 +0000 [r396196-396240]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix Registration Failure When A Peer And TLS
	  Are Used If a peer is used in a register line and TLS is defined
	  as the transport, the registration fails since the transport on
	  the dialog is never set properly resulting in UDP being used
	  instead of TLS. This patch sets the dialog's transport based on
	  the transport that was defined in the register line. If the
	  register line does not specify a transport, the parsing function
	  for the register line always defaults back to UDP. (closes issue
	  ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey
	  Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael
	  L. Young (license 5026)

	* channels/chan_sip.c: Restore Extra Line Break Between Peers When
	  Running AMI Action SIPPeers The commit (r387133) for fixing
	  ASTERISK-21466 accidentally removed an extra line break between
	  the peers returned by the AMI action SIPPeers. This results in
	  some parsers breaking because they expect this extra line break.
	  This patch restores that extra line break. (closes issue
	  ASTERISK-22239) Reported by: Jacek Konieczny Tested by: Jacek
	  Konieczny, Michael L. Young Patches:
	  asterisk-ami_sippeers_separator.patch by Jacek Konieczny (license
	  6298)

	* UPGRADE.txt: Adding a note to UPGRADE.txt about a change made to
	  res_agi in order to indicate when streaming an audio file fails
	  like it is done in other parts of the code to indicate an error.
	  Note was requested by Paul Belanger:
	  http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
	  (related to issue ASTERISK-21903)

2013-07-22 13:49 +0000 [r394886-395032]  Matthew Jordan <mjordan@digium.com>

	* main/asterisk.c: Update copyright year to 2013 in asterisk.c;
	  some whitespace fixes (closes issue ASTERISK-22179) Reported by:
	  Malcolm Davenport

	* funcs/func_channel.c: Clean up documentation This patch cleans up
	  documentation in func_channel for the following items: *
	  rtpsource * secure_signaling * secure_media (closes issue
	  ASTERISK-20969) Reported by: snuffy patches:
	  func_chan-update.diff uploaded by snuffy (License 5024)

	* configs/indications.conf.sample: Provide proper ring tone in
	  indications.conf for Malaysia The ring tone provided in the
	  sample indications.conf was incorrect. This patch modifies the
	  sample ring tone to be what it should: ring =
	  425/400,0/200,425/400,0/2000 This brings it in line with the tone
	  definition in DAHDI 2.7.0. (zonedata.c) (closes issue
	  ASTERISK-21997) Reported by: Filip Jenicek patches:
	  malaysia_ring.patch uploaded by phill (License 6277)

	* main/http.c: Tolerate presence of RFC2965 Cookie2 header by
	  ignoring it This patch modifies parsing of cookies in Asterisk's
	  http server by doing an explicit comparison of the "Cookie"
	  header instead of looking at the first 6 characters to determine
	  if the header is a cookie header. This avoids parsing "Cookie2"
	  headers and overwriting the previously parsed "Cookie" header.
	  Note that we probably should be appending the cookies in each
	  "Cookie" header to the parsed results; however, while clients can
	  send multiple cookie headers they never really do. While this
	  patch doesn't improve Asterisk's behavior in that regard, it
	  shouldn't make it any worse either. Note that the solution in
	  this patch was pointed out on the issue by the issue reporter,
	  Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
	  Stuart Henderson Tested by: mjordan, Stuart Henderson

	* contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
	  realtime scripts with schema for queue_log table This patch
	  updates the realtime SQL scripts with an entry that will create
	  the queue_log table. This brings the PostgreSQL scripts inline
	  with the MySQL scripts, with respect to what tables they will
	  create. (closes issue ASTERISK-21021) Reported by: Eugene
	  patches: queue_log.sql uploaded by varnav (license 6360)

	* configs/iax.conf.sample: Document connectedline parameter for
	  chan_iax2 The connectedline parameter for a chan_iax2 peer was
	  undocumented. This patch documents the options in the sample
	  configuration file. (closes issue ASTERISK-21953) Reported by:
	  Birger "WIMPy" Harzenetter

2013-07-18 12:51 +0000 [r394640]  Michael L. Young <elgueromexicano@gmail.com>

	* res/res_agi.c: Properly indicate failure to open an audio stream
	  in res_agi If there is an error streaming an audio file, the
	  current return status makes it difficult for an AGI script to
	  determine that there was an error with the audio file. This
	  patches changes the result to return -1 and the function returns
	  RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
	  parts of res_agi, this would appear to be the proper way to
	  handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
	  Wainer Tested by: Ariel Wainer Patches:
	  asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/2625/

2013-07-14 01:53 +0000 [r394302]  Matthew Jordan <mjordan@digium.com>

	* funcs/func_strings.c: Clarify documentation for function PASSTHRU
	  It is not apparent to the average user that the PASSTHRU function
	  should not be passed as ${PASSTHRU(string)} but just as
	  PASSTHRU(string) to functions which take a variable name and not
	  its contents. This patch clarifies the behavior in the
	  documentation and provides an example. (closes issue
	  ASTERISK-21717) Reported by: Richard Miller patches:
	  func_strings.diff uploaded by Richard Miller (license 5685)

2013-07-11 16:25 +0000 [r394106]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: Fix a longstanding issue with MFC-R2
	  configuration that prevented users from mixing different variants
	  or general MFC-R2 settings within the same E1 line. Most users do
	  not have a problem with this since MFC-R2 lines are usually
	  fractional E1s, or the whole E1 has the same country variant and
	  R2 settings. In Venezuela however is common to have inbound
	  MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
	  properly parses the chan_dahdi.conf file to generate a new openr2
	  context every time a new channel => section is found and the
	  configuration was changed. (closes issue ASTERISK-21117) Reported
	  by: Rafael Angulo Related Elastix issue:
	  http://bugs.elastix.org/view.php?id=1612

2013-07-10 01:41 +0000 [r393928]  Russell Bryant <russell@russellbryant.com>

	* configs/sla.conf.sample, include/asterisk/utils.h,
	  apps/app_meetme.c: astobj2-ify the SLA code The SLA code within
	  app_meetme was written before asotbj2 had been merged into
	  Asterisk. Worse, support for reloads did not exist at first and
	  was added later as a bolt-on feature. I knew at the time that
	  reloading was not safe at all while SLA was in use, so the reload
	  would be queued up to execute when the system was idle.
	  Unfortunately, this approach was still prone to errors beyond the
	  fact that this was the only place in Asterisk where configuration
	  was not reloaded instantly when requested. This patch converts
	  various SLA objects to be reference counted objects using
	  astobj2. This allows reloads to be processed while the system is
	  in use. The code ensures that the objects will not disappear
	  while one of the other threads is using them. However, they will
	  be immediately removed from the global trunk and station
	  containers so no new calls will use them if removed from
	  configuration. Review: https://reviewboard.asterisk.org/r/2581/

2013-07-03 23:27 +0000 [r393627]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: chan_dahdi: Fix segfault reloading
	  chan_dahdi when round robin is used. * Clear round_robin[] in
	  dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
	  Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
	  patch uploaded by rmudgett

2013-06-14 16:14 +0000 [r391778]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c: app_mixmonitor: Fix crashes caused by
	  unloading app_mixmonitor Unloading app_mixmonitor while active
	  mixmonitors were running would cause a segfault. This patch fixes
	  that by making it impossible to unload app_mixmonitor while
	  mixmonitors are active. Review:
	  https://reviewboard.asterisk.org/r/2624/

2013-06-12 02:19 +0000 [r391489]  Matthew Jordan <mjordan@digium.com>

	* main/loader.c: Fix memory leak while loading priority modules
	  When we load a module with the LOAD_PRIORITY flag, we remove its
	  entry from the load order list. Unfortunately, we don't free the
	  memory associated with entry in the list. This patch corrects
	  that and properly frees the memory for the module in the list.

2013-06-11 08:03 +0000 [r391333]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_iax2.c: IAX2: Transfer Reject: Lock bridgecallno
	  before touching it, refactor 1). When touching the bridgecallno,
	  we need to lock it. 2). Remove magic number '0' and replace with
	  TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
	  indentation. Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2613/

2013-07-15  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.23.0 Released.

2013-07-12  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.23.0-rc2 Released.

	* Properly lock and safely handle a transfer failure in IAX2

	  When touching the bridgecallno, we need to lock it - otherwise a
	  race condition can occur. This patch does the proper locking
	  of the bridgecallno before modifying its state.

2013-06-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.23.0-rc1 Released.

2013-06-10 14:15 +0000 [r391215]  Matthew Jordan <mjordan@digium.com>

	* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Add
	  announce-to-first-user option for app_queue In r386792, the
	  ability to play prompts to the first caller in a call queue was
	  added. While this is arguably a bug fix for those who expect the
	  first caller to continue receiving prompts while the agent is
	  dialed, it has the side effect of preventing the first caller
	  from hearing the agent immediately upon bridging. This may not be
	  a problem for those who really want this option, but for those
	  who didn't care whether or not the first caller in queue heard
	  their position, it was an issue. This patch disables the ability
	  for the first caller in the queue to hear prompts and adds a new
	  option, announce-to-first-user, to queues.conf. Those who the
	  behavior can enable it by setting this value to True. Note that
	  if we ever implement the ability to have the prompts be stopped
	  upon bridging, this option can be removed. (closes issue
	  ASTERISK-21782) Reported by: Remi Quezada

2013-06-10 09:30 +0000 [r391062-391143]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
	  unlock bridgecallno

	* channels/chan_iax2.c: fix bad edit after conflict resolution

	* channels/chan_iax2.c: IAX2: refactor nativebridge transfer remove
	  triple checking of iaxs[fr->callno]->transferring reduce
	  indentation. Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2602/

	* channels/chan_iax2.c: IAX2: fix race condition with nativebridge
	  transfers. 1). When touching the bridgecallno, we need to lock
	  it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes
	  the lock on the pvt is already held, when iax2_destroy_helper()
	  is called. Thus we need to lock the bridgecallno pvt before we
	  call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When
	  evaluating the state of 'callno->transferring' of the current
	  leg, we can't change it to READY unless the bridgecallno is
	  locked. Why, if we are interrupted by the other call leg before
	  'transferring = TRANSFER_RELEASED', the interrupt will find that
	  it is READY and that the bridgecallno is also READY so Releases
	  the legs. (closes issue ASTERISK-21409) Reported by: alecdavis
	  Tested by: alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2594/

2013-05-31 08:10 +0000 [r390181]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: reject call attempts when gatekeeper is
	  configured but not registered (closes issue ASTERISK-21800)
	  Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
	  Tested by: Dmitry Melekhov

2013-05-29 20:10 +0000 [r390044]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix segfault when dealing with chan_agent
	  channels. Check the returned bridged pointer for NULL to avoid a
	  crash. It looks like chan_agent is returning a NULL pointer when
	  it probably should be returning a pointer to the channel the
	  Agent channel is pretending to be. (closes issue ASTERISK-21793)
	  Reported by: Rodrigo P. Telles Patches:
	  jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Rodrigo P. Telles

2013-05-28 17:35 +0000 [r389895]  Jonathan Rose <jrose@digium.com>

	* main/slinfactory.c: Fix a memory copying bug in slinfactory which
	  was causing mixmonitor issues. Reported by: Michael Walton Tested
	  by: Jonathan Rose Patches: slinfactory.c.ASTERISK-21799.patch
	  uploaded by Michael Walton (license 6502) (closes issue
	  ASTERISK-21799)

2013-05-24 11:42 +0000 [r389676]  Matthew Jordan <mjordan@digium.com>

	* main/logger.c: Print all logger messages on shutdown When
	  Asterisk shuts down and shuts down the loggin gsubsystem, any
	  messages currently in flight will not get logged. This patch
	  prevents the loop writing messages from breaking out prematurely,
	  such that all of the messages are logged. (closes issue
	  ASTERISK-21716) Reported by: Corey Farrell patches:
	  logger-process-all-messages.patch uploaded by Corey Farrell
	  (license 5909)

2013-05-20 17:43 +0000 [r389244]  Jason Parker <jparker@digium.com>

	* /: Add doxygen.log to svn:ignore property.

2013-05-15 15:54 +0000 [r388838]  kharwell <kharwell@localhost>:

	* main/lock.c: Fix for segfault in __ast_rwlock_destroy with
	  DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
	  causes a segfault while trying to access a possible NULL t->track
	  object. A NULL check has been added before trying to access the
	  memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
	  Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
	  uploaded by Corey Farrell (license 5909)

2013-05-15 12:37 +0000 [r388768]  Kinsey Moore <kmoore@digium.com>

	* res/res_srtp.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Use srtp_shutdown when available This allows the
	  SRTP library to be shut down properly when the functionality is
	  offered by libsrtp. Review:
	  https://reviewboard.asterisk.org/r/2538/ (closes issue
	  ASTERISK-21719)

2013-05-13 20:34 +0000 [r388596]  Kinsey Moore <kmoore@digium.com>

	* res/res_srtp.c: Revert r388529 for now Adding the cleanup
	  function needs some deeper thought since it apparently doesn't
	  exist for all variants of libsrtp.

2013-05-13 18:16 +0000 [r388532]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c: pbx: Fix lack of cleanup on macrolock and
	  context_table (closes issue ASTERISK-21723) Reported by: Corey
	  Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
	  Farrell (license 5909)

2013-05-13 18:05 +0000 [r388529]  Kinsey Moore <kmoore@digium.com>

	* res/res_srtp.c: Close libsrtp properly Ensure that libsrtp is
	  shutdown properly when res_srtp is unloaded. (closes issue
	  ASTERISK-21719) Reported by: Corey Farrell Patches:
	  res_srtp-library-shutdown.patch uploaded by Corey Farrell

2013-05-13 14:24 +0000 [r388477]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c: Fix SendText AMI action to never return non-zero.
	  AMI actions must never return non-zero unless they intend to
	  close the AMI connection. (Which is almost never.) (closes issue
	  ASTERISK-21779) Reported by: Paul Goldbaum

2013-05-10 22:09 +0000 [r388423-388425]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
	  messsage. * Made isdn_msg_parser.c build a progress message with
	  the mandatory progress indicator IE. (The mISDNuser NT state
	  machine rejected sending the incomplete message.) Note: The
	  associated mISDN and mISDNuser patches respectively are viewable
	  here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
	  http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
	  issue AST-1153) Reported by: Guenther Kelleter Patches:
	  progress-chan_misdn.diff (license #6372) patch uploaded by
	  Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
	  uploaded by Guenther Kelleter progress-misdnuser.diff (license
	  #6372) mISDNuser patch uploaded by Guenther Kelleter

	* utils: Add version.c to list of ignored files in the utils
	  directory.

2013-05-10 20:28 +0000 [r388376]  Mark Michelson <mmichelson@digium.com>

	* pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added an
	  io context without removing it. This caused a memory leak when
	  the module was unloaded. (closes ASTERISK-21718) Reported by
	  Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
	  Corey Farrell (License #5909)

2013-05-09 03:58 +0000 [r388111]  Michael L. Young <elgueromexicano@gmail.com>

	* res/res_rtp_asterisk.c: Fix The Payload Being Set On CN Packets
	  And Do Not Set Marker Bit When we send out a CN packet (for
	  instance, in the case of using rtpkeepalives), we are not setting
	  the payload code properly. Also, we are setting the marker bit
	  when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN
	  is not defined by AST_FORMAT codes. Therefore, we should be using
	  ast_rtp_codecs_payload_code() rather than
	  ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
	  appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
	  * Remove the setting of the marker bit * Fix the debug message by
	  incrementing the seqno after the debug message is set in order to
	  display the correct seqno that was sent out (closes issue
	  ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
	  Katzmann, Michael L. Young Patches:
	  asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2500/

2013-05-08 07:17 +0000 [r387875]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing up
	  and fail to be sent out after retries fail RFC6665 4.2.2: ...
	  after a failed State NOTIFY transaction remove the subscription
	  The problem is that the State Notify requests rely on the 200OK
	  reponse for pacing control and to not confuse the notify
	  susbsystem. The issue is, the pendinginvite isn't cleared if a
	  response isn't received, thus further notify's are never sent.
	  The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
	  subscription after failure. (closes issue ASTERISK-21677)
	  Reported by: Dan Martens Tested by: Dan Martens, David Brillert,
	  alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2475/

2013-05-06 15:52 +0000 [r387688]  Russell Bryant <russell@russellbryant.com>

	* apps/app_meetme.c: Make SLA reload more paranoid. Reload support
	  was originally not included for SLA. It was added later, but in a
	  fairly non-traditional way. It basically sets a flag indicating
	  that a reload is pending, and then waits for a time where it
	  thinks everything SLA related is idle and unused, and *then*
	  executes the reload. It does this because the reload process is
	  destructive. It starts by throwing everything away and starting
	  over. There are a number of problems with this approach. One of
	  them is that the check to see if anything in use was incomplete.
	  This patch makes it more complete and thus less likely for a
	  crash to occur during reload processing. However, this approach
	  still has problems so some much more significant reworking of
	  this code will need to come in as a next step. Patch credit and
	  testing by CoreDial, LLC.

2013-05-02 17:11 +0000 [r387421]  Matthew Jordan <mjordan@digium.com>

	* utils/Makefile: Update utils Makefile to handle r387294 Alec's
	  patch that added the Asterisk version to 'core show locks'
	  angered the items in utils, as they exist somewhat outside of the
	  Asterisk build system. Some day, this Makefile should get nuked
	  from high orbit, but for now, include version.c in its list of
	  stuff to pile in.

2013-05-02 07:53 +0000 [r387294-387344]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
	  Session-Expires: Set timer to correctly expire at (~2/3) of the
	  interval when not the refresher RFC 4028 Section 10 if the side
	  not performing refreshes does not receive a session refresh
	  request before the session expiration, it SHOULD send a BYE to
	  terminate the session, slightly before the session expiration.
	  The minimum of 32 seconds and one third of the session interval
	  is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
	  Session-Expires interval, or if the remote device was the
	  refresher, asterisk would timeout at interval end. Now, when not
	  refresher, timeout as per RFC noted above. (closes issue
	  ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2488/

	* channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
	  response when it's a RE-INVITE when asterisk is the refresher.
	  RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
	  Session-Expires header field in a response, even if none were
	  present in the request." What changed After ASTERISK-20787,
	  inbound calls to asterisk with no Session-Expires in the INVITE
	  are now are offered a Session-Expires (1800 asterisk default) in
	  the response, with asterisk as the refresher. Symptom: After 900
	  seconds (asterisk default refresher period 1800), asterisk
	  RE-INVITEs the device, the device may respond with a much lower
	  Session-Expires (180 in our case) value that it is now using.
	  Asterisk ignores this response, as it's deemed both an INBOUND
	  CALL, and a RE-INVITE. After 180 seconds the device times out and
	  sends BYE (hangs up), asterisk is still working with the
	  refresher period of 1800 as it ignored the 'Session Expires: 180'
	  in the previous 200OK response. Fix: handle_response_invite()
	  when 200OK, remove check for outbound and reinvite. (closes issue
	  ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2463/

	* channels/chan_dahdi.c: chan_dahdi: fix lower bound check with -ve
	  integer conversion from a float Lower bound of a 16bit signed int
	  is -32768 not -32767 (closes issue ASTERISK-21744) Reported by:
	  alecdavis Tested by: alecdavis alecdavis (license 585)

	* main/utils.c: Add Asterisk Version to core show locks Assist with
	  reporting 'core show locks' when submitting bug reports. Example
	  below: =========================== == SVN-branch-1.8-... ==
	  Currently Held Locks =========================== (closes issue
	  ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585)

2013-05-01 21:15 +0000 [r387036-387213]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c: Clear the DTMF sending digit tracking on
	  off nominal paths In certain situations, when the RTP engine goes
	  to send a DTMF end digit it may be in a situation where the
	  remote address is no longer available, or the digit that was
	  supposed to be sent is invalid. In such cases, we need to clear
	  the RTP counters appropriately. Otherwise, when the RTP source is
	  set again, we'll continue to think that we're in the middle of
	  sending a DTMF digit, which can confuse the remote party
	  (signficantly). (closes issue ASTERISK-21522) Reported by: Corey
	  Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
	  Farrell (License 5909)

	* channels/chan_sip.c: Prevent crash in 'sip show peers' when the
	  number of peers on a system is large When you have lots of SIP
	  peers (according to the issue reporter, around 3500), the 'sip
	  show peers' CLI command or AMI action can crash due to a poorly
	  placed string duplication that occurs on the stack. This patch
	  refactors the command to not allocate the string on the stack,
	  and handles the formatting of a single peer in a separate
	  function call. (closes issue ASTERISK-21466) Reported by:
	  Guillaume Knispel patches:
	  fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
	  uploaded by gknispel (License 6492)

	* main/features.c: Fix CDR not being created during an externally
	  initiated blind transfer Way back when in the dark days of
	  Asterisk 1.8.9, blind transferring a call in a context that
	  included the 'h' extension would inadvertently execute the hangup
	  code logic on the transferred channel. This was a "bad thing".
	  The fix was to properly check for the softhangup flags on the
	  channel and only execute the 'h' extension logic (and, in later
	  versions, hangup handler logic) if the channel was well and truly
	  dead (Jim). Unfortunately, CDRs are fickle. Setting the
	  softhangup flag when we detected that the channel was leaving the
	  bridge (but not to die) caused some crucial snippet of CDR code,
	  lying in ambush in the middle of the bridging code, to not get
	  executed. This had the effect of blowing away one of the CDRs
	  that is typically created during a blind transfer. While we live
	  and die by the adage "don't touch CDRs in release branches", this
	  was our bad. The attached patch restores the CDR behavior, and
	  still manages to not run the 'h' extension during a blind
	  transfer (at least not when it's supposed to). Thanks to Steve
	  Davies for diagnosing this and providing a fix. Review:
	  https://reviewboard.asterisk.org/r/2476 (closes issue
	  ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
	  Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
	  one47 (License 5012)

2013-04-30 13:45 +0000 [r386929]  Sean Bright <sean@malleable.com>

	* include/asterisk/utils.h: Use the proper lower bound when doing
	  saturation arithmetic. 16 bit signed integers have a range of
	  [-32768, 32768). The existing code was using the interval
	  (-32768, 32768) instead. This patch fixes that. Review:
	  https://reviewboard.asterisk.org/r/2479/

2013-04-29 23:34 +0000 [r386877]  Rusty Newton <rnewton@digium.com>

	* sounds/Makefile: Modifying sounds/Makefile to pull down 1.4.24
	  core sounds 1.4.24 core sounds includes a full set of Italian
	  prompts for core sounds and a fix for the missing voicemail
	  prompts in the Russian language. (closes issue ASTERISK-19431)
	  (closes issue ASTERISK-19721)

2013-04-29 08:36 +0000 [r386792]  Olle Johansson <oej@edvina.net>

	* CHANGES, apps/app_queue.c: Play periodic prompst for first call
	  in a call queue Review: https://reviewboard.asterisk.org/r/2263/

2013-04-26 21:26 +0000 [r386641-386672]  Matthew Jordan <mjordan@digium.com>

	* main/config.c: Clean up memory leak in config file on off nominal
	  paths when glob is allowed If a system allows for its usage,
	  Asterisk will use glob to help parse Asterisk .conf files. The
	  config file loading routine was leaking the memory allocated by
	  the glob() routine when the config file was in an unmodified or
	  invalid state. This patch properly calls globfree in those off
	  nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
	  Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
	  (license 5909)

	* main/features.c: Clean up resources in features on exit This
	  patch cleans up two things features: * It properly unregisters
	  the CLI commands that features registered * It cancels and
	  performs a pthread_join on the created parking thread. This not
	  only properly joins a non-detached thread, but also prevents
	  disposing of the parking lots prior to the parking thread
	  completely exiting. (closes issue ASTERISK-21407) Reported by:
	  Corey Farrell patches: features_shutdown-r2.patch uploaded by
	  Corey Farrell (License 5909)

2013-04-25 02:43 +0000 [r386483]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Change Case On Forcerport For Consistency *
	  Change "ForcerPort" to "Forcerport" to match everywhere else it
	  is displayed

2013-04-22 16:10 +0000 [r386256]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix crash when AMI redirect action redirects two
	  channels out of a bridge. The two party bridging loops were
	  changing the bridge peer pointers without the channel locks held.
	  Thus when ast_channel_massquerade() tested and used the pointer
	  there is a small window of opportunity for the pointers to become
	  NULL even though the masquerade code has the channels locked.
	  (closes issue ASTERISK-21356) Reported by: William luke Patches:
	  jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
	  rmudgett Tested by: William luke

2013-04-19 15:59 +0000 [r386109]  Matthew Jordan <mjordan@digium.com>

	* res/res_timing_pthread.c: Prevent res_timing_pthread from
	  blocking callers There were several reports of deadlock when
	  using res_timing_pthread. Backtraces indicated that one thread
	  was blocked waiting for the write to the pipe to complete and
	  this thread held the container lock for the timers. Therefore any
	  thread that wanted to create a new timer or read an existing
	  timer would block waiting for either the timer lock or the
	  container lock and deadlock ensued. This patch changes the way
	  the pipe is used to eliminate this source of deadlocks: 1) The
	  pipe is placed in non-blocking mode so that it would never block
	  even if the following changes someone fail... 2) Instead of
	  writing bytes into the pipe for each "tick" that's fired the pipe
	  now has two states--signaled and unsignaled. If signaled, the
	  pipe is hot and any pollers of the read side filedescriptor will
	  be woken up. If unsigned the pipe is idle. This eliminates even
	  the chance of filling up the pipe and reduces the potential
	  overhead of calling unnecessary writes. 3) Since we're tracking
	  the signaled / unsignaled state, we can eliminate the exta poll
	  system call for every firing because we know that there is data
	  to be read. (closes issue ASTERISK-21389) Reported by: Matt
	  Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
	  0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
	  uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
	  Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
	  Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
	  by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
	  isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
	  https://reviewboard.asterisk.org/r/2441/

2013-04-19 05:18 +0000 [r386049]  David M. Lee <dlee@digium.com>

	* main/cli.c: cli.c: Properly initialize debug_modules and
	  verbose_modules. This avoids some lock errors on the core set
	  {debug,verbose} commands.

2013-04-16 23:11 +0000 [r385916]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/devicestate.c, res/res_jabber.c: Distributed Device State
	  broken at sites using res_xmpp or res_jabber where Secuity
	  Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not
	  adding AST_EVENT_IE_CACHABLE to the event as each message came
	  in, then devstate_change_collector_cb() was unable to find
	  AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
	  AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
	  ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
	  ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
	  alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2452/

2013-04-15 17:07 +0000 [r385745]  Jason Parker <jparker@digium.com>

	* Makefile: Don't unnecessarily rebuild things on every run of
	  'make'. Review: https://reviewboard.asterisk.org/r/2449/

2013-04-15 14:38 +0000 [r385683]  David M. Lee <dlee@digium.com>

	* BSDmakefile, contrib/realtime/mysql/voicemail_data.sql,
	  build_tools/sha1sum-sh, res/res_mutestream.c,
	  configs/res_curl.conf.sample, tests/test_func_file.c,
	  include/asterisk/select.h, res/res_rtp_multicast.c,
	  include/asterisk/bridging_technology.h, tests/test_locale.c,
	  include/asterisk/bridging_features.h, doc/Makefile,
	  tests/test_poll.c, res/res_timing_kqueue.c,
	  contrib/realtime/mysql/musiconhold.sql,
	  contrib/realtime/mysql/queue_log.sql, channels/sig_ss7.c,
	  channels/sig_ss7.h, channels/chan_multicast_rtp.c,
	  tests/test_expr.c, apps/app_saycounted.c,
	  contrib/realtime/mysql/voicemail_messages.sql: Fix the
	  svn:keywords property on several files. Normally I think keyword
	  expansion is silly, but the one time it would have been good, it
	  didn't work because the property had quotes in it. This patch
	  fixes obviously busted svn:keywords properties.

2013-04-14 02:58 +0000 [r385633-385636]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_multicast.c: Calculate the timestamp for outbound RTP
	  if we don't have timing information This patch calculates the
	  timestamp for outbound RTP when we don't have timing information.
	  This uses the same approach in res_rtp_asterisk. Thanks to both
	  Pietro and Tzafrir for providing patches. (closes issue
	  ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
	  Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
	  by tzafrir (License 5035) rtp-timestamp.patch uploaded by
	  pbertera (License 5943)

	* channels/chan_alsa.c: Don't attempt to create a voice frame on a
	  read error Prior to this patch, a read error in snd_pcm_readi
	  would still be treated as a nominal result when constructing a
	  voice frame from the expected data. Since the value returned is
	  negative, as opposed to the number of samples read, this could
	  result in a crash. With this patch, we now return a null frame
	  when a read error is detected. Note that the patch on
	  ASTERISK-21329 was modified slightly for this commit, in that we
	  bail immediately on detecting the read error, rather than
	  bypassing the construction of the voice frame. (closes issue
	  ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
	  chan_alsa.diff uploaded by kawasaki (License 6489)

2013-04-12 22:34 +0000 [r385551-385593]  Michael L. Young <elgueromexicano@gmail.com>

	* apps/app_queue.c: Fix Manager Segfault When app_queue Is Unloaded
	  When app_queue is unloaded, some manager commands are not being
	  unregistered which result in a segfault. This patch corrects
	  this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann,
	  Corey Farrell Tested by: Corey Farrell Patches:
	  asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
	  Young (license 5026)
	  asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/2444/

	* apps/app_voicemail.c: Fix app_voicemail Segfault And A Few Memory
	  Leaks The original report was that app_voicemail would crash.
	  This was caused by ast_config_load() returning
	  CONFIG_STATUS_FILEINVALID but no checks being performed for that
	  return status. After adding the initial patch to fix this issue,
	  Jaco Kroon (jkroon) added some fixes to memory leaks he had
	  discovered. During review, Walter Doekes (wdoekes) suggested
	  adding a helper function in order to determine if we had a valid
	  configuration or not. This patch does the following: * Creates a
	  helper function to check if the configuration is valid * Adds
	  calls to the new helper function where appropiate * Fixes memory
	  leaks where the code returned without running
	  ast_config_destroy() on the configuration that was loaded (closes
	  issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
	  Kroon, Michael L. Young Patches:
	  asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
	  (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2443/

2013-04-12 08:46 +0000 [r385402-385429]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_iax2.c: IAX2 defer_full_frames fail to get sent
	  Ensure iax2_process_thread is signalled when a deferred frame is
	  queued to it. (issue ASTERISK-18827) Reported by: alecdavis
	  Tested by: alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2426/

	* channels/chan_iax2.c: IAX2, prevent network thread starting
	  before all helper threads are ready On startup, it's possible for
	  a frame to arrive before the processing threads were ready. In
	  iax2_process_thread() the first pass through falls into
	  ast_cond_wait, should a frame arrive before we are at
	  ast_cond_wait, the signal will be ignored. The result
	  iax2_process_thread stays at ast_cond_wait forever, with deferred
	  frames being queued. Fix: When creating initial idle
	  iax2_process_threads, wait for init_cond to be signalled after
	  each thread is started. (issue ASTERISK-18827) Reported by:
	  alecdavis Tested by: alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2427/

2013-04-10 14:22 +0000 [r385170-385190]  Matthew Jordan <mjordan@digium.com>

	* res/res_config_ldap.c: Use LDAP memory management functions
	  instead of Asterisk's When MALLOC_DEBUG is enabled with
	  res_config_ldap, issues (munmap_chunk: invalid pointer errors)
	  can occur as the memory is being allocated with Asterisk's
	  wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
	  library's wrappers. This patch uses the LDAP library's wrappers
	  where appropriate, so that compiling with MALLOC_DEBUG doesn't
	  cause more problems than it solves. Note that the patch listed
	  below was modified slightly for this commit to account for some
	  additional memory allocation/deallocations. (closes issue
	  ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
	  patches: issue18789-1.8-r316873.patch uploaded by seanbright
	  (License 5060)

	* channels/chan_sip.c: Fix crash in chan_sip when a core initiated
	  op occurs at the same time as a BYE When a BYE request is
	  processed in chan_sip, the current SIP dialog is detached from
	  its associated Asterisk channel structure. The tech_pvt pointer
	  in the channel object is set to NULL, and the dialog persists for
	  an RFC mandated period of time to handle re-transmits. While this
	  process occurs, the channel is locked (which is good).
	  Unfortunately, operations that are initiated externally have no
	  way of knowing that the channel they've just obtained (which is
	  still valid) and that they are attempting to lock is about to
	  have its tech_pvt pointer removed. By the time they obtain the
	  channel lock and call the channel technology callback, the
	  tech_pvt is NULL. This patch adds a few checks to some channel
	  callbacks that make sure the tech_pvt isn't NULL before using it.
	  Prime offenders were the DTMF digit callbacks, which would crash
	  if AMI initiated a DTMF on the channel at the same time as a BYE
	  was received from the UA. This patch also adds checks on
	  sip_transfer (as AMI can also cause a callback into this
	  function), as well as sip_indicate (as lots of things can queue
	  an indication onto a channel). Review:
	  https://reviewboard.asterisk.org/r/2434/ (closes issue
	  ASTERISK-20225) Reported by: Jeff Hoppe

2013-04-08 23:34 +0000 [r385047]  Rusty Newton <rnewton@digium.com>

	* configs/extconfig.conf.sample: Modified the list of keys for the
	  driver backends for sake of sample clarity Added a line showing
	  the mapping of "mysql" to res_config_mysql available in add-ons.
	  We used "mysql" as an example driver key in the sample, but
	  didn't show what module it mapped too. Also added a subtitle
	  above the list of keys for driver backends.

2013-04-08 19:55 +0000 [r385008]  Michael L. Young <elgueromexicano@gmail.com>

	* UPGRADE.txt, channels/chan_sip.c: Fix For Not Overriding The
	  Default Settings In chan_sip The initial report was that the
	  "nat" setting in the [general] section was not having any effect
	  in overriding the default setting. Upon confirming that this was
	  happening and looking into what was causing this, it was
	  discovered that other default settings would not be overriden as
	  well. This patch works similar to what occurs in build_peer(). We
	  create a temporary ast_flags structure and using a mask, we
	  override the default settings with whatever is set in the
	  [general] section. In the bug report, the reporter who helped to
	  test this patch noted that the directmedia settings were being
	  overriden properly as well as the nat settings. (closes issue
	  ASTERISK-21225) Reported by: Alexandre Vezina Tested by:
	  Alexandre Vezina, Michael L. Young Patches:
	  asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2386/

2013-04-04 19:31 +0000 [r384779]  Michael L. Young <elgueromexicano@gmail.com>

	* contrib/realtime/postgresql/realtime.sql,
	  contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c:
	  Backport Appropiate NAT Setting Cleanup In ASTERISK-20904, the
	  focus was around the changes to NAT that took place in Asterisk
	  11. Since the report stated that 1.8 was fine, we didn't take a
	  look at 1.8 at the time. While working on ASTERISK-21225, I could
	  see that 1.8 would benefit from having some of those changes
	  applied to it. This patch does the following: * The important
	  part of this patch is that it sets the peer's flags earlier in
	  build_peer so that the code properly uses the peer's flags based
	  on the peer's configuration. * constify req parameter in
	  check_via() * update realtime schemas under the contrib directory
	  to handle properly the NAT settings available in 1.8 as well as
	  to handle the changes made in 11 to make upgrading easier when
	  installing newer versions of Asterisk (closes issue
	  ASTERISK-21243) Reported by: Michael L. Young Patches:
	  asterisk-20904-changes_for_1.8.diff Michael L. Young (license
	  5026) Review: https://reviewboard.asterisk.org/r/2422/

2013-04-03 20:13 +0000 [r384685]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample: chan_dahdi: Add
	  inband_on_proceeding compatibility option. The new
	  inband_on_proceeding option causes Asterisk to assume inband
	  audio may be present when a PROCEEDING message is received. Q.931
	  Section 5.1.2 says the network cannot assume that the CPE side
	  has attached to the B channel at this time without explicitly
	  sending the progress indicator ie informing the CPE side to
	  attach to the B channel for audio. However, some non-compliant
	  ISDN switches send a PROCEEDING without the progress indicator ie
	  indicating inband audio is available and assume that the CPE
	  device has connected the media path for listening to ringback and
	  other messages. ASTERISK-17834 which causes this issue was
	  dealing with a non-compliant network switch. (closes issue
	  ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett

2013-04-03 17:05 +0000 [r384640]  Matthew Jordan <mjordan@digium.com>

	* funcs/func_channel.c: Update documentation for CHANNEL function
	  Document that you can read/write the 'accountcode' and 'amaflags'
	  on a channel.

2013-04-02 17:33 +0000 [r384544]  David M. Lee <dlee@digium.com>

	* Makefile: Fixed spurious rebuilds of func_version.
	  func_version.so was being rebuilt every time, because build.h was
	  changing every build, because of the cleantest dependency that
	  was added in r384410 to fix parallel make bugs. Now build.h will
	  only be created if it does not exist, which was the original
	  behavior of the Makefile.

2013-04-01 13:18 +0000 [r384410]  David M. Lee <dlee@digium.com>

	* Makefile: Fix parallel make problems. Occasionally, make -j would
	  fail due to missing includes, or other unusual errors. This was
	  due to the 'cleantest' target, which was designed to force a make
	  clean when some change in the code would cause the typical
	  depedency checking to fail. Several targets in the main Makefile
	  did not depend upon cleantest, hence would run in parallel to it.
	  By adding the dependency, make -j runs happily now. Review:
	  https://reviewboard.asterisk.org/r/2418/

2013-03-29 16:23 +0000 [r384325]  Jonathan Rose <jrose@digium.com>

	* apps/app_voicemail.c: app_voicemail: Add blank argument to
	  externnotify if no context argument At least one call to
	  run_externnotify provides a NULL context parameter and because
	  the snprintf statement doesn't account for a NULL context
	  parameter, it simply writes '(null)' to the arguments string
	  instead. This patch makes it write two quotes back to back for
	  that argument instead in the event of a NULL context. (closes
	  issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
	  modified from patch-20130306 uploaded by Karsten Wemheuer
	  (License 5930)

2013-05-17  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.22.0 Released.

2013-05-13  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.22.0-rc2 Released.

	* Distributed Device State broken at sites using res_xmpp or res_jabber
	  where Secuity Advisory AST-2012-015 is inplace

	  res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the
	  event as each message came in, then devstate_change_collector_cb()
	  was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted
	  incorrectly to AST_DEVSTATE_NOT_CACHABLE.

	* Fix CDR not being created during an externally initiated blind
	  transfer

	  Way back when in the dark days of Asterisk 1.8.9, blind transferring
	  a call in a context that included the 'h' extension would
	  inadvertently execute the hangup code logic on the transferred
	  channel. This was a "bad thing". The fix was to properly check for
	  the softhangup flags on the channel and only execute the 'h'
	  extension logic (and, in later versions, hangup handler logic) if
	  the channel was well and truly dead (Jim).

	  Unfortunately, CDRs are fickle. Setting the softhangup flag when we
	  detected that the channel was leaving the bridge (but not to die)
	  caused some crucial snippet of CDR code, lying in ambush in the
	  middle of the bridging code, to not get executed. This had the
	  effect of blowing away one of the CDRs that is typically created
	  during a blind transfer.

	  While we live and die by the adage "don't touch CDRs in release
	  branches", this was our bad. The attached patch restores the CDR
	  behavior, and still manages to not run the 'h' extension during a
	  blind transfer (at least not when it's supposed to).

	  Thanks to Steve Davies for diagnosing this and providing a fix.

	* Prevent res_timing_pthread from blocking callers

	  There were several reports of deadlock when using res_timing_pthread.
	  Backtraces indicated that one thread was blocked waiting for the
	  write to the pipe to complete and this thread held the container lock
	  for the timers.  Therefore any thread that wanted to create a new
	  timer or read an existing timer would block waiting for either the
	  timer lock or the container lock and deadlock ensued.

	  This patch changes the way the pipe is used to eliminate this source
	  of deadlocks:

	  1) The pipe is placed in non-blocking mode so that it would never
	     block even if the following changes someone fail...

	  2) Instead of writing bytes into the pipe for each "tick" that's
	     fired the pipe now has two states--signaled and unsignaled. If
	     signaled, the pipe is hot and any pollers of the read side
	     filedescriptor will be woken up. If unsigned the pipe is idle.
	     This eliminates even the chance of filling up the pipe and reduces
	     the potential overhead of calling unnecessary writes.

	  3) Since we're tracking the signaled / unsignaled state, we can
	  eliminate the exta poll system call for every firing because we know
	  that there is data to be read.

	* Fix crash when AMI redirect action redirects two channels out of a
	  bridge.

	  The two party bridging loops were changing the bridge peer pointers
	  without the channel locks held.  Thus when ast_channel_massquerade()
	  tested and used the pointer there is a small window of opportunity
	  for the pointers to become NULL even though the masquerade code has
	  the channels locked.

2013-03-28  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.22.0-rc1 Released.

2013-03-27 19:50 +0000 [r384162]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Address uninitialized conditional that
	  valgrind found

2013-03-27 18:49 +0000 [r384118]  Matthew Jordan <mjordan@digium.com>

	* main/http.c: Fix a file descriptor leak in off nominal path While
	  looking at the security vulnerability in ASTERISK-20967, Walter
	  noticed a file descriptor leak and some other issues in off
	  nominal code paths. This patch corrects them. Note that this
	  patch is not related to the vulnerability in ASTERISK-20967, but
	  the patch was placed on that issue. (closes issue ASTERISK-20967)
	  Reported by: wdoekes patches:
	  issueA20967_file_leak_and_unused_wkspace.patch uploaded by
	  wdoekes (License 5674)

2013-03-27 17:02 +0000 [r384048]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c: Fix white noise on SRTP decryption When
	  res_rtp_asterisk.c was altered to avoid attempting to apply
	  unprotect algorithms to non-audio RTP packets, the test used was
	  incorrect. This caused the audio packets to not be decrypted and
	  resulted in loud white noise on the other endpoint (or both
	  endpoints depending on the call legs involved). The test now
	  properly checks the version field in the RTP header to ensure
	  that RTP and RTCP are decrypted while other types of packets are
	  not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
	  Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
	  uploaded by Kinsey Moore

2013-03-27 14:53 +0000 [r383976-383981]  Matthew Jordan <mjordan@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: AST-2013-003:
	  Prevent username disclosure in SIP channel driver When
	  authenticating a SIP request with alwaysauthreject enabled,
	  allowguest disabled, and autocreatepeer disabled, Asterisk
	  discloses whether a user exists for INVITE, SUBSCRIBE, and
	  REGISTER transactions in multiple ways. The information is
	  disclosed when: * A "407 Proxy Authentication Required" response
	  is sent instead of a "401 Unauthorized" response * The presence
	  or absence of additional tags occurs at the end of "403
	  Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response
	  is sent instead of "403 Forbidden" response after a
	  retransmission * Retransmission are sent when a matching peer did
	  not exist, but not when a matching peer did exist. This patch
	  resolves these various vectors by ensuring that the responses
	  sent in all scenarios is the same, regardless of the presence of
	  a matching peer. This issue was reported by Walter Doekes, OSSO
	  B.V. A substantial portion of the testing and the solution to
	  this problem was done by Walter as well - a huge thanks to his
	  tireless efforts in finding all the ways in which this setting
	  didn't work, providing automated tests, and working with Kinsey
	  on getting this fixed. (closes issue ASTERISK-21013) Reported by:
	  wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8
	  uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10
	  uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11
	  uploaded by kmoore, wdoekes (License 6273, 5674)

	* main/http.c: AST-2013-002: Prevent denial of service in HTTP
	  server AST-2012-014, fixed in January of this year, contained a
	  fix for Asterisk's HTTP server for a remotely-triggered crash.
	  While the fix put in place fixed the possibility for the crash to
	  be triggered, a denial of service vector still exists with that
	  solution if an attacker sends one or more HTTP POST requests with
	  very large Content-Length values. This patch resolves this by
	  capping the Content-Length at 1024 bytes. Any attempt to send an
	  HTTP POST with Content-Length greater than this cap will not
	  result in any memory allocation. The POST will be responded to
	  with an HTTP 413 "Request Entity Too Large" response. This issue
	  was reported by Christoph Hebeisen of TELUS Security Labs (closes
	  issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
	  AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
	  AST-2013-002-10.diff uploaded by mmichelson (License 5049)
	  AST-2013-002-11.diff uploaded by mmichelson (License 5049)

2013-03-26 02:23 +0000 [r383839-383863]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Resolve deadlock between SIP registration
	  and channel based functions In r373424, several reentrancy
	  problems in chan_sip were addressed. As a result, the SIP channel
	  driver is now properly locking the channel driver private
	  information in certain operations that it wasn't previously. This
	  exposed two latent problems either in register_verify or by
	  functions called by register_verify. This includes: * Holding the
	  private lock while calling sip_send_mwi_to_peer. This can create
	  a new sip_pvt via sip_alloc, which will obtain the channel
	  container lock. This is a locking inversion, as any channel
	  related lock must be obtained prior to obtaining the SIP channel
	  technology private lock. * Holding the privat elock while calling
	  sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
	  sip_poke_peer can create a new SIP private, causing the same
	  locking inversion. Note that this locking inversion typically
	  occured when CLI commands were run while a SIP REGISTER request
	  was being processed, as many CLI commands (such as 'sip show
	  channels', 'core show channels', etc.) have to obtain the channel
	  container lock. (issue ASTERISK-21068) Reported by: Nicolas
	  Bouliane (issue ASTERISK-20550) Reported by: David Brillert
	  (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
	  ASTERISK-21296) Reported by: Gabriel Birke

	* main/cdr.c: Resolve deadlock between pending CDR and batch CDR
	  locks r375757 attempted to resolve a race condition between
	  multiple submissions of CDRs while in batch mode from attempting
	  to destroy the scheduled batch submission by extending the batch
	  CDR lock. Unfortunately, this causes a deadlock between the
	  pending CDR lock and the batch CDR lock. This patch resolves the
	  intent of r375757 by simply providing a new lock that protects
	  the scheduling of the batches. The original batch CDR lock is
	  kept to protect manipulation of the batch CDR settings, but has
	  been placed such that it is not held when the pending lock is
	  held. Thanks to Chase Venters for providing lock analysis on the
	  issue. (issue ASTERISK-21162) Reported by: Chase Venters

2013-03-26 01:32 +0000 [r383835]  Russell Bryant <russell@russellbryant.com>

	* apps/app_meetme.c: Fix multi-station answer race condition. When
	  an SLA trunk is ringing (inbound call on the trunk) Asterisk will
	  make outbound calls to the stations that have that trunk. If more
	  than one station answers the call at the same time, all channels
	  other than the first one to answer are left in a bad state. The
	  channel gets leaked, is not connected to anything, and there's no
	  way to get rid of it. We now properly clean up these losing
	  channels by hanging up on them. Since they lost the race, as we
	  process their answer, there is no ringing trunk for them to
	  answer.

2013-03-25 23:19 +0000 [r383796]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Set the CALLERID(dnid-num-plan) for incoming
	  ISDN calls. The CALLEDTON channel variable is set for incoming
	  ISDN calls to the lower 7 bits of the Q.931
	  type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
	  should have the same value. (closes issue ASTERISK-21248)
	  Reported by: rmudgett

2013-03-25 12:35 +0000 [r383667]  Sean Bright <sean@malleable.com>

	* res/res_config_curl.c: Properly delimit post data in
	  res_config_curl.

2013-03-20 20:22 +0000 [r383460]  Walter Doekes <walter+asterisk@wjd.nu>

	* funcs/func_curl.c: Have func_curl log a warning when a curl
	  request fails. Review: https://reviewboard.asterisk.org/r/2403/

2013-03-19 15:50 +0000 [r383340]  David M. Lee <dlee@digium.com>

	* codecs/Makefile: Removed codecs/g722/*.i on make clean

2013-03-15 12:49 +0000 [r383165]  Kinsey Moore <kmoore@digium.com>

	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c:
	  tcptls: Prevent unsupported options from being set AMI, HTTP, and
	  chan_sip all support TLS in some way, but none of them support
	  all the options that Asterisk's TLS core is capable of
	  interpreting. This prevents consumers of the TLS/SSL layer from
	  setting TLS/SSL options that they do not support. This also gets
	  tlsverifyclient closer to a working state by requesting the
	  client certificate when tlsverifyclient is set. Currently, there
	  is no consumer of main/tcptls.c in Asterisk that supports this
	  feature and so it can not be properly tested. Review:
	  https://reviewboard.asterisk.org/r/2370/ Reported-by: John
	  Bigelow Patch-by: Kinsey Moore (closes issue AST-1093)

2013-03-15 01:32 +0000 [r383120-383124]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: When a session timer expires during a T.38
	  call, re-invite with correct SDP When a session timer expires
	  during a dialog that has re-negotiated to T.38 and Asterisk is
	  the refresher, Asterisk will send a re-INVITE with an SDP
	  containing audio media only. This causes some hilarity with the
	  poor fax session under weigh. This patch corrects that by sending
	  T.38 parameters if we are in the middle of a T.38 session.
	  (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches:
	  dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
	  uploaded by nbansal (License 6418)

	* pbx/pbx_spool.c: Fix processing of call files when using KQueue
	  on OS X In certain situations, call files are not processed when
	  using KQueue with pbx_spool. Asterisk was sending an invalid
	  timeout value when the spool directory is empty, causing the call
	  to kevent to error immediately. This can create a tight loop,
	  increasing the CPU load on the system. (closes issue
	  ASTERISK-21176) Reported by: Carlton O'Riley patches:
	  kqueue_osx.patch uploaded by coriley (License 6473)

2013-03-14 16:56 +0000 [r383061]  Jason Parker <jparker@digium.com>

	* autoconf/ast_ext_lib.m4: Fix whitespace in AST_EXT_LIB_CHECK
	  macro.

2013-03-12 21:15 +0000 [r382939-382942]  Michael L. Young <elgueromexicano@gmail.com>

	* addons/res_config_mysql.c: Fix Sorting Order For Parking Lots
	  Stored In Static Realtime When retrieving the parking lots from a
	  MySQL database table, the current order is "filename, cat_metric
	  desc, var_metric asc, category". If there are multiple parking
	  lots with the same cat_metric but different categories,
	  everything is being sorted on cat_metric first resulting in
	  errors when loading the parking lots. This patch fixes the
	  problem by sorting on the category field first, then the
	  cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
	  Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
	  (license 5026)

	* contrib/realtime/postgresql/realtime.sql,
	  contrib/realtime/mysql/sippeers.sql: Update Contributed Realtime
	  Schema Files - IPv6 Addresses This commit updates some fields in
	  the contributed realtime schema files to handle IPv6 addresses.
	  (closes issue ASTERISK-21173) Reported by: Torrey Searle Patches:
	  realtime_sql.patch Torrey Searle (license 5334)
	  asterisk-21173-update-ip-fields.diff Michael L. Young (license
	  5026)

2013-03-12 16:20 +0000 [r382847]  Matthew Jordan <mjordan@digium.com>

	* UPGRADE.txt, channels/chan_sip.c: Include the Username field in
	  SIP Registry events when Status is registered In ASTERISK-17888,
	  the AMI Registry event during SIP registrations was supposed to
	  include the Username field. Somehow, one of the events was
	  missed. This patch corrects that - the Username field should be
	  included in all AMI Registry events involving SIP registrations.
	  (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by:
	  Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov
	  (license 6479)

2013-03-06 18:22 +0000 [r382513]  Kinsey Moore <kmoore@digium.com>

	* apps/app_page.c: Correct app_page documentation The 'A' and 'n'
	  options for Page() mention that the announcement will be played
	  simultaneously. This is not necessarily the case.

2013-03-05 03:46 +0000 [r382409]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Fix several unreleased mutex locks that
	  cause problem with processing calls Reported by: Daniel Bohling
	  Tested by: Daniel Bohling (Closes issue ASTERISK-21119)

2013-02-28 17:09 +0000 [r382227-382233]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
	  attempting to set caller ID A deadlock can occur in chan_iax2
	  when it attempts to set the caller ID, as it already holds the
	  iax2 private lock and improperly fails to obtain the channel lock
	  before calling ast_set_callerid. By not safely obtaining the
	  channel lock, a locking inversion can take place, causing a
	  deadlock. This patch solves this by calling the required deadlock
	  avoidance functions that obtain the channel lock before setting
	  the caller ID. Thanks to Pavel for fixing my syntax errors and
	  testing this patch out. (closes issue ASTERISK-21128) Reported
	  by: Pavel Troller Tested by: Pavel Troller patches:
	  ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
	  ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
	  (license 6302)

	* UPGRADE.txt, apps/app_meetme.c: Let channels joining a MeetMe
	  conference opt out of the denoiser For some channel drivers,
	  specifically those that have a varying rate in the number of
	  audio samples, the audio quality for a MeetMe conference can be
	  exceedingly poor. This is due to a unilateral application of the
	  DENOISE function in func_speex to channels joining the
	  conference. The denoiser function in the speex library is
	  initialized with the number of audio samples in each sample that
	  will be provided to it. If the number of audio samples changes,
	  the denoiser has to be thrown away and re-initialized. While this
	  could be worked around by removing func_speex, that doesn't help
	  if you actually use the denoiser with other channels on the
	  system. This patches does the following: * Checks for the
	  presence of func_speex as opposed to codec_speex when determining
	  if the DENOISE function is present (which is where the function
	  is actually implemented) * Adds an option to MeetMe 'n' that
	  causes the denoiser to not be applied to a channel when it joins.
	  This keeps the current behavior the default, but let's users
	  disable the denoiser if it causes problems on their system.
	  Review: https://reviewboard.asterisk.org/r/2358 (closes issue
	  AST-1062) Reported by: Thomas Arimont

2013-02-27 16:16 +0000 [r382153-382171]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Relax dialog checking in
	  get_sip_pvt_byid_locked so it works when the dialog is forked.
	  (closes issue ASTERISK-20638) Reported by: eelcob Patches:
	  pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
	  6442)

	* configure, include/asterisk/autoconfig.h.in: Fix the configure
	  script over here as well.

2013-02-26 19:37 +0000 [r382110]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configure, configure.ac: Consider linux-gnuspe as linux-gnu * The
	  powerpcspe Linux port uses linux-gnuspe as the OS string. * Our
	  build system shouldn't really care for that, so just call it
	  linux-gnu. * Original report: Roland Stigge ,
	  http://bugs.debian.org/701505 Review:
	  https://reviewboard.asterisk.org/r/2357/

2013-02-26 19:30 +0000 [r382107]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Correct RPID parsing for unquoted
	  display-name. Parsing Remote-Party-ID will now succeed if
	  display-name is of the *(token LWS) kind and not just the
	  quoted-string kind. Review:
	  https://reviewboard.asterisk.org/r/2341/

2013-02-26 19:06 +0000 [r382087]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/Makefile: Remove unneeded linux-gnueabi* As of r380520 the
	  configure scripts converts the value of linux-gnueabi* of OSARCH
	  to "linux-gnu". So no point in testing for those values.

2013-02-25 12:48 +0000 [r381916-382021]  Matthew Jordan <mjordan@digium.com>

	* addons/res_config_mysql.c: Clean up use of va_end/va_args in
	  res_config_mysql There were several problems using variadic
	  argument macros in res_config_mysql. * Improper use of va_end.
	  Multiple calls to va_end were possible resulting in an unbalanced
	  matching of va_start/va_end. * Calls to va_arg after a possible
	  encounter of a SENTINEL value. This patch corrects those errors.
	  (closes issue ASTERISK-19451) Reported by: wdoekes patches:
	  ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)

	* channels/chan_jingle.c: Set the sin_family on the bind address
	  socket during initialization Somehow, chan_jingle has managed to
	  operate for years without setting the sin_family on its bindaddr
	  socket. This patch properly sets the field during initial module
	  load to AF_INET. Note that the patch on the issue was modified
	  slightly to change the initialization of the socket from
	  allocation of a chan_jingle private to the module initialization,
	  as the bindaddr object (which is static) only needs to have the
	  address set once. (closes issue ASTERISK-19341) Reported by:
	  andre valentin patches: 0105-chan_jingle.patch uploaded by
	  avalentin (License 6064)

	* main/manager.c: Don't display the AMI ALL class authorization for
	  users if they don't have it When converting AMI class
	  authorizations to a string representation, the method always
	  appends the ALL class authorization. This is especially important
	  for events, as they should always communicate that class
	  authorization - even if the event itself does not specify ALL as
	  a class authorization for itself. (Events have always assumed
	  that the ALL class authorization is implied when they are raised)
	  Unfortunately, this did mean that specifying a user with
	  restricted class authorizations would show up in the 'manager
	  show user' CLI command as having the ALL class authorization.
	  Rather then modifying the existing string manipulation function,
	  this patch adds a function that will only return a string if the
	  field being compared explicitly matches class authorization field
	  it is being compared against. This prevents ALL from being
	  returned unless it is actually specified for the user. (closes
	  issue ASTERISK-20397) Reported by: Johan Wilfer

	* apps/app_parkandannounce.c: Make ParkAndAnnounce return to
	  priority + 1 when return context is not defined The
	  ParkAndAnnounce application documentation for the optional
	  return_context parameter states the following: return_context The
	  goto-style label to jump the call back into after timeout.
	  Default 'priority+1'. Unfortunately, the application was sending
	  the channel back into the dialplan at 'priority', which is the
	  ParkAndAnnounce application call. This causes an infinite loop of
	  the channel constantly being parked, announced, timed out,
	  parked, announced, timed out... while fun, especially for those
	  callers you wish to drive to the end of madness, this was not the
	  intent of the application. (closes issue ASTERISK-20113) Reported
	  by: serginuez patches: app_parkandannounce.diff uploaded by
	  serginuez (License 6405)

2013-02-21 22:44 +0000 [r381847]  Matthew Jordan <mjordan@digium.com>

	* configure, configure.ac: Properly detect launchd Asterisk was a
	  little too pro-active in claiming that it found launchd. On
	  systems without launchd - such as FreeBSD - this resulted in
	  certain items in Asterisk that conflict with launchd to not be
	  selectable, such as res_timing_kqueue. (closes issue
	  ASTERISK-20749) Reported by: Oleg Baranov

2013-02-19 19:16 +0000 [r381770]  kharwell <kharwell@localhost>:

	* main/features.c: Write the correct callid to the data1 field in
	  queue_log for transfer events. The incorrect callid was being
	  written to the "data1" field in queue_log table for transfer
	  events. The callid of the queue was being written instead of the
	  transfer target's callid. This now gets the correct "transfer to"
	  number and places that in the "data1" field of the queue_log
	  table when a transfer event is triggered. (closes issue
	  ASTERISK-19960) Reported by: vladimir shmagin

2013-02-18 20:28 +0000 [r381668]  Walter Doekes <walter+asterisk@wjd.nu>

	* configs/sip.conf.sample: Remove "registertrying" and add
	  "rtp_engine" from/to sip.conf.sample The "registertrying" option
	  was removed in r343220. The "rtp_engine" option was added in
	  r186078 but erroneously named "engine" in the sample. Note that
	  there is no global sip setting for a different engine.

2013-02-14 19:41 +0000 [r381466]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO
	  because it isn't a real hangup. It doesn't hurt to check
	  AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside
	  of a bridge. (issue ASTERISK-20492)

2013-02-14 03:42 +0000 [r381364]  Matthew Jordan <mjordan@digium.com>

	* apps/app_db.c: Don't throw a spurious error when using DBdeltree
	  The function call ast_db_deltree returns the number of row
	  deleted, or a negative number if it failed. DBdeltree was
	  treating any non-zero return as an error, causing a spurious
	  verbose error message to be displayed. This patch handles the
	  return code of ast_db_deltree correctly. (closes issue
	  ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff
	  uploaded by ianc (License #5955)

2013-02-12 20:16 +0000 [r381281]  Mark Michelson <mmichelson@digium.com>

	* main/rtp_engine.c: Do not allow native RTP bridging if
	  packetization of media streams differs. The RTP engine will no
	  longer allow for local and remote native RTP bridges if
	  packetization of streams differs. Allowing native bridging in
	  this scenario has been known to cause FAX failures. (closes
	  ASTERISK-20650) Reported by: Maciej Krajewski Patches:
	  ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
	  Review: https://reviewboard.asterisk.org/r/2319

2013-02-11 20:46 +0000 [r381216]  kharwell <kharwell@localhost>:

	* apps/app_playback.c: Properly load say.conf upon reload of module
	  app_playback. If say.conf did not exists prior to originally
	  loading module app_playback it would not load on subsequent
	  reloads of the module once it had been created. This occurred
	  because upon reload of the app_playback module it would only load
	  a new configuration if an old one had previously existed. This
	  fix simply removed the association between checking if an old
	  configuration existed and the loading of the new one. (closes
	  issue ASTERISK-20800) Reported by: pgoergler

2013-02-06 20:10 +0000 [r380973]  David M. Lee <dlee@digium.com>

	* channels/chan_sip.c: Fixed failing test from r380696. When I
	  added my extensive suite of session timer unit tests, apparently
	  one of them was failing and I never noticed. If neither Min-SE
	  nor Session-Expires is set in the header, it was responding with
	  a Session-Expires of the global maxmimum instead of the
	  configured max for the endpoint. (issue ASTERISK-20787)

2013-02-05 18:09 +0000 [r380853]  Richard Mudgett <rmudgett@digium.com>

	* main/dial.c: Separate option_types[] from the struct definition.
	  Updated the option_types[] doxygen comment.

2013-01-31 19:56 +0000 [r380696]  David M. Lee <dlee@digium.com>

	* channels/chan_sip.c: Process session timers, even if
	  Session-Expires header is missing Previously, Asterisk only
	  processed session timer information if both the 'Supported:
	  timer' and 'Session-Expires' headers were present. However, the
	  Session-Expires header is optional. If we were to receive a
	  request with a Min-SE greater than our configured
	  session-expires, we would respond with a 'Session-Expires' header
	  that was too small. This patch cleans the situation up a bit,
	  always processing timer information if the 'Supported: timer'
	  header is present. (closes issue ASTERISK-20787) Reported by:
	  Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/

2013-01-31 00:22 +0000 [r380572-380611]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h: Make CHECK_BLOCKING() debug message
	  more useful. Change the displayed pthread value to hex format so
	  it can be easily matched with CLI core show threads or gdb.

	* channels/chan_dahdi.c: chan_dahdi: Fix "dahdi show channels
	  group" for groups greater than 31. The variable type used was not
	  large enough to hold a group bit field.

2013-03-27  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.21.0-rc2 Released.

	* Do not allow native RTP bridging if packetization of media streams
	differs.

	The RTP engine will no longer allow for local and remote native RTP
	bridges	if packetization of streams differs. Allowing native bridging
	in this	scenario has been known to cause FAX failures.

	* Resolve deadlock between pending CDR and batch CDR locks

	r375757 attempted to resolve a race condition between multiple
	submissions of CDRs while in batch mode from attempting to destroy the
	scheduled batch	submission by extending the batch CDR lock. Unfortunately,
	this causes a deadlock between the pending CDR lock and the batch CDR lock.
	This patch resolves the intent of r375757 by simply providing a new lock
	that protects the scheduling of the batches. The original batch CDR lock
	is kept to protect manipulation of the batch CDR settings, but has been
	placed such that it is not held when the pending lock is held.

	Thanks to Chase Venters for providing lock analysis on the issue.

	* Resolve deadlock between SIP registration and channel based
	functions

	In r373424, several reentrancy problems in chan_sip were addressed. As
	a result, the SIP channel driver is now properly locking the channel
	driver private information in certain operations that it wasn't previously.
	This exposed two latent problems either in register_verify or by functions
	called by register_verify. This includes:
	 * Holding the private lock while calling sip_send_mwi_to_peer. This
	 can create a new sip_pvt via sip_alloc, which will obtain the channel
	 container lock. This is a locking inversion, as any channel related lock
	 must be obtained prior to obtaining the SIP channel technology private
	 lock.
	 * Holding the private lock while calling sip_poke_peer. In the same vein as
         sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
         the same locking inversion.

	Note that this locking inversion typically occured when CLI commands were run
	while a SIP REGISTER request was being processed, as many CLI commands (such
	as 'sip show channels', 'core show channels', etc.) have to obtain the channel
	container lock.

	* AST-2013-002: Prevent denial of service in HTTP server

	AST-2012-014, fixed in January of this year, contained a fix for
	Asterisk's HTTP server for a remotely-triggered crash. While the fix put in
	place fixed the possibility for the crash to be triggered, a denial of
	service vector still exists with that solution if an attacker sends one or
	more HTTP POST requests with very large Content-Length values. This patch
	resolves this by capping the Content-Length at 1024 bytes. Any attempt to send
	an HTTP POST with Content-Length greater than this cap will not result in any
	memory allocation. The POST will be responded to with an HTTP 413 "Request
	Entity Too Large" response.

	This issue was reported by Christoph Hebeisen of TELUS Security Labs

	* AST-2013-003: Prevent username disclosure in SIP channel driver

	When authenticating a SIP request with alwaysauthreject enabled,
	allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether
	a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in
	multiple ways. The information is disclosed when:
	 * A "407 Proxy Authentication Required" response is sent instead of a
	   "401 Unauthorized" response
	 * The presence or absence of additional tags occurs at the end of
	   "403 Forbidden" (such as "(Bad Auth)")
	 * A "401 Unauthorized" response is sent instead of "403 Forbidden"
	   response after a retransmission
	 * Retransmission are sent when a matching peer did not exist, but not
	   when a matching peer did exist.
	This patch resolves these various vectors by ensuring that the responses sent
	in all scenarios is the same, regardless of the presence of a matching peer.

	This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
	the testing and the solution to this problem was done by Walter as well - a
	huge thanks to his tireless efforts in finding all the ways in which this
	setting didn't work, providing automated tests, and working with Kinsey on
	getting this fixed.

	* Fix white noise on SRTP decryption

	When res_rtp_asterisk.c was altered to avoid attempting to apply
	unprotect algorithms to non-audio RTP packets, the test used was
	incorrect. This caused the audio packets to not be decrypted and
	resulted in loud white noise on the other endpoint (or both endpoints
	depending on the call legs involved). The test now properly checks the
	version field in the RTP header to ensure that RTP and RTCP are
	decrypted while other types of packets are not.

2013-01-30  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.21.0-rc1 Released.

2013-01-30 17:44 +0000 [r380451-380520]  Matthew Jordan <mjordan@digium.com>

	* configure, configure.ac: Support building Asterisk for Raspberry
	  Pi/Raspbian with hard-float support Building Asterisk on Raspbian
	  with hard-float support fails as it uses the string
	  'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'.
	  This patch modifies the configure script for Asterisk such that
	  it will match on any string beginning with 'linux-gnueabi', as
	  opposed to requiring an explicit match. (closes issue
	  ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian
	  Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse
	  (license 6459) linux-gnueabihf-autoconf.patch uploaded by
	  Christian Hesse (license 6459)

	* channels/chan_sip.c: Perform case insensitive comparisons for
	  T.38 attributes RFC5347 section 2.5.2 states the following: ...
	  The attribute "T38MaxBitRate" was once incorrectly registered
	  with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
	  T.38 examples and common implementation practice, the form
	  "T38MaxBitRate" SHOULD be generated by implementations conforming
	  to this package. In general, it is RECOMMENDED that
	  implementations of this package accept lowercase, uppercase, and
	  mixed upper/lowercase encodings of all the T.38 attributes. ...
	  Asterisk currently does not perform case insensitive matching on
	  the T.38 attributes. This causes the T38MaxBitRate attribute to
	  be negotiated at 2400 baud instead of 14400 (or whatever value
	  you actually wanted). This patch makes it so that when we compare
	  T.38 attributes, we do so in a case insensitive fashion. Note
	  that while the issue reporter did not directly write the patch,
	  they contributed to it (and would have provided one themselves if
	  the license had gone through a tad faster), and hence get
	  attribution for it. (closes issue ASTERISK-20897) Reported by:
	  Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill

	* res/res_calendar_icalendar.c: Fix memory leak in
	  res_calendar_icalendar The ICalendar module had a systemic memory
	  leak on each fetch of data from the ICalendar source. The
	  previous fetched data was not being properly disposed. This patch
	  makes it so that before each fetch of data, we dispose of the
	  previously fetched data. (closes issue ASTERISK-21012) Reported
	  by: Joel Vandal Tested by: Joel Vandal

2013-01-29 17:22 +0000 [r380364]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_agent.c: chan_agent: Prevent multiple channels from
	  logging in as the same agent. Multiple channels logging in as the
	  same agent can result in dead channels waiting for a condition
	  signal that will never come because another channel thread stole
	  it. A symptom is chan_sip repeatedly generating warning messages
	  about rescheduling autodestruction of dialogs with an agent
	  channel owner. * Made only login_exec() (the app AgentLogin)
	  clear the agent_pvt->chan pointer to prevent multiple channels
	  from logging in as the same agent. agent_read(), agent_call(),
	  and agent_set_base_channel() no longer disconnect the agent
	  channel from the agent_pvt. This also eliminates the need to keep
	  checking for agent_pvt->chan being NULL. * Made agent_hangup()
	  not wake up the AgentLogin agent thread until it is done. * Made
	  agent_request() not able to get the agent until he has logged in
	  and any wrapup time has expired. * Made agent_request() use
	  ast_hangup() instead of agent_hangup() to correctly dispose of a
	  channel. * Removed agent_set_base_channel(). Nobody calls it and
	  it is a bad thing in general. * Made only agent_devicestate()
	  determine the current device state of an agent. Note: Agent group
	  device states have never been supported. Review:
	  https://reviewboard.asterisk.org/r/2260/

2013-01-29 17:05 +0000 [r380347]  David M. Lee <dlee@digium.com>

	* channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER for
	  SRTP. (again) The original fix (r380043) for getting Asterisk to
	  respond with the correct tag overlooked some corner cases, and
	  the fact that the same code is in 1.8. This patch moves the
	  building of the crypto line out of sdp_crypto_process(). Instead,
	  it merely copies the accepted tag. The call to sdp_crypto_offer()
	  will build the crypto line in all cases now, using a tag of "1"
	  in the case of sending offers. (closes issue ASTERISK-20849)
	  Reported by: José Luis Millán Review:
	  https://reviewboard.asterisk.org/r/2295/

2013-01-29 02:02 +0000 [r380297]  Matthew Jordan <mjordan@digium.com>

	* autoconf/ast_check_pwlib.m4, configure: Update configure script
	  to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the
	  configure script fails due to grep returning multiple matches for
	  the pattern it searches for. This patch updates the pattern
	  matching to return only the actual version for the symbol
	  searched for, PTLIB_VERSION. (closes issue ASTERISK-20980)
	  Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch
	  uploaded by Stefan Reuter (license 5339)

2013-01-28 21:06 +0000 [r380254]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c, channels/iax2.h: Correct the number of
	  available call numbers in IAX2. There is currently an edge case
	  where call number 32768 might be allocated for a call, even
	  though the IAX2 protocol requires call numbers be only 15 bits.
	  This resulted in some unpredictable behavior when call number
	  32678 is chosen. This patch was mostly written by Richard Mudgett
	  via ReviewBoard. I'm just committing it. Review:
	  https://reviewboard.asterisk.org/r/2293/

2013-01-28 01:52 +0000 [r380210]  Russell Bryant <russell@russellbryant.com>

	* main/file.c: Change cleanup ordering in filestream destructor.
	  This patch came about due to a problem observed where wav files
	  had an empty header. The header is supposed to be updated in
	  wav_close(). It turns out that this was broken when the
	  cache_record_files option from asterisk.conf was enabled. The
	  cleanup code was moving the file to its final destination
	  *before* running the close() method of the file destructor, so
	  the header didn't get updated. Another problem here is that the
	  move was being done before actually closing the FILE *. Finally,
	  the last bug fixed here is that I noticed that wav_close() checks
	  for stream->filename to be non-NULL. In the previous cleanup
	  order, it's checking a pointer to freed memory. This doesn't
	  actually cause anything to break, but it's treading on dangerous
	  waters. Now the free() of stream->filename is happening after the
	  format module's close() method gets called, so it's safer.
	  Review: https://reviewboard.asterisk.org/r/2286/

2013-01-23 00:19 +0000 [r379963]  Richard Mudgett <rmudgett@digium.com>

	* main/astobj2.c: Attempt to be more helpful when using a bad ao2
	  object pointer. Backport of -r360626 with some enhancements. Put
	  the external obj pointer in the message instead of the internal
	  version.

2013-01-22 18:21 +0000 [r379885]  Jonathan Rose <jrose@digium.com>

	* sounds/Makefile, apps/app_meetme.c: app_meetme: Use new prompts
	  for administrator menu The old prompts for the administrator menu
	  were inadequate. They didn't mention that the menu had additional
	  options through the 8 key and pressing the 8 key wouldn't reveal
	  what those options were. This patch fixes all of that while also
	  organizing code pertaining to each individual menu type which was
	  previously all stored in one gigantic function along with many of
	  the basic conference functions. (closes issue AST-996) Reported
	  by: John Bigelow Review:
	  http://reviewboard.digium.internal/r/360/

2013-01-22 14:43 +0000 [r379760-379825]  Matthew Jordan <mjordan@digium.com>

	* apps/app_meetme.c: Fix station ringback; trunk hangup issues in
	  SLA This patch fixes two bugs: * If an outbound call is made from
	  a SLA phone using SLAStation, then there is no ringtone audible
	  to the phone that originates the call. The indication of the
	  ringing was not being passed to the SLA station; this patch fixes
	  that by passing through the progress indications. * If an SLA
	  station hangs up before the called party answers, then the
	  channel to the called party continues to ring until a timeout
	  occurs. If the called party manages to answer, Asterisk attempts
	  to connect the called party to a non-existant MeetMe room. This
	  patch corrects the behavior by abandoning the call attempt if it
	  detects that the SLA station is no longer in use while attempting
	  to call the called party. Review:
	  https://reviewboard.asterisk.org/r/2275/ (closes issue
	  ASTERISK-20462) Reported by: dkerr patches:
	  asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
	  5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
	  5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
	  asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
	  asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
	  5558)

	* UPGRADE.txt, contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk,
	  contrib/init.d/rc.archlinux.asterisk,
	  contrib/scripts/safe_asterisk, main/asterisk.c,
	  contrib/init.d/rc.suse.asterisk,
	  contrib/init.d/rc.mandriva.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.redhat.asterisk: Update init.d scripts to
	  handle stderr; readd splash screen for remote consoles When
	  r376428 was commited to re-order start up sequences to be more
	  tolerant of forking with thread primitives, a few items were
	  changed that caused changes in behavior on some distros. This
	  includes: * Not displaying the splash screen on a remote console.
	  * Displaying an error message on stderr when a remote console
	  cannot connect to a running instance of Asterisk. In the first
	  case, the splash screen was re-added (thanks to Michael L.
	  Young). In the second case, the various init.d scripts were
	  modified to pipe stderr to /dev/null, as the error message is
	  useful - if you execute a remote console or a remote console
	  command execution and it fail, it should tell you. Note that the
	  error message was always present, it just failed to be printed
	  prior to r376428. Much thanks to the folks who quickly reported
	  this problem, provided solutions, and promptly tested the various
	  init.d scripts on a variety of distros. (closes issue
	  ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
	  Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
	  asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
	  5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
	  (license 6283)

2013-01-21 18:27 +0000 [r379718]  Kinsey Moore <kmoore@digium.com>

	* codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
	  frames When iLBC is being used with a jitter buffer and the jb
	  has to interpolate frames, it generates frames with a null
	  pointer and a non-zero datalen. This is now handled properly.
	  (closes issue ASTERISK-20914) Reported By: John McEleney Patches:
	  ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)

2013-01-21 04:59 +0000 [r379645]  Andrew Latham <lathama@gmail.com>

	* contrib/scripts/install_prereq: Add LDAP libraries to install
	  script Add LDAP dev package to Debian/Ubuntu install list.
	  Existed in Redhat already. Merged from 11 to Trunk in 379643.
	  Sorry for forgeting 1.8 (issue ASTERISK-20886)

2013-01-21 04:05 +0000 [r379608]  Matthew Jordan <mjordan@digium.com>

	* apps/app_minivm.c: Fix crash in app_minivm when mime encoding
	  string An incorrect string initializations was left in
	  ast_str_encode_mime from the patch that converted string
	  manipulations to use ast_str strings (r191140). The string
	  initialization causes a crash when ast_str_set is called on the
	  string later on in the function. (closes issue ASTERISK-18697)
	  Reported by: Chris Boot patches:
	  minivm-null-pointer-dereference-fix.patch uploaded by bootc
	  (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
	  Tested by: Chris Warr

2013-01-19 20:41 +0000 [r379547]  Walter Doekes <walter+asterisk@wjd.nu>

	* configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
	  builtin roundf() for systems lacking it. (closes issue
	  ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
	  Reported-by: Ovidiu Sas

2013-01-18 23:26 +0000 [r379509]  Matthew Jordan <mjordan@digium.com>

	* main/asterisk.c: Fix astcanary startup problem due to wrong pid
	  value from before daemon call When Asterisk forks itself into the
	  background via a call to daemon, it must re-set the pid value of
	  the new process. Otherwise, astcanary gets the pid value of the
	  process before the fork, which prevents it from running. Asterisk
	  eventually starts lowering its priority, as it can no longer
	  communicate with the proverbial canary in the coal mine. This
	  patch ensures that the correct process identifier is used by
	  astcanary. (closes issue ASTERISK-20947) Reported by: Jakob
	  Hirsch Tested by: mjordan patches:
	  asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
	  (license 6113)

2013-01-18 05:23 +0000 [r379392]  David M. Lee <dlee@digium.com>

	* channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
	  channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
	  headers. Record-Route parsing copied the header into a char[256]
	  array, which can be a problem if the header is longer than that.
	  This patch parses the header in place, without the copy, avoiding
	  the issue. In addition to the original patch, I added a unit test
	  for the new get_in_brackets_const function. (closes issue
	  ASTERISK-20837) Reported by: Corey Farrell Patches:
	  chan_sip-build_route-optimized-rev1.patch uploaded by Corey
	  Farrell (license 5909) (with minor changes by dlee)

2013-01-17 02:28 +0000 [r379342]  Matthew Jordan <mjordan@digium.com>

	* addons/chan_mobile.c: Fix issue where chan_mobile fails to bind
	  to first available port Per the bluez API, in order to bind to
	  the first available port, the rc_channel field of the socket
	  addressing structure used to bind the socket should be set to 0.
	  Previously, Asterisk had set the rc_channel field set to 1,
	  causing it to connect to whatever happens to be on port 1. We
	  could probably not explicitly set rc_channel to 0 since we memset
	  the struct earlier, but explicitly setting it will hopefully
	  prevent someone from coming in and setting it to some explicit
	  port in the future. (closes issue ASTERISK-16357) Reported by:
	  challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
	  eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
	  Nikolay Ilduganov (license 6253)

2013-01-16 22:45 +0000 [r379310]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Further fix misinformation in the description of
	  manager MailboxStatus command. The description still claimed that
	  it returned the number of messages rather than whether there were
	  messages waiting.

2013-01-16 21:12 +0000 [r379276]  Jason Parker <jparker@digium.com>

	* contrib/scripts/install_prereq: Reduce number of packages
	  install_prereq installs on Debian systems. 'search' will look for
	  any package containing the name provided, so we need to force a
	  more exact search.

2013-01-16 17:40 +0000 [r379226]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: chan_misdn: Fix compile error. (issue
	  ASTERISK-15456)

2013-01-16 04:10 +0000 [r379091-379178]  Matthew Jordan <mjordan@digium.com>

	* addons/chan_mobile.c: Fix parsing SMSSRC for SMS messages The
	  parser for SMS messages would incorrectly parse out the from
	  number. The parsing would incorrectly start scanning for the from
	  number at the same index as the first double quote ("); this
	  would inadvertently cause it to treat the first double quote as
	  the terminating double quote for the from number as well. The
	  SMSSRC should now populate correctly. (closes issue
	  ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
	  patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
	  issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
	  sms-sender-fix.diff uploaded by roeften (license 5884)

	* channels/chan_misdn.c: Set the INVALID_EXTEN channel variable
	  when chan_misdn forces the 'i' extension The chan_misdn channel
	  driver will send a channel with an invalid destination to the 'i'
	  extension itself if said extension can be reached. It forgot,
	  however, to set the INVALID_EXTEN channel variable when it
	  bounces the channel to this extension. Dialplan writers
	  everywhere moaned at yet another inconsistency. This is yet
	  another example of why duplicating logic in multiple places
	  results in bugs that stick around in Jira for just under three
	  years. Yes: ASTERISK-15456 was created on January 18th, 2010.
	  Patch committed on January 15th, 2013. Ouch. (closes issue
	  ASTERISK-15456) Reported by: Thomas Omerzu patches:
	  chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
	  5927)

	* bridges/bridge_softmix.c: Prevent crash in ConfBridge due to race
	  condition when channels leave bridge When a channel leaves a
	  bridge, a race condition existed where the bridge_channel's pvt
	  structure would be accessed after it was disposed of. This patch
	  prevents that by setting the pointer to the pvt to NULL prior to
	  disposing of it. Note that this patch is a backport from Asterisk
	  10. This particular race condition was fixed as part of the
	  larger code rework that occurred for that release. The solution
	  to this problem was pointed out by Gunnar Harms in
	  ASTERISK-16640. (closes issue ASTERISK-16640) Reported by:
	  thomas987 (closes issue ASTERISK-16835) Reported by: saghul

2013-01-14 15:11 +0000 [r379001]  David M. Lee <dlee@digium.com>

	* channels/chan_sip.c: Fix XML encoding of 'identity display' in
	  NOTIFY messages, continued. When r378933 was merged into 1.8, it
	  should have also escaped remote_display, since it will have the
	  same XML encoding problem when the caller/callee roles are
	  reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter

2013-01-13 21:15 +0000 [r378967]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c: Reset RTP timestamp; sequence number on
	  SSRC change In r370252 for ASTERISK-18404, Asterisk's handling of
	  RTP was modified to better account for out of order RTP packets.
	  This was accomplished by using the RTP timestamp and sequence
	  number to check for out of order packets. However, when a SSRC
	  change occurs, the timestamp and sequence number will no longer
	  have any relation to the previously received packets. The
	  variables tracking the timestamp and sequence number therefore
	  have to be reset. (closes issue ASTERISK-20906) Reported by:
	  Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
	  Brolman (license #6442)

2013-01-12 06:26 +0000 [r378933]  David M. Lee <dlee@digium.com>

	* main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
	  tests/test_xml_escape.c (added): Fix XML encoding of 'identity
	  display' in NOTIFY messages. XML encoding in chan_sip is
	  accomplished by naively building the XML directly from strings.
	  While this usually works, it fails to take into account escaping
	  the reserved characters in XML. This patch adds an
	  'ast_xml_escape' function, which works similarly to
	  'ast_uri_encode'. This is used to properly escape the
	  local_display attribute in XML formatted NOTIFY messages. Several
	  things to note: * The Right Thing(TM) to do would probably be to
	  replace the ast_build_string stuff with building an ast_xml_doc.
	  That's a much bigger change, and out of scope for the original
	  ticket, so I refrained myself. * It is with great sadness that I
	  wrote my own ast_xml_escape function. There's one in libxml2, but
	  it's knee-deep in libxml2-ness, and not easily used to one-off
	  escape a string. * I only escaped the string we know is causing
	  problems (local_display). At least some of the other strings are
	  URI-encoded, which should be XML safe. Rather than figuring out
	  what's safe and escaping what's not, it would be much cleaner to
	  simply build an ast_xml_doc for the messages and let the XML
	  library do the XML escaping. Like I said, that's out of scope.
	  (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
	  Guenther Kelleter Review:
	  http://reviewboard.digium.internal/r/365/ ........ Merged
	  revision 378919 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

2013-01-09 20:26 +0000 [r378733-378776]  David M. Lee <dlee@digium.com>

	* main/rtp_engine.c: Fix end condition in
	  ast_rtp_lookup_mime_multiple2. The erroneous end condition would
	  never include the AST_RTP_CISCO_DTMF flag in the debug output.
	  (closes issue ASTERISK-20772) Reported by: Xavier Hienne

	* include/asterisk/causes.h: Replace errant tabs with spaces in
	  causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
	  Patches: notabs.dif uploaded by snuffy (license 5024)

2013-01-08 20:22 +0000 [r378663]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: app_queue: Fix multiple calls to a queue member
	  that is in only one queue. When ringinuse=no queue members can
	  receive more than one call if these calls happen at nearly the
	  same time. * Fix so a queue member does not receive more than one
	  call from a queue. NOTE: This fix does not prevent multiple calls
	  to a member if the member is in more than one queue. * Did some
	  refactoring to eliminate some code redundancy. (issue
	  ASTERISK-16115) Reported by: nik600 Patches:
	  jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Modified

2013-01-04 22:54 +0000 [r378591]  Jonathan Rose <jrose@digium.com>

	* res/res_srtp.c: res_srtp: Prevent a crash from occurring due to
	  srtp_create failures in srtp_create Under some circumstances,
	  libsrtp's srtp_create function deallocates memory that it wasn't
	  initially responsible for allocating. Because we weren't
	  initially aware of this behavior, this memory was still used in
	  spite of being unallocated during the course of the
	  srtp_unprotect function. A while back I made a patch which would
	  set this value to NULL, but that exposed a possible condition
	  where we would then try to check a member of the struct which
	  would cause a segfault. In order to address these problems,
	  ast_srtp_unprotect will now set an error value when it ends
	  without a valid SRTP session which will result in the caller of
	  srtp_unprotect observing this error and hanging up the relevant
	  channel instead of trying to keep using the invalid session
	  address. (closes issue ASTERISK-20499) Reported by: Tootai
	  Review:
	  https://reviewboard.asterisk.org/r/2228/diff/#index_header

2013-01-04 21:12 +0000 [r378554]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix SIP Notify Messages To Have The Proper
	  IP Address In The FROM Field On a multihomed server when sending
	  a NOTIFY message, we were not figuring out which network should
	  be used to contact the peer. This patch fixes the problem by
	  calling ast_sip_ouraddrfor() and then build_via() so that our
	  NOTIFY message contains the correct IP address. Also, a debug
	  message is being added to help follow the call-id changes that
	  occur. This was helpful for confirming that the IP address was
	  set properly since the call-id contains the IP address. It also
	  will be helpful for troubleshooting purposes when following a
	  call in the debug logs. (closes issue ASTERISK-20805) Reported
	  by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches:
	  asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/2255/

2013-01-04 21:12 +0000 [r378553]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Don't pass STUN packets through the SRTP
	  unprotect function. (closes issue AST-1036) Reported by: jbigelow

2013-01-03 22:09 +0000 [r378514]  Michael L. Young <elgueromexicano@gmail.com>

	* apps/app_queue.c: Fix Queue Log Reporting Every Call
	  COMPLETECALLER With "h" Extension Present When the "h" extension
	  is present within the context of the queue, all calls are being
	  reported COMPLETECALLER even when the agent is hanging up the
	  call. This patch checks to see if the agent hung-up or not
	  instead of only relying on checking if the queue (caller) channel
	  hung-up or not. It would appear that having the h extension in
	  the mix, the pbx goes to the h extension, "hanging-up" the queue
	  channel and triggering the reporting of COMPLETECALLER. (closes
	  issue ASTERISK-20743) Reported by: call Tested by: call, Michael
	  L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2256/

2013-01-03 19:40 +0000 [r378456-378486]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_agent.c: chan_agent: Fix wrapup time wait response.
	  * Made agent_cont_sleep() and agent_ack_sleep() stop waiting if
	  the wrapup time expires. agent_cont_sleep() had tried but
	  returned the wrong value to stop waiting. * Made
	  agent_ack_sleep() take a struct agent_pvt pointer instead of a
	  void pointer for better type safety.

	* channels/chan_agent.c: chan_agent: Misc code cleanup. * Fix
	  off-nominal path resource cleanup in agent_request(). * Create
	  agent_pvt_destroy() to eliminate inlined versions in many places.
	  * Pull invariant code out of loop in add_agent(). * Remove
	  redundant module user references in login_exec(). * Remove unused
	  struct agent_pvt logincallerid[] member. * Remove some redundant
	  code in agent_request().

2013-01-03 18:35 +0000 [r378455]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c: Add missing test event This test event was
	  missing from channel.c causing the dial_LS_options test to fail
	  intermittently because of a race condition where most code paths
	  emitted the test event but this one did not. The dial_LS_options
	  test should stop bouncing now.

2013-01-03 17:41 +0000 [r378427]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_agent.c: chan_agent: Fix agent_indicate() locking.
	  Avoid deadlock potential with local channels and simplify the
	  locking.

2013-01-02 21:48 +0000 [r378375]  Matthew Jordan <mjordan@digium.com>

	* main/config.c, funcs/func_realtime.c: Prevent crashes from
	  occurring when reading from data sources with large values When
	  reading configuration data from an Asterisk .conf file or when
	  pulling data from an Asterisk RealTime backend, Asterisk was
	  copying the data on the stack for manipulation. Unfortunately, it
	  is possible to read configuration data or realtime data from some
	  data source that provides a large blob of characters. This could
	  potentially cause a crash via a stack overflow. This patch
	  prevents large sets of data from being read from an ARA backend
	  or from an Asterisk conf file. (issue ASTERISK-20658) Reported
	  by: wdoekes Tested by: wdoekes, mmichelson patches: *
	  issueA20658_dont_process_overlong_config_lines.patch uploaded by
	  wdoekes (license 5674) * issueA20658_func_realtime_limit.patch
	  uploaded by wdoekes (license 5674)

2013-01-02 21:08 +0000 [r378356]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, main/manager.c, main/features.c: Fix
	  AMI redirect action with two channels failing to redirect both
	  channels. The AMI redirect action can fail to redirect two
	  channels that are bridged together. There is a race between the
	  AMI thread redirecting the two channels and the bridge thread
	  noticing that a channel is hungup from the redirects. * Made the
	  bridge wait for both channels to be redirected before exiting. *
	  Made the AMI redirect check that all required headers are present
	  before proceeding with the redirection. * Made the AMI redirect
	  require that any supplied ExtraChannel exist before proceeding.
	  Previously the code fell back to a single channel redirect
	  operation. (closes issue ASTERISK-18975) Reported by: Ben Klang
	  (closes issue ASTERISK-19948) Reported by: Brent Dalgleish
	  Patches: jira_asterisk_19948_v11.patch (license #5621) patch
	  uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak
	  Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/

2013-01-02 16:54 +0000 [r378269-378303]  Matthew Jordan <mjordan@digium.com>

	* main/devicestate.c, include/asterisk/channel.h,
	  channels/chan_iax2.c, res/res_jabber.c, main/channel.c,
	  channels/chan_dahdi.c, include/asterisk/event_defs.h,
	  channels/chan_skinny.c, main/features.c, main/event.c,
	  apps/app_confbridge.c, funcs/func_devstate.c, res/res_calendar.c,
	  include/asterisk/devicestate.h, channels/chan_local.c,
	  apps/app_meetme.c, channels/chan_sip.c, channels/chan_agent.c:
	  Prevent exhaustion of system resources through exploitation of
	  event cache Asterisk maintains an internal cache for devices in
	  the event subsystem. The device state cache holds the state of
	  each device known to Asterisk, such that consumers of device
	  state information can query for the last known state for a
	  particular device, even if it is not part of an active call. The
	  concept of a device in Asterisk can include entities that do not
	  have a physical representation. One way that this occurred was
	  when anonymous calls are allowed in Asterisk. A device was
	  automatically created and stored in the cache for each anonymous
	  call that occurred; this was possible in the SIP and IAX2 channel
	  drivers and through channel drivers that utilized the
	  res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
	  These devices are never removed from the system, allowing
	  anonymous calls to potentially exhaust a system's resources. This
	  patch changes the event cache subsystem and device state
	  management to no longer cache devices that are not associated
	  with a physical entity. (issue ASTERISK-20175) Reported by:
	  Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
	  patches: event-cachability-3.diff uploaded by jcolp (license
	  5000)

	* res/res_jabber.c, channels/sip/include/sip.h,
	  channels/chan_sip.c, main/http.c: Resolve crashes due to large
	  stack allocations when using TCP Asterisk had several places
	  where messages received over various network transports may be
	  copied in a single stack allocation. In the case of TCP, since
	  multiple packets in a stream may be concatenated together, this
	  can lead to large allocations that overflow the stack. This patch
	  modifies those portions of Asterisk using TCP to either favor
	  heap allocations or use an upper bound to ensure that the stack
	  will not overflow: * For SIP, the allocation now has an upper
	  limit * For HTTP, the allocation is now a heap allocation instead
	  of a stack allocation * For XMPP (in res_jabber), the allocation
	  has been eliminated since it was unnecesary. Note that the HTTP
	  portion of this issue was independently found by Brandon Edwards
	  of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
	  wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
	  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
	  5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
	  wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
	  uploaded by wdoekes (license 5674)

2012-12-31 14:41 +0000 [r378217]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
	  without crypto info This ensures that Asterisk rejects encrypted
	  media streams (RTP/SAVP audio and video) that are missing
	  cryptographic keys and ensures that the incoming SDP is
	  consistent with RFC4568 as far as having a crypto attribute
	  present for any SAVP streams. Review:
	  https://reviewboard.asterisk.org/r/2204/

2012-12-20 21:38 +0000 [r378164]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Give the causes[] a struct name.

2012-12-20 20:26 +0000 [r378147]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/rtp_engine.h: Adjust RTP instance's
	  available_formats callback to return the correct type. The RTP
	  engine public function that gets the available formats expects a
	  format_t to be returned; however when calling into an RTP
	  instance's callback to get the available formats, the callback
	  returned an int. This never was noticed in Asterisk because the
	  two RTP engines included do not provide an available_formats
	  callback. This introduces an API change, and the proposal for
	  this change was brought up on the Asterisk developers mailing
	  list [1]. There was no public objection to this change, so it is
	  now being put in. (closes AST-1054) reported by Doug Bailey [1]
	  http://lists.digium.com/pipermail/asterisk-dev/2012-December/058058.html

2012-12-18 17:35 +0000 [r378119]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c: Add test events for time limit-related hangups
	  This patch adds hangup-related test events in order to support
	  testing of time-limited bridges. This aids in testing the S() and
	  L() bridge options. (issue SWP-4713)

2012-12-17 23:07 +0000 [r378088-378092]  Richard Mudgett <rmudgett@digium.com>

	* main/loader.c: Fix potential double free when unloading a module.

	* channels/chan_local.c: Make chan_local module references tied to
	  local_pvt lifetime. The chan_local module references were
	  manually tied to the existence of the ;1 and ;2 channel links. *
	  Made chan_local module references tied to the existence of the
	  local_pvt structure as well as automatically take care of the
	  module references. * Tweaked the wording of the local_fixup()
	  failure warning message to make sense. Review:
	  https://reviewboard.asterisk.org/r/2181/

2012-12-14 21:23 +0000 [r378036]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: app_queue: Revert bad ringinuse=no patch. With
	  the option ringinuse=no set, the patch committed for
	  ASTERISK-16115 causes non-SIP queue members to never be called
	  because the device state is checked after a channel is created to
	  determine if the member is busy. These queue members always get
	  the "Member %s is busy, cannot dial" message. Most channel
	  drivers other than chan_sip use the default device state
	  handling. The default device-state state is considered in use or
	  unknown if the channel exists or not respectively. (closes issue
	  ASTERISK-20801) Reported by: rmudgett Patches:
	  jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
	  patch uploaded by rmudgett

2012-12-13 13:43 +0000 [r377946]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure Min-SE is included in outbound
	  INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
	  value is not 90 (the default) and session timers are not
	  disabled. This has the effect of Asterisk following RFC4028 more
	  closely with regard to 422 responses and preventing situations in
	  which Asterisk would be forced to temporarily accept a call to
	  tear it down based on a Session-Expires below the locally
	  configured Min-SE. (issue SWP-5051) Review:
	  https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
	  Moore Patch-by: Kinsey Moore

2012-12-12 22:39 +0000 [r377922]  Rusty Newton <rnewton@digium.com>

	* sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
	  sounds/Makefile to 1.4.12 for new Extra Sounds releases See
	  CHANGES-* files in English extra 1.4.12 tarballs for new sound
	  prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
	  (closes AST-755) Reported by: John Bigelow

2012-12-11 21:54 +0000 [r377847-377881]  Richard Mudgett <rmudgett@digium.com>

	* main/aoc.c, main/image.c, main/cel.c, main/timing.c,
	  main/channel.c, main/data.c, main/stun.c, main/file.c,
	  main/http.c: Cleanup CLI commands on exit for several files.
	  (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
	  unregister-cli-multiple-all.patch (license #5909) patch uploaded
	  by Corey Farrell

	* main/udptl.c: Cleanup udptl on exit. * Cleanup CLI commands on
	  exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
	  udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
	  Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
	  uploaded by Corey Farrell Modified

2012-12-11 20:45 +0000 [r377840]  Mark Michelson <mmichelson@digium.com>

	* res/res_clialiases.c: Fix crash that can occur if CLI
	  registration fails for an aliased command. A recent memory leak
	  fix in main/cli.c causes an ast_cli_entry's command field to be
	  freed and NULLed if ast_cli_register() fails. res_clialiases was
	  ignoring the return value of ast_cli_register() and was then
	  passing the NULL command off to a a hash function. This resulted
	  in a crash. The fix is not to ignore the erroneous return value.
	  If ast_cli_register() fails, then we do not continue trying to
	  process the current alias.

2012-12-11 20:37 +0000 [r377688-377837]  Richard Mudgett <rmudgett@digium.com>

	* main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
	  CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
	  Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch
	  (license #5909) patch uploaded by Corey Farrell
	  taskprocessor-cleanup-10-only.patch (license #5909) patch
	  uploaded by Corey Farrell Modified

	* main/pbx.c: Cleanup pbx on exit. * Cleanup CLI commands on exit.
	  * Unreference hints and statecbs containers on exit. (issue
	  ASTERISK-20649) Reported by: Corey Farrell Patches:
	  pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
	  Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
	  Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
	  uploaded by Corey Farrell Modified

	* main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
	  destroy verbosers and logchannels lists on exit. (issue
	  ASTERISK-20649) Reported by: Corey Farrell Patches:
	  logger-cleanup-all.patch (license #5909) patch uploaded by Corey
	  Farrell Modified

	* main/indications.c: Cleanup indications on exit. * Made
	  ast_unregister_indication_country() unlink the found tone zone
	  before selecting a new default_tone_zone to make it impossible to
	  select the tone zone being unregistered again. * Ringcadence is
	  no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
	  commands and destroy default_tone_zone on exit. (issue
	  ASTERISK-20649) Reported by: Corey Farrell Patches:
	  indications-cleanup-all.patch (license #5909) patch uploaded by
	  Corey Farrell Modified

	* main/frame.c: Cleanup frame on exit. * Cleanup CLI commands on
	  exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
	  frame-cleanup-1_8-only.patch (license #5909) patch uploaded by
	  Corey Farrell

	* main/event.c: Cleanup event on exit. * Cleanup CLI commands on
	  exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
	  event_shutdown-10-only.patch (license #5909) patch uploaded by
	  Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909)
	  patch uploaded by Corey Farrell

	* main/dnsmgr.c: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
	  and CLI commands on exit. (issue ASTERISK-20649) Reported by:
	  Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
	  patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
	  (license #5909) patch uploaded by Corey Farrell Modified

	* main/db.c: Cleanup astdb on exit. * Cleanup astdb thread and CLI
	  commands on exit. (issue ASTERISK-20649) Reported by: Corey
	  Farrell Patches: db-cleanup-1_8-only.patch (license #5909) patch
	  uploaded by Corey Farrell Modified

2012-12-10 16:51 +0000 [r377623-377655]  Kinsey Moore <kmoore@digium.com>

	* res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
	  When using res_fax_digium, the T.38 CED tone was not being
	  provided properly which would cause some incoming faxes to fail.
	  This was not an issue with res_fax_spandsp since it does not
	  strictly honor the send_ced flag and sends the CED tone whenever
	  receiving a T.38 fax. (closes issue FAX-343) Reported-by:
	  Benjamin Tietz Patch-by: Kinsey Moore

	* channels/chan_sip.c: Handle Session-Expires less than local
	  Min-SE in 200 OK Ensure that a call is immediately torn down if a
	  Session-Expires value received in a 200 OK is less than the local
	  Min-SE. This also prevents Asterisk from allowing calls with
	  Session-Expires below the RFC4028-mandated minimum (90s). (closes
	  issue ASTERISK-20653) Review:
	  https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore

2012-12-10 06:40 +0000 [r377557-377591]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Fix codec mismatch Fix code to send in
	  both rx and tx open stream messages correct codecs. Found that on
	  phase 0/1 phones wrong codecs cause to no audio in some
	  situations. (issue ASTERISK-20183)

	* channels/chan_unistim.c: Fix crash on transfer initiated from
	  insreeen menu on Unistim phones. Removed CDR-related code that
	  moved to do_masquarade before. (closes issue ASTERISK-20417)
	  Reported by: Rudolf Migalin

2013-01-14  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.20.0 Released.

2013-01-09  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.20.0-rc2 Released.

	* AST-2012-014: Resolve crashes due to large stack allocations when
	  using TCP

	  Asterisk had several places where messages received over various
	  network transports may be copied in a single stack allocation. In
	  the case of TCP, since multiple packets in a stream may be
	  concatenated together, this can lead to large allocations that
	  overflow the stack.

	  This patch modifies those portions of Asterisk using TCP to either
	  favor heap allocations or use an upper bound to ensure that the
	  stack will not overflow:
	  * For SIP, the allocation now has an upper limit
	  * For HTTP, the allocation is now a heap allocation instead of a
	    stack allocation
	  * For XMPP, the allocation has been eliminated since it was
	    unnecessary.

	* AST-2012-015: Prevent exhaustion of system resources through
	  exploitation of event cache 

	  Asterisk maintains an internal cache for devices in the event
	  subsystem. The device state cache holds the state of each device
	  known to Asterisk, such that consumers of device state information
	  can query for the last known state for a particular device, even if
	  it is not part of an active call. The concept of a device in
	  Asterisk can include entities that do not have a physical
	  representation. One way that this occurred was when anonymous calls
	  are allowed in Asterisk. A device was automatically created and
	  stored in the cache for each anonymous call that occurred; this was
	  possible in the SIP and IAX2 channel drivers and through channel
	  drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk,
	  Jingle, and Motif). These devices are never removed from the system,
	  allowing anonymous call to potentially exhaust a system's resources.

	  This patch changes the event cache subsystem and device state
	  management to no longer cache devices that are not associated with a
	  physical entity.

	* Revert bad ringinuse=no patch.

	  With the option ringinuse=no set, the patch committed previous for
	  ASTERISK-16115 causes non-SIP queue members to never be called
	  because the device state is checked after a channel is created to
	  determine if the member is busy. These queue members always get the
	  "Member %s is busy, cannot dial" message.

	  Most channel drivers other than chan_sip use the default device
	  state handling. The default device state is considered in use or
	  unknown if the channel exists or not, respectively.

	* Fix multiple calls to a queue member that is only in queue.

	  When ringinuse=no queue members can receive more than one call if
	  these calls happen at nearly the same time. This patch fixes it so a
	  queu member does not receive more than one call from a queue. note
	  that this fix does not prevent multiple calls to a member if hte
	  member is in more than one queue (see ASTERISK-16115).

2012-12-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.20.0-rc1 Released.

2012-12-10 01:38 +0000 [r377487-377509]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/xmldoc.c: Improve documentation by making all of the colors
	  used readable, no matter what the background color is. Dark blue
	  on a black background is unreadable, as is yellow on a light
	  background. This patch turns on the bright attribute for colors
	  when on a dark background and turns *off* the bright attribute
	  when the -W command line option is used (indicating a _light_
	  background). This ensures that text is readable in both cases.
	  Patch by: tilghman Review:
	  https://reviewboard.asterisk.org/r/2224

	* addons/cdr_mysql.c: Remove some dead code and additionally handle
	  a case that wasn't handled.

2012-12-08 00:28 +0000 [r377398-377431]  Richard Mudgett <rmudgett@digium.com>

	* contrib/realtime/mysql/sippeers.sql: Fix order of SIP
	  allow/disallow in MySQL contrib script. Using the contrib
	  sippeers.sql script to create the sippeers MySQL table would
	  result in being unable to place calls if you set the disallow
	  value to all. (closes issue ASTERISK-20756) Reported by: Andre
	  Luis Patches: sippeers.patch patch uploaded by Andre Luis

	* main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
	  allocation dumps.

2012-12-05 16:48 +0000 [r377257]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a SIP request memory leak with TLS
	  connections. During the TLS re-work in chan_sip some TLS specific
	  code was moved into a separate function. This function operates
	  on a copy of the incoming SIP request. This copy was never
	  deinitialized causing a memory leak for each request processed.
	  This function is now given a SIP request structure which it can
	  use to copy the incoming request into. This reduces the amount of
	  memory allocations done since the internal allocated components
	  are reused between packets and also ensures the SIP request
	  structure is deinitialized when the TLS connection is torn down.
	  (closes issue ASTERISK-20763) Reported by: deti

2012-12-05 16:46 +0000 [r377256]  Jonathan Rose <jrose@digium.com>

	* res/res_srtp.c: res_srtp: Fix a crash caused by srtp_dealloc on
	  an already dealloced session When srtp_create fails, the session
	  may be dealloced or just not alloced. At the same time though,
	  the session pointer might not be set to NULL in this process and
	  attempting to srtp_dealloc it again will cause a segfault. This
	  patch checks for failure of srtp_create and sets the session
	  pointer to NULL if it fails. (closes issue ASTERISK-20499)
	  Reported by: tootai Review:
	  https://reviewboard.asterisk.org/r/2228/

2012-12-03 22:51 +0000 [r377037-377165]  Richard Mudgett <rmudgett@digium.com>

	* main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
	  Convert atexits list to a mutex instead of a rd/wr lock. The lock
	  is only write locked. * Move CLI verbose Asterisk ending message
	  to where AMI message is output in really_quit() to avoid further
	  surprises about using stuff already shutdown. (issue
	  ASTERISK-20649) Reported by: Corey Farrell

	* include/asterisk/_private.h, main/stdtime/localtime.c,
	  main/asterisk.c: Cleanup core main on exit. * Cleanup time zones
	  on exit. * Make exit clean/unclean report consistent for AMI and
	  CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey
	  Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch
	  uploaded by Corey Farrell core-cleanup-11-trunk.patch (license
	  #5909) patch uploaded by Corey Farrell Modified

	* main/config.c: Cleanup config cache on exit. (issue
	  ASTERISK-20649) Reported by: Corey Farrell Patches:
	  config-cleanup-all.patch (license #5909) patch uploaded by Corey
	  Farrell

	* main/cli.c: Cleanup CLI resources on exit and CLI command
	  registration errors. (issue ASTERISK-20649) Reported by: Corey
	  Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
	  uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
	  #5909) patch uploaded by Corey Farrell Modified

	* main/cdr.c: Cleanup CDR resources on exit. * Simplify do_reload()
	  return handling since it never returned anything other than 0.
	  (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
	  cdr-cleanup-1_8.patch (license #5909) patch uploaded by Corey
	  Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
	  uploaded by Corey Farrell Modified

	* main/ccss.c: Fix CCSS CLI commands and logger level not
	  unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
	  Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
	  Corey Farrell

2012-11-30 21:30 +0000 [r376950]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: chan_misdn: Fix sending
	  RELEASE_COMPLETE in response to SETUP. Fix sending a
	  RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
	  have a B channel available to assign to the call. (closes issue
	  ABE-2869) Reported by: Guenther Kelleter Patches:
	  setup-reject_2.diff (license #6372) patch uploaded by Guenther
	  Kelleter Modified ........ Merged revision 376949 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

2012-11-30 17:04 +0000 [r376919]  Sean Bright <sean@malleable.com>

	* funcs/func_volume.c: Minor spelling fix to the VOLUME
	  documentation.

2012-11-30 16:12 +0000 [r376901]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix potential crashes during SIP attended
	  transfers. The principal behind this patch is simple. During a
	  transfer, we manipulate channels that are owned by a separate
	  thread than the one we currently are running in, so it makes
	  sense that we need to grab a reference to the channels so that
	  they cannot disappear out from under us. In the wild, crashes
	  were sometimes seen when the transferring party would hang up the
	  call before the transfer target answered the call. The most
	  common place to see the crash occur was when attempting to send a
	  connected line update to the transferer channel. (closes issue
	  ASTERISK-20226) Reported by Jared Smith Patches:
	  ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
	  Tested by: Jared Smith

2012-11-29 22:55 +0000 [r376864-376868]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c: chan_local: Fix local_pvt ref leak in
	  local_devicestate(). Regression introduced by ASTERISK-20390 fix.
	  (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
	  rmudgett

	* channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)

2012-11-29 21:49 +0000 [r376834]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Improve Code Readability And Fix Setting
	  natdetected Flag For 1.8, 10, 11 and trunk we are are improving
	  the code readability. For 11 and trunk, auto nat detection was
	  added. The natdetected flag was being set to 1 when the host
	  address in the VIA header did not specifiy a port. This patch
	  fixes this by setting the port on the temporary sock address used
	  to SIP_STANDARD_PORT in order for the sock address comparison to
	  work properly. (closes issue ASTERISK-20724) Reported by: Michael
	  L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
	  Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2206/

2012-11-29 00:42 +0000 [r376758-376788]  Richard Mudgett <rmudgett@digium.com>

	* main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG atexit unreleased
	  malloc memory summary. * Adds the following CLI commands to
	  control MALLOC_DEBUG reporting of unreleased malloc memory when
	  Asterisk is shut down. memory atexit list on memory atexit list
	  off memory atexit summary byline memory atexit summary byfunc
	  memory atexit summary byfile memory atexit summary off * Made
	  check all remaining allocated region blocks atexit for fence
	  violations. * Increased the allocated region hash table size by
	  about three times. It still isn't large enough considering the
	  number of malloced blocks Asterisk uses. * Made CLI "memory show
	  allocations anomalies" use regions_check_all_fences(). Review:
	  https://reviewboard.asterisk.org/r/2196/

	* main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
	  "memory show allocations" misspelling of anomalies option. The
	  command will still accept the original misspelling. *
	  Miscellaneous tweaks to CLI "memory show allocations" command
	  output format. * Made CLI "memory show summary" summarize by line
	  number instead of by function if a filename is given. * Made CLI
	  "memory show summary" sort its output by filename or
	  function-name/line-number depending upon request. * Miscellaneous
	  tweaks to CLI "memory show summary" command output format.

2012-11-28 16:23 +0000 [r376725]  Jonathan Rose <jrose@digium.com>

	* main/manager.c: manager: Make challenge work with
	  allowmultiplelogin=no Prior to this patch, challenge would yield
	  a multiple logins error if used without providing the username
	  (which isn't really supposed to be an argument to challenge) if
	  allowmultiplelogin was set to no because allowmultiplelogin finds
	  a user with a zero length login name. This check is simply
	  disabled for the challenge action when the username is empty by
	  this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
	  Patches: challenge_action_nomultiplelogin.diff uploaded by
	  Jonathan Rose (license 6182)

2012-11-27 23:47 +0000 [r376627-376688]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt, main/pbx.c: Fix extension matching with the '-'
	  char. The '-' char is supposed to be ignored by the dialplan
	  extension matching. Unfortunately, it's treatment is not handled
	  consistently throughout the extension matching code. * Made the
	  old exten matching code consistently ignore '-' chars. * Made the
	  old exten matching code consistently handle case in the matching.
	  * Made ignore empty character sets. * Fixed ast_extension_cmp()
	  to return -1, 0, or 1 as documented. The only user of it in
	  pbx_lua.c was testing for -1. It was originally returning the
	  strcmp() value for less than which is not usually going to be -1.
	  * Fix character set sorting if the sets have the same number of
	  characters and start with the same character. Character set [0-9]
	  now sorts before [02-9a] as originally intended. * Updated some
	  extension label and priority already in use warnings to also
	  indicate if the extension is aliased. (closes issue
	  ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
	  Harzenetter Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/2201/

	* pbx/pbx_dundi.c, addons/res_config_mysql.c,
	  apps/app_celgenuserevent.c: Remove unnecessary channel module
	  references. * Removed call to ast_module_user_hangup_all() in
	  res_config_mysql.c since it is effectively a noop. No channels
	  can attach a reference to that module. * Removed call to
	  ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
	  of unload_module() has already called it. * Removed redundant
	  channel module references in pbx_dundi.c. The registered dialplan
	  function callback dispatchers for the read/read2/write callbacks
	  already reference the module before calling. * pbx_dundi: Moved
	  unregistering CLI commands, DUNDi switch, and dialplan functions
	  to the first thing the unload_module() does. This will reduce the
	  chance of new channels using DUNDi services while the module is
	  being torn down.

	* include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
	  and use better names. * Update doxygen of AST_LIST_REMOVE().

2012-11-22 23:51 +0000 [r376586]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/lock.h, main/lock.c, main/logger.c:
	  Re-initialize logmsgs mutex upon logger initialization to prevent
	  lock errors Similar to the patch that moved the fork earlier in
	  the startup sequence to prevent mutex errors in the recursive
	  mutex surrounding the read/write thread registration lock, this
	  patch re-initializes the logmsgs mutex. Part of the start up
	  sequence before forking the process into the background includes
	  reading asterisk.conf; this has to occur prior to the call to
	  daemon in order to read startup parameters. When reading in a
	  conf file, log statements can be generated. Since this can't be
	  avoided, the mutex instead is re-initialized to ensure a reset of
	  any thread tracking information. This patch also includes some
	  additional debugging to catch errors when locking or unlocking
	  the recursive mutex that surrounds locks when the DEBUG_THREADS
	  build option is enabled. DO_CRASH or THREAD_CRASH will cause an
	  abort() if a mutex error is detected. (issue ASTERISK-19463)
	  Reported by: mjordan Tesetd by: mjordan

2012-11-20 16:45 +0000 [r376521]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: Add "Require:
	  timer" to 200 OK responses when appropriate. The method by which
	  the Require header is added to 200 responses is inspired by the
	  method that Olle Johansson uses in his darjeeling-prack branch.
	  (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
	  behest of Olle Johansson Review:
	  https://reviewboard.asterisk.org/r/2172

2012-11-19 19:30 +0000 [r376469]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/indications.c, channels/chan_sip.c, main/security_events.c:
	  Fix most leftover non-opaque ast_str uses. Instead of calling
	  str->str, one should use ast_str_buffer(str). Same goes for
	  str->used as ast_str_strlen(str) and str->len as
	  ast_str_size(str). Review:
	  https://reviewboard.asterisk.org/r/2198

2012-11-18 20:11 +0000 [r376428]  Matthew Jordan <mjordan@digium.com>

	* main/utils.c, main/stdtime/localtime.c, main/asterisk.c: Reorder
	  startup sequence to prevent lockups when process is sent to
	  background Although it is very rare and timing dependent, the
	  potential exists for the call to 'daemon' to cause what appears
	  to be a deadlock in Asterisk during startup. This can occur when
	  a recursive mutex is obtained prior to the daemon call executing.
	  Since daemon uses fork to send the process into the background,
	  any threading primitives are unsafe to re-use after the call.
	  Implementations of pthread recursive mutexes are highly likely to
	  store the thread identifier of the thread that previously
	  obtained the mutex. If the mutex was locked prior to the fork, a
	  subsequent unlock operation will potentially fail as the thread
	  identifier is no longer valid. Since the mutex is still locked,
	  all subsequent attempts to grab the mutex by other threads will
	  block. This behavior exhibited itself most often when
	  DEBUG_THREADS was enabled, as this compile time option surrounds
	  the mutexes in Asterisk with another recursive mutex that
	  protects the storage of thread related information. This made it
	  much more likely that a recursive mutex would be obtained prior
	  to daemon and unlocked after the call. This patch does the
	  following: a) It backports a patch from Asterisk 11 that prevents
	  the spawning of the localtime monitoring thread. This thread is
	  now spawned after Asterisk has fully booted. b) It re-orders the
	  startup sequence to call daemon earlier during Asterisk startup.
	  This limits the potential of threading primitives being accessed
	  by initialization calls before daemon is called. c) It removes
	  calls to ast_verbose/ast_log/etc. prior to daemon being called.
	  Developers should send error messages directly to stderr prior to
	  daemon, as calls to ast_log may access recursive mutexes that
	  store thread related information. d) It reorganizes when thread
	  local storage is created for storing lock information during the
	  creation of threads. Prior to this patch, the read/write lock
	  protecting the list of threads in ast_register_thread would
	  utilize the lock in the thread local storage prior to it being
	  initialized; this patch prevents that. On a very related note,
	  this patch will *greatly* improve the stability of the Asterisk
	  Test Suite. Review: https://reviewboard.asterisk.org/r/2197
	  (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
	  mjordan

2012-11-16 19:31 +0000 [r376389]  Jonathan Rose <jrose@digium.com>

	* res/res_monitor.c: monitor: prevent attempts to move/remove
	  recordings skipped with 'i' and 'o'. The i and o options for
	  monitor skip the input and output sides of a recording
	  respectively. This patch addresses a problem in those options
	  when monitor is called without specifying a specific filename
	  where monitor will try to move the recording that was skipped.
	  Since this usually doesn't exist when these options are used, it
	  would produce a warning when it does this in most cases, but it
	  is conceivable that there are use cases where this could result
	  in moving/removing a file unintentionally. (closes issue
	  ASTERISK-20641) Reported by: Jonathan Rose Review:
	  https://reviewboard.asterisk.org/r/2190/

2012-11-15 23:58 +0000 [r376340]  David M. Lee <dlee@digium.com>

	* utils/extconf.c: Fixed extconf.c breakage introduced in r376306.
	  To quote wdoekes: > Note that I'm not confirming legitimacy of
	  having that file in tree at > all. Is anyone using
	  aelparse/conf2ael?

2012-11-15 22:40 +0000 [r376307]  Jonathan Rose <jrose@digium.com>

	* apps/app_meetme.c: app_meetme: Fix channels lingering when hung
	  up under certain conditions Channels would get stuck and MeetMe
	  would repeatedly display an Unable to write frame to channel
	  error in the conf_run function if hung up during certain sound
	  prompts such as during user count announcements. This patch fixes
	  that by reintroducing a hangup check in the meetme's main loop
	  (also in conf_run). (closes issue ASTERISK-20486) Reported by:
	  Michael Cargile Review: https://reviewboard.asterisk.org/r/2187/
	  Patches: meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by
	  Jonathan Rose (license 6182)

2012-11-15 22:27 +0000 [r376306]  David M. Lee <dlee@digium.com>

	* tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
	  include/asterisk/hashtab.h, utils/Makefile,
	  tests/test_astobj2_thrash.c (added), utils/utils.xml,
	  utils/hashtest.c (removed): Migrate hashtest/hashtest2 to be unit
	  tests. Both hashtest and hashtest2 are manual testing apps that
	  thrash hash tables (hashtab and ao2 containers, respectively), by
	  spinning up several threads that randomly insert, delete, lookup
	  and iterate over the hash table. If the app doesn't crash, the
	  hash table probably passes the test. Those utils are not a part
	  of the typical Asterisk build, so they do not usually get
	  compiled. This all makes them less that useful. This patch
	  removes those manual test programs and replaces them with
	  Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
	  also attempts to make the tests more deterministic. * Rather than
	  spinning up some number of threads that operate on the hash table
	  randomly, spin up four threads that concurrenly add, remove,
	  lookup and iterate over the hash table. * Each thread checks the
	  state of the hash table both during and after execution, and
	  indicates a test failure if things are not as expected. * Each
	  thread times out after 60 seconds to prevent deadlocking the unit
	  test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/2189/

2012-11-15 01:43 +0000 [r376262]  Rusty Newton <rnewton@digium.com>

	* apps/app_voicemail.c: Patch to play correct sound file when a
	  voicemail's urgent status is removed We were attempting to play
	  "vm-urgent-removed", which didn't exist. Now we play
	  "vm-marked-nonurgent" which exists and is the correct sound file.
	  Previous behavior was silence and a warning on the CLI. (issue
	  ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
	  Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
	  uploaded by Rusty Newton (license 5829)

2012-11-14 19:48 +0000 [r376232]  Richard Mudgett <rmudgett@digium.com>

	* pbx/pbx_spool.c: Fix call files when astspooldir is relative.
	  Future dated call files are ignored when astspooldir is relative
	  to the current directory. The queue_file() assumed that the qdir
	  needed to be prepended if the given filename did not start with a
	  '/'. If astspooldir is relative it is not going to start from the
	  root directory obviously so it will not start with a '/'. The
	  filename used in queue_file() ultimately results in qdir
	  prepended multiple times. * Made queue_file() not prepend qdir if
	  the filename contains a '/'. (closes issue ASTERISK-20593)
	  Reported by: James Le Cuirot Patches:
	  0004-Fix-future-call-files-from-relative-directories.patch
	  (license #6439) patch uploaded by James Le Cuirot

2012-11-13 18:10 +0000 [r376199]  Brent Eagles <beagles@digium.com>

	* main/channel.c: Patch to prevent stopping the active generator
	  when it is not the silence generator. This patch introduces an
	  internal helper function to safely check whether the current
	  generator is the one that is expected before deactivating it. The
	  current externally accessible ast_channel_stop_generator()
	  function has been modified to be implemented in terms of the new
	  function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad

2012-11-12 20:44 +0000 [r376166]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: Properly check if the "Context" and "Extension"
	  headers are empty in a ShowDialPlan action. The code which
	  handles the ShowDialPlan action wrongly assumed that a non-NULL
	  return value from the function which retrieves headers from an
	  action indicates that the header has a value. This is incorrect
	  and the contents must be checked to see if they are blank.
	  (closes issue ASTERISK-20628) Reported by: jkroon Patches:
	  asterisk-showdialplan-incorrect-error.patch uploaded by jkroon

2012-11-12 20:13 +0000 [r376142]  Michael L. Young <elgueromexicano@gmail.com>

	* main/pbx.c: Fix Dynamic Hints Variable Substition - Underscore
	  Problem When adding a dynamic hint, if an extension contains an
	  underscore no variable subsitution is being performed. This patch
	  changes from checking if the extension contains an underscore to
	  checking if the extension begins with an underscore. (closes
	  issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
	  Steven T. Wheeler, Michael L. Young Patches:
	  asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
	  L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2188/

2012-11-08 21:56 +0000 [r376087]  Mark Michelson <mmichelson@digium.com>

	* res/res_fax.c: Fix a "set but not used" warning on newer gccs.
	  Turns out the "helpful" setting of ms and res in this macro is
	  completely useless after the timeout antipattern fix. If you're a
	  new guy looking to write code, don't write a macro like this one.

2012-11-08 21:05 +0000 [r376029-376058]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c: chan_dahdi/SS7: Made reject incoming call for
	  an in-alarm or blocked channel. If a SS7 call comes in requesting
	  a CIC that is in-alarm, the call is accepted and connects if the
	  extension exists in the dialplan. The call does not have any
	  audio. * Made release the call immediately with circuit
	  congestion cause. (closes issue ASTERISK-20204) Reported by: Tuan
	  Le Patches: jira_asterisk_20204_v1.8.patch (license #5621) patch
	  uploaded by rmudgett

	* main/utils.c, main/astmm.c, main/asterisk.c,
	  include/asterisk/utils.h, include/asterisk/astmm.h: Add
	  MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc().
	  It will return a memory block filled with 0x55. A nonzero value.
	  * Makes free() fill the released memory block and boundary
	  fence's with 0xdeaddead. Any pointer use after free is going to
	  have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is
	  usually an invalid memory address so a crash is expected. * Puts
	  the freed memory block into a circular array so it is not reused
	  immediately. * When the circular array rotates out a memory block
	  to the heap it checks that the memory has not been altered from
	  0xdeaddead. * Made the astmm_log message wording better. * Made
	  crash if the DO_CRASH menuselect option is enabled and something
	  is found. * Fixed a potential alignment issue on 64 bit systems.
	  struct ast_region.data[] should now be aligned correctly for all
	  platforms. * Extracted region_check_fences() from
	  __ast_free_region() and handle_memory_show(). * Updated
	  handle_memory_show() CLI usage help. Review:
	  https://reviewboard.asterisk.org/r/2182/

2012-11-07 17:08 +0000 [r375993-375994]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove some debugging that accidentally made
	  it in the last commit.

	* main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
	  channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  apps/app_waitforring.c, include/asterisk/time.h, apps/app_jack.c,
	  apps/app_dial.c, main/pbx.c, main/rtp_engine.c,
	  channels/chan_sip.c, apps/app_meetme.c, res/res_fax.c,
	  apps/app_record.c, channels/chan_agent.c: Fix misuses of timeouts
	  throughout the code. Prior to this change, a common method for
	  determining if a timeout was reached was to call a function such
	  as ast_waitfor_n() and inspect the out parameter that told how
	  many milliseconds were left, then use that as the input to
	  ast_waitfor_n() on the next go-around. The problem with this is
	  that in some cases, submillisecond timeouts can occur, resulting
	  in the out parameter not decreasing any. When this happens
	  thousands of times, the result is that the timeout takes much
	  longer than intended to be reached. As an example, I had a
	  situation where a 3 second timeout took multiple days to finally
	  end since most wakeups from ast_waitfor_n() were under a
	  millisecond. This patch seeks to fix this pattern throughout the
	  code. Now we log the time when an operation began and find the
	  difference in wall clock time between now and when the event
	  started. This means that sub-millisecond timeouts now cannot play
	  havoc when trying to determine if something has timed out. Part
	  of this fix also includes changing the function ast_waitfor() so
	  that it is possible for it to return less than zero when a
	  negative timeout is given to it. This makes it actually possible
	  to detect errors in ast_waitfor() when there is no timeout.
	  (closes issue ASTERISK-20414) reported by David M. Lee Review:
	  https://reviewboard.asterisk.org/r/2135/

2012-11-06 18:18 +0000 [r375964]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, .cleancount,
	  include/asterisk/features.h, main/channel.c, main/features.c: Fix
	  stuck DTMF when bridge is broken. When a bridge is broken by an
	  AMI Redirect action or the ChannelRedirect application, an in
	  progress DTMF digit could be stuck sending forever. * Made
	  simulate a DTMF end event when a bridge is broken and a DTMF
	  digit was in progress. (closes issue ASTERISK-20492) Reported by:
	  Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt (license
	  #6358) patch uploaded by Jeremiah Gowdy Modified to
	  jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch
	  (license #5621) patch uploaded by rmudgett Tested by: rmudgett
	  Review: https://reviewboard.asterisk.org/r/2169/

2012-12-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.19.0 Released.

2012-12-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.19.0-rc3 Released.

	* chan_local: Fix local_pvt ref leak in local_devicestate().

	Regression introduced by ASTERISK-20390 fix.

	(closes issue ASTERISK-20769)
	Reported by: rmudgett

2012-12-05  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.19.0-rc2 Released.

	* Fix a SIP request memory leak with TLS connections.

	During the TLS re-work in chan_sip some TLS specific code was moved
	into a separate function. This function operates on a copy of the
	incoming SIP request. This copy was never deinitialized causing a
	memory leak for each request processed.

	This function is now given a SIP request structure which it can use
	to copy the incoming request into. This reduces the amount of memory
	allocations done since the internal allocated components are reused
	between packets and also ensures the SIP request structure is
	deinitialized when the TLS connection is torn down.

	(closes issue ASTERISK-20763)
	Reported by: deti

2012-11-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.19.0-rc1 Released.

2012-11-05 22:50 +0000 [r375893]  Matthew Jordan <mjordan@digium.com>

	* bridges/bridge_softmix.c, include/asterisk/timing.h,
	  res/res_musiconhold.c, channels/chan_iax2.c,
	  res/res_fax_spandsp.c, res/res_timing_kqueue.c, main/timing.c,
	  main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
	  res/res_timing_timerfd.c: Refactor ast_timer_ack to return an
	  error and handle the error in timer users Currently, if an
	  acknowledgement of a timer fails Asterisk will not realize that a
	  serious error occurred and will continue attempting to use the
	  timer's file descriptor. This can lead to situations where errors
	  stream to the CLI/log file. This consumes significant resources,
	  masks the actual problem that occurred (whatever caused the timer
	  to fail in the first place), and can leave channels in odd
	  states. This patch propagates the errors in the timing resource
	  modules up through the timer core, and makes users of these
	  timers handle acknowledgement failures. It also adds some
	  defensive coding around the use of timers to prevent using bad
	  file descriptors in off nominal code paths. Note that the patch
	  created by the issue reporter was modified slightly for this
	  commit and backported to 1.8, as it was originally written for
	  Asterisk 10. (issue ASTERISK-20032) Reported by: Jeremiah Gowdy
	  patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy
	  (license 6358)

2012-11-05 21:36 +0000 [r375862]  Richard Mudgett <rmudgett@digium.com>

	* main/loader.c: Add safety NULL pointer check in module user
	  references. Made __ast_module_user_remove() check for NULL
	  pointers. ........ Merged revision 375860 from C.3

2012-11-04 03:06 +0000 [r375727-375800]  Matthew Jordan <mjordan@digium.com>

	* main/manager.c: Don't attempt to purge sessions when no sessions
	  exist Manager's tcp/tls objects have a periodic function that
	  purge old manager sessions periodically. During shutdown, the
	  underlying container holding those sessions can be disposed of
	  and set to NULL before the tcp/tls periodic function is stopped.
	  If the periodic function fires, it will attempt to iterate over a
	  NULL container. This patch checks for whether or not the sessions
	  container exists before attempting to purge sessions out of it.
	  If the sessions container is NULL, we simply return. Note that
	  this error was also caught by the Asterisk Test Suite.

	* main/manager.c: Properly clean up manager resources on exit This
	  patch does two things: 1) It properly unregisters the manager CLI
	  commands 2) It cleans up AMI users on exit. Prior to this patch,
	  the AMI users were not being disposed of properly, resulting in a
	  memory leak. (closes issue ASTERISK-20646) Reported by: Corey
	  Farrell patches: manager_shutdown.patch uploaded by Corey Farrell
	  (license 5909)

	* main/xmldoc.c: Fix memory leak when unloading XML documentation
	  This patch is a modified version of a patch originally committed
	  for the Asterisk 11 branch in r375756. A portion of that patch,
	  that fixed the memory leak during unloading XML documentation,
	  applies to branches 1.8 and 10 as well. The patch for this issue
	  was modified for these two branches. (issue ASTERISK-20648)
	  Reported by: Corey Farrell Tested by: mjordan patches:
	  xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)

	* main/cdr.c: Prevent multiple CDR batches from conflicting when
	  scheduling the CDR write The Asterisk Test Suite caught an error
	  condition where a scheduled CDR batch write can be deleted twice
	  if two channels attempt to post their CDRs at the same time. The
	  batch CDR mutex is locked while the CDRs are appended to the
	  current batch list; however, it is unlocked prior to actually
	  scheduling the CDR write. As such, two threads can attempt to
	  remove the currently scheduled batch write at the same time,
	  resulting in an assertion error. This patch extends the time that
	  the mutex is locked to encompass actually scheduling the write.
	  This prevents two threads from unscheduling the currently
	  scheduled write at the same time.

2012-11-03 03:11 +0000 [r375698]  Andrew Latham <lathama@gmail.com>

	* README, include/asterisk/doxyref.h: Doxygen Updates Replace links
	  to missing text files removed in the 1.6.x series with links to
	  the wiki. Doxygen can handle URLs fine, don't atempt to quote
	  them. Also update the wiki link in the Readme to get everyone on
	  the same page. (issue ASTERISK-20259)

2012-11-02 20:48 +0000 [r375625-375658]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, channels/chan_misdn.c, main/ccss.c: Things don't
	  need to be that const.

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Multiple
	  revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
	  16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
	  primitives must be handled first. The frm->addr is a different
	  "address space" than the stack/instance address of other Lx
	  primitives. The test for B channel instance address could fail.
	  Patches: patch01_timers.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
	  2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
	  chan_misdn: Free memory in error paths and other memory leaks.
	  The one line commented with BUG is not easily fixable because
	  there is no de-init function one can call. Patches:
	  patch02_memory.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
	  16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
	  L2 de-establish/establish * An NT-PTMP cannot de/establish L2
	  since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
	  is finally active in handle_l1. * L2 deactivation logging
	  cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
	  as "UNKN". * Removed unused functions and code for L2 handling.
	  Patches: patch03_L2estab.diff (license #6372) patch uploaded by
	  Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
	  rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
	  lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
	  prim via lower_id layer (3 or 1) simply does not work. For TE (3)
	  it returns an error (len=-6) which is not evaluated by
	  handle_l1(), so the L1 layer status ends up wrong. Instead PH
	  must be sent via L4, only then does it reach L1 without an error
	  message. And NT PH prims only reach L1 when they are sent to
	  layer 2 id. --> use upper_id to send PH primitives. * Check for
	  errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
	  improved. * The lower_id is now not used for anything, except:
	  Why is lower_id layer deleted when it wasn't created? I removed
	  this code since it looks very wrong. Patches:
	  patch04_l1activation.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
	  2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
	  chan_misdn: Fix loss of B channels if L1 is down. If you make 2
	  calls out an NT PTMP port which is not connected to any phone,
	  the B channel associated with that call becomes unusable until
	  Asterisk is restarted. The problem is the EVENT_SETUP is queued
	  when L1 is not up in misdn_lib_send_event(). If L1 cannot be
	  activated the event won't be dequeued. It gets even worse when
	  the call is hung up. The queued EVENT_SETUP will be overwritten
	  by an EVENT_DISCONNECT. The reserved B channel then will never be
	  freed. If later someone connects a phone to the port, L1 will
	  eventually activate and the queued EVENT_DISCONNECT is sent down
	  the stack. However, it is ignored because it is the wrong call
	  state. The real fix would be that activation and queueing for a
	  new SETUP is done by the NT stack. But since it doesn't, the
	  workaround must be removed because it doesn't always work. Fix:
	  The event is no longer queued but immediately sent to the stack.
	  If L1 cannot be activated, the L3 state machine that was started
	  by the EVENT_SETUP will do its work, i.e. a timeout will release
	  the B channel properly. The SETUP possibly cannot be sent the
	  first time but is resent by T303 in case L1 could be activated.
	  Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
	  by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
	  rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
	  lines chan_misdn: Remove some calls to exit(). Try proper cleanup
	  when something goes wrong in misdn_lib_init(). Especially do not
	  call exit()! * Fix memory leak because stack_destroy() does not
	  free the stack struct. Patches: patch06_cleanup-init.diff
	  (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
	  ABE-2888 ........ Merged revisions 375519-375524 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

2012-11-02 16:58 +0000 [r375594]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
	  Origin Processing While looking at some debug logs, I noticed
	  that it was being reported that the SDP origin line was
	  unsupported or failed. Upon looking into this on my local
	  machine, I found that I too was getting this debug message yet
	  everything seemed to be getting processed properly. What was
	  discovered is, that, the variable to determine what is displayed
	  in the debug message for the SDP line that was processed, was not
	  being set for the origin line when the result was successful.
	  This patch fixes this and was tested on local machine.

2012-10-31 14:23 +0000 [r375528]  Matthew Jordan <mjordan@digium.com>

	* res/res_calendar_ews.c: Properly extract the Body information of
	  an EWS calendar item Unlike all other calendar modules,
	  res_calendar_ews fails to extract the Body information for a
	  calendar item. This is due, in part, to a quirk in the schema in
	  the XML - not only does a CalendarItem contain a Body element,
	  but the CalendarItem exists as a descendant of a different Body
	  element. The neon parser was erroneously skipping all Body
	  elements. This patch fixes that by bypassing Body elements that
	  are not a child of CalendarItem, and parsing the Body element out
	  if it is a child. Note that the original patch by Terry Wilson
	  only needed slight modifications to make it properly pull the
	  Body information out; as such, while I've linked to the patch
	  that I uploaded for Dmitry, I've attributed the patch to Terry.
	  (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
	  by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
	  uploaded by Terry Wilson (license 6283)

2012-10-30 18:48 +0000 [r375484]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c: mixmonitor: Add a test event This test
	  event is being used to fix the mixmonitor_audiohook_inherit test.

2012-10-30 02:07 +0000 [r375450]  Matthew Jordan <mjordan@digium.com>

	* apps/app_queue.c: Ensure that the Queue application tracks busy
	  members in off nominal situations There are a few code paths
	  where the Queue application fails to count a paused or in use
	  queue member as being 'busy'. This can cause callers to get stuck
	  in the Queue until a paused agent unpauses themselves. (closes
	  issue ASTERISK-20623) Reported by: Bryan Walters patches:
	  app_queue.patch uploaded by Bryan Walters (license 5851)

2012-10-29 21:01 +0000 [r375415]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Prevent resetting of NATted realtime peer
	  address on reload. If a "sip reload" is issued for a SIP peer,
	  then his IP address will be cleared, thus resulting in forgetting
	  the public IP address. Asterisk will then attempt to route SIP
	  traffic to the private IP address. The fix here is to make "sip
	  reload" ignore realtime peers when "host = dynamic" is spotted.
	  Realtime peers can now only have their IP address reset if they
	  have gone from being not dynamic to being dynamic. (closes issue
	  ASTERISK-18203) reported by daren ferreira (closes issue
	  ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
	  uploaded by JoshE (license #6075)

2012-10-29 19:26 +0000 [r375361-375388]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix the Park 'r' option when a channel parks
	  itself. When a channel uses the Park appliation to park itself
	  with the 'r' option, the channel hears music-on-hold instead of
	  the requested ringing. * Added a missing check for the 'r' option
	  when a channel parks itself. (closes issue ASTERISK-19382)
	  Reported by: James Stocks Patches by: dsessions Review:
	  https://reviewboard.asterisk.org/r/2148/

	* channels/chan_dahdi.c: chan_dahdi: Fix segfault dereferencing a
	  NULL tech_pvt. The tech support customer was using the AMI
	  Redirect action shortly after a call was placed. While the
	  channel tried to do an ast_read(), the masquerade resulting from
	  the channel redirect took place. The masquerade in the middle of
	  the ast_read() resulted in the segfault. (closes issue AST-1025)
	  Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
	  (license #5621) patch uploaded by rmudgett

2012-10-23 16:20 +0000 [r375272-375325]  Jonathan Rose <jrose@digium.com>

	* contrib/scripts/ast_tls_cert: ast_tls_cert script: Better
	  response for various exit conditions to openssl (closes issue
	  ASTERISK-20260) Reported by: Daniel O'Connor Patches:
	  ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
	  6419)

	* main/app.c: core: Fix a memory leak in app.c from an early return
	  ast_app_group_match_get_count allocates memory with the regcomp
	  function and we previously forgot to free it when bailing out due
	  to a regex compilation failure against category. (closes issue
	  AST-1018) Reported by: Guenther Kelleter Patches:
	  regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)

	* codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM
	  (closes issue ASTERISK-20457) Reported by: Richard Miller
	  Patches: code.patch uploaded by Richard Miller (license 5685)

2012-10-18 21:36 +0000 [r375216-375244]  Jonathan Rose <jrose@digium.com>

	* UPGRADE.txt: Correct version number in Upgrade.txt release notes
	  pertaining to queue order Showed 1.8.17 to 1.8.18, needs to be
	  1.8.18 to 1.8.19

	* UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds notes
	  describing behavioral changes to rrmemory strategy caused by
	  375216 (issue AST-989) Reported by: Thomas Arimont

	* apps/app_queue.c: app_queue: Make ordering of rrmemory/rrordered
	  persist over add/remove members Prior to this patch, adding,
	  removing or reloading members to rrmemory would cause the order
	  to become completely jumbled. Now it behaves more or less like
	  rrordered other than the fact that it stores the members on a
	  hash table rather than a linked list. This patch also prevents
	  removal of members and member reloads from jumbling rrordered
	  queues. (issue AST-989) Reported by: Thomas Arimont Review:
	  https://reviewboard.asterisk.org/r/2164/

2012-10-18 19:42 +0000 [r375189]  Richard Mudgett <rmudgett@digium.com>

	* makeopts.in, Makefile, build_tools/make_version, configure,
	  configure.ac: build_tools: Allow Asterisk to report git SHAs in
	  version string. Make git more attractive for managing
	  work-in-progress. Especially convenient when a potential patch
	  set needs to be tested on multiple platforms since one can use
	  git to keep all the test environments in sync independent of a
	  subversion server. Now the Asterisk version will show the exact
	  git SHA5 that was used when building (still appended by "M" if
	  there are local modifications) from a git clone of the Asterisk
	  repository so the developer can more easily know what is actually
	  under test. You will now get this: $ asterisk -V Asterisk
	  GIT-1698298 Instead of this: $ asterisk -V Asterisk
	  UNKNOWN__and_probably_unsupported This has zero impact for those
	  not using git with the exception of an extra test in the
	  configure script to gather git's path. This is necessary to
	  prevent "sudo make install" from failing since git may not be in
	  the path in make's shell environment. (closes issue
	  ASTERISK-20483) Reported by: Shaun Ruffell Patches:
	  0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
	  (license #5417) patch uploaded by Shaun Ruffell Modified

2012-10-17 18:55 +0000 [r375146]  Kinsey Moore <kmoore@digium.com>

	* main/tcptls.c: Ensure Asterisk fails TCP/TLS SIP calls when
	  certificate checking fails When placing a call to a TCP/TLS SIP
	  endpoint whose certificate is not signed by a configured CA
	  certificate, Asterisk would issue a warning and continue to
	  process the call as if there was not an issue with the
	  certificate. Asterisk now properly fails the call if the
	  certificate fails verification or if the certificate does not
	  exist when certificate checking is enabled (the default
	  behavior). (closes issue ASTERISK-20559) Reported by: kmoore
	  Review: https://reviewboard.asterisk.org/r/2163/

2012-10-16 21:41 +0000 [r375074-375111]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
	  Don't crash on large user input. Allow SIP headers without space.
	  Optimize code a bit. Review:
	  https://reviewboard.asterisk.org/r/2162

	* channels/chan_sip.c: Update sip_request_call SIP dial string
	  documentation. This was missed when merging review r1859.

2012-10-16 19:13 +0000 [r375059]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/autosupport: autosupport: fix bashism '==' is
	  bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
	  'case' works better there. (closes issue ASTERISK-20567) Reported
	  by: Tzafrir Cohen

2012-10-15 21:00 +0000 [r375025]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/strings.h, channels/chan_iax2.c,
	  apps/app_dial.c, main/ccss.c: Fix some potential misuses of
	  ast_str in the code. Passing an ast_str pointer by value that
	  then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or
	  ast_str_append_va() can result in the pointer originally passed
	  by value being invalidated if the ast_str had to be reallocated.
	  This fixes places in the code that do this. Only the example in
	  ccss.c could result in pointer invalidation though since the
	  other cases use a stack-allocated ast_str and cannot be
	  reallocated. I've also updated the doxygen in strings.h to
	  include notes about potential misuse of the functions mentioned
	  previously. Review: https://reviewboard.asterisk.org/r/2161

2012-10-14 08:59 +0000 [r374977]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* config.guess, config.sub: Update config.guess and config.sub:
	  2012-10-10 Update config.guess and config.sub to revision
	  fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
	  savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
	  64bit). config.guess:timestamp='2012-09-25'
	  config.sub:timestamp='2012-10-10'

2012-10-12 15:57 +0000 [r374905]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c: Do
	  not use a FILE handle when doing SIP TCP reads. This is used to
	  solve an issue where a poll on a file descriptor does not
	  necessarily correspond to the readiness of a FILE handle to be
	  read. This change makes it so that for TCP connections, we do a
	  recv() on the file descriptor instead. Because TCP does not
	  guarantee that an entire message or even just one single message
	  will arrive during a read, a loop has been introduced to ensure
	  that we only attempt to handle a single message at a time. The
	  tcptls_session_instance structure has also had an overflow buffer
	  added to it so that if more than one TCP message arrives in one
	  go, there is a place to throw the excess. Huge thanks goes out to
	  Walter Doekes for doing extensive review on this change and
	  finding edge cases where code could fail. (closes issue
	  ASTERISK-20212) reported by Phil Ciccone Review:
	  https://reviewboard.asterisk.org/r/2123

2012-10-11 15:42 +0000 [r374843]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c: Fix incorrect billing duration reported when batch
	  mode is enabled Similar to r369351, the billing duration can be
	  skewed when batch mode is enabled. This happened much more rarely
	  than the duration, as it only occured when the call was answered
	  (thereby indicating an actual answer time) and immediately hung
	  up on (indicating a billsec of 0). Since a billing time of '0'
	  can either mean that the call immediately ended or that the CDR
	  was improperly answered, we have to use additional information to
	  know whether or not we can trust the CDR billsec value. Prior to
	  this patch, we looked to see if we had a valid answer time. If we
	  did, and billsec was zero, we used the current time to calculate
	  what billsec value we could from the CDR being written. If batch
	  mode is enabled, this will incorrectly report a billsec value
	  being much greater than the actual duration of the call. Instead
	  of relying on the presence of an answer time to know whether or
	  not we can re-calculate the billsec for the CDR, we now also use
	  the presence of the CDR's end time to know if we need to
	  re-calculate or whether we can trust the billsec value that we
	  have. This prevents erroneous jumps in the billsec value, while
	  still making sure that in the worst case, some billing time will
	  be calculated. (closes issue AST-1016) Reported by: Thomas
	  Arimont Tested by: Thomas Arimont

2012-10-10 20:52 +0000 [r374686-374802]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: app_queue: Made pass connected line updates
	  from the caller to ringing queue members. Party A calls Party B
	  Party B puts Party A on hold. Party B calls a queue. Ringing
	  queue member D sees Party B identification. Party B transfers
	  Party A to the queue. Queue member D does not get a connected
	  line update for Party A. Queue member D answers the call and
	  still sees Party B information. However, if Party A later
	  transfers the call to Party C then queue member D gets a
	  connected line update for Party C. * Made pass connected line
	  updates from the caller to queue members while the queue members
	  are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
	  (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
	  rmudgett ........ Merged revisions 374801 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

	* main/pbx.c: Fix execution of 'i' extension due to uninitialized
	  variable. The fix for ASTERISK-18243 added code that could
	  potentially use dst_exten[] uninitialized. As a result the 'i'
	  exten may not be executed when it should. (closes issue
	  ASTERISK-20455) Reported by: Richard Miller Patches:
	  pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
	  Miller Made some cosmetic modifications.

	* configs/chan_dahdi.conf.sample: dahdi.conf.sample: Add
	  description for "buffers" setting. This contains an edited
	  version of the patch originally created by John Bigelow. (closes
	  issue ASTERISK-14435) Reported by: John Bigelow Patches:
	  buffers.patch (license #5091) patch uploaded by John Bigelow
	  0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
	  (license #5417) patch uploaded by Shaun Ruffell Modified

	* pbx/pbx_spool.c: Fix deletion of unopenable spool files. If
	  scan_service() cannot open the spool file, it logs a message
	  saying that it will delete the file and calls remove_from_queue()
	  to do it. However, remove_from_queue() fails to delete the spool
	  file because struct outgoing has not yet been fully initialized.
	  * Merged allocating a new struct outgoing and init_outgoing()
	  into new_outgoing(). Allocation is initialization. * Made
	  apply_outgoing() not initialize the spool filename in struct
	  outgoing. * Made apply_outgoing() call ast_trim_blanks() and
	  ast_skip_blanks() rather than manually inlining them. * Reduced
	  indentation levels in apply_outgoing(). * Fixed a garbled comment
	  in remove_from_queue(). * Reworked scan_service() to simplify it.
	  (closes issue ASTERISK-17231) Reported by: David Chappell
	  Patches: spool_open_failure.diff (license #4997) patch uploaded
	  by David Chappell Started with this patch.

2012-11-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.18.0 Released.

2012-10-08  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.18.0-rc1 Released.

2012-10-05 20:20 +0000 [r374570-374581]  dlee <dlee@localhost>:

	* main/manager.c: I've committed too much. Reverting part of
	  r374570.

	* main/manager.c: Improve AMI long line error handling In AMI's
	  parser, when it receives a long line (> 1024 characters), it
	  discards that line, but continues to process the message
	  normally. Typically, this is not a problem because a) who has
	  lines that long and b) usually a discarded line results in an
	  invalid message. But if that line is specifying an optional
	  field, then the message will be processed, you get a 'Response:
	  Success', but things don't work the way you expected them to.
	  This patch changes the behavior when a line-too-long parse error
	  occurs. * Changes the log message to avoid way-too-long (and
	  truncated anyways) log messages * Adds a 'parsing' status flag to
	  Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
	  well, a line is too long * Responds with an appropriate error if
	  parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
	  Bigelow Review: https://reviewboard.asterisk.org/r/2142/

2012-10-05 18:20 +0000 [r374536]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c: Merged
	  revisions 374515-374535 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
	  (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
	  Made setup_bc() static. Patches: patch1_unused-code.diff (license
	  #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
	  ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
	  (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
	  states Patches: patch2_unused-states.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
	  | 16 lines chan_misdn: Remove unnecessary null pointer checks and
	  checks for stack->nt * cleanup_bc() is always called with valid
	  bc (or it would've crashed before). * Value of stack->nt is known
	  in advance at some places. * Rename handle_event() to
	  handle_event_te(), handle_frm() to handle_frm_te(). Patches:
	  patch3_checks.diff (license #6372) patch uploaded by Guenther
	  Kelleter Modified JIRA ABE-2882 ................ r374518 |
	  rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Fix spelling in log messages Patches:
	  patch4_spelling.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
	  2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
	  chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
	  calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
	  emptied, cleaned and set not in use, although
	  misdn_lib_send_event() already did the same. This is bad. When
	  it's not in use we are not allowed to touch it. * Moved log
	  message in front of the resulting actions and fixed it to match
	  the case. Patches: patch5_bccleanup.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
	  | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
	  etc., really bad stuff. * Fix return codes of cb_events() for
	  EVENT_SETUP to use caller's cleanup mechanisms. * Move
	  cl_queue_chan() call after bearer check. Patches:
	  patch6_leaks.diff (license #6372) patch uploaded by Guenther
	  Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
	  2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
	  chan_misdn: We must initialize cause on sending a DISCONNECT. We
	  must initialize cause on sending a DISCONNECT, so it is later
	  correctly indicated to ast_channel in case the answer
	  (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
	  patch7_hangupcause.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2882 ................ r374522 |
	  rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Remove unused code for upqueue Patches:
	  patch8_unused-upqueue.diff (license #6372) patch uploaded by
	  Guenther Kelleter JIRA ABE-2882 ................ r374523 |
	  rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
	  chan_misdn: Improve debugging (port number, messages fixed, dups
	  removed) Patches: patch9_debug.diff (license #6372) patch
	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
	  r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
	  | 8 lines chan_misdn: Better debug: we can print_bc_info even if
	  there's no ast leg. Patches: patch10_debug-bc-2.diff (license
	  #6372) patch uploaded by Guenther Kelleter Modified. JIRA
	  ABE-2882 ................ r374534 | rmudgett | 2012-10-05
	  12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
	  setup_bc() is called too early for an incoming SETUP on TE. This
	  prevents the B channel from being setup for HDLC mode when
	  requested by the bearer capability and config option hdlc=yes. It
	  violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
	  connect to the channel until a CONNECT ACKNOWLEDGE message has
	  been received." * Call setup_bc() on receipt of
	  CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
	  PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
	  Guenther Kelleter Modified. JIRA ABE-2881 ................
	  r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
	  | 2 lines chan_misdn: Remove some more deadcode. ................

2012-10-04 20:15 +0000 [r374475-374479]  Alec L Davis <sivad.a@paradise.net.nz>

	* CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
	  Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
	  a recompile, allow values to be adjusted in dsp.conf For binary
	  distributions allows easy adjustment for wobbly GSM calls, and
	  other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
	  DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
	  alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2144/

	* main/dsp.c: dsp.c fix incorrect DTMF Digit_Duration. it's always
	  short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2
	  (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license
	  585) Review https://reviewboard.asterisk.org/r/2145/

2012-10-04 17:39 +0000 [r374456]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a regression from direct media ACLs
	  where the directrtpsetup option no longer works. A check was
	  added for direct media ACLs that immediately forbid remote
	  bridging if there was no bridged channel. This caused
	  directrtpsetup to no longer function as it needs this information
	  before bridging actually occurs. Logic has now been adjusted so
	  if there is no bridged channel a remote bridge will still be
	  attempted. (closes issue ASTERISK-20511) Reported by: kristoff
	  Review: https://reviewboard.asterisk.org/r/2146/

2012-10-04 15:25 +0000 [r374426]  dlee <dlee@localhost>:

	* main/db.c, res/res_agi.c: Fix DBDelTree error codes for AMI, CLI
	  and AGI The AMI DBDelTree command will return Success/Key tree
	  deleted successfully even if the given key does not exist. The
	  CLI command 'database deltree' had a similar problem, but was
	  saved because it actually responded with '0 database entries
	  removed'. AGI had a slightly different error, where it would
	  return success if the database was unavailable. This came from
	  confusion about the ast_db_deltree retval, which is -1 in the
	  event of a database error, or number of entries deleted
	  (including 0 for deleting nothing). * Adds a Doxygen comment to
	  process_db_keys explaining its retval * Changed some poorly named
	  res variables to num_deleted * Specified specific errors when
	  calling ast_db_deltree (database unavailable vs. entry not found
	  vs. success) * Fixed similar bug in AGI database deltree, where
	  'Database unavailable' results in successful result (closes issue
	  AST-967) Reported by: John Bigelow Review:
	  https://reviewboard.asterisk.org/r/2138/

2012-10-04 04:39 +0000 [r374365-374384]  Alec L Davis <sivad.a@paradise.net.nz>

	* CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
	  configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
	  Asterisk's DTMF Specifications are based on AT&T specs, which may
	  not be compatible in other countries. Various countries have
	  different specifications for the maximum power level differences
	  between the DTMF low group and high group of frequencies. Power
	  level difference between frequencies for different
	  Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
	  8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
	  = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
	  (2006-03) Now allow 4 variables to be individually configured in
	  dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
	  specifications Add's the following variables to dsp.conf
	  ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
	  ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
	  (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
	  tbsky,alecdavis alecdavis (license 585) Review
	  https://reviewboard.asterisk.org/r/2141/

	* main/dsp.c: _dsp_init: bring inline with trunk preparation for
	  clean merge of DTMF TWIST patch No functional changes, just
	  style. alecdavis (license 585) Reported by: Alec Davis Tested by:
	  alecdavis related https://reviewboard.asterisk.org/r/2141

2012-10-04 02:09 +0000 [r374177-374335]  Matthew Jordan <mjordan@digium.com>

	* res/res_jabber.c: Check for presence of buddy in info/dinfo
	  handlers The res_jabber resource module uses the ASTOBJ library
	  for managing its ref counted objects. After calling
	  ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
	  the object has to be checked to see if the buddy existed. Prior
	  to this patch, the buddy object was not checked for NULL; with
	  this patch in both aji_client_info_handler and aji_dinfo_handler
	  the pointer is checked before used and, if no buddy object was
	  found, the handlers return an error code. This patch does not
	  take the approach that our JID can be used to log in from another
	  resource. If that approach is desired, an improvement could be
	  made to this patch to create the buddy on the fly. This patch
	  seeks only to prevent Asterisk from crashing. Note that multiple
	  people have proposed patches for this issue; the patch being
	  committed here is based on those. (closes issue ASTERISK-19532)
	  Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
	  fix-jabber uploaded by Karsten Wemheuer (license #5930)
	  xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
	  (license #6157) (closes issue ASTERISK-19557) Reported by:
	  ulugutz

	* main/ccss.c: Destroy the generic_monitors container after the
	  core_instances in ccss For each item in core_instances disposed
	  of in the shutdown of ccss, any generic monitor instances
	  referenced by the objects will be removed from generic_monitors
	  during their destruction. Hilarity ensues if generic_monitors no
	  longer exists. Thanks to the Asterisk Test Suite's generic_ccss
	  test for complaining loudly when it ran into this.

	* main/asterisk.c: Ensure Shutdown AMI event is still fired during
	  Asterisk shutdown Richard pointed out that having the manager
	  dispose of itself gracefully during shutdown meant that the
	  Shutdown event will no longer get fired. This patch moves the AMI
	  event just prior to running the atexit callbacks.

	* main/event.c, main/taskprocessor.c, res/res_musiconhold.c,
	  main/cel.c, main/indications.c, main/channel.c, main/data.c,
	  main/pbx.c, main/manager.c, main/ccss.c, main/features.c: Fix a
	  variety of ref counting issues This patch resolves a number of
	  ref leaks that occur primarily on Asterisk shutdown. It adds a
	  variety of shutdown routines to core portions of Asterisk such
	  that they can reclaim resources allocate duringd initialization.
	  Review: https://reviewboard.asterisk.org/r/2137

2012-10-01 16:45 +0000 [r374108]  Sean Bright <sean@malleable.com>

	* tests/test_db.c, apps/app_queue.c, main/db.c,
	  include/asterisk/astdb.h: app_queue: Support persisting and
	  loading of long member lists. Greenlight in #asterisk brought up
	  that he was receiving an error message "Could not create
	  persistent member string, out of space" when running app_queue in
	  Asterisk 10. dump_queue_members() made an assumption that 8K
	  would be enough to store the generated string, but with queues
	  that have large member lists this is not always the case. This
	  patch removes the limitation and uses ast_str instead of a fixed
	  sized buffer. The complicating factor comes from the fact that
	  ast_db_get requires a buffer and buffer size argument, which
	  doesn't let us pull back more than what we pass in, so I
	  introduced a new ast_db_get_allocated() which returns an
	  ast_strdup()'d copy of the value from astdb. As an aside, I did
	  some testing on the maximum size of data that we can store in the
	  BDB library we distribute and was able to store a 10MB string and
	  retrieve it with no problems, so I feel this is a safe patch.
	  Review: https://reviewboard.asterisk.org/r/2136/

2012-09-28 19:03 +0000 [r374032]  Jonathan Rose <jrose@digium.com>

	* res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
	  The opinion of development was that it is both improper to have
	  Matt's personal email address used in the source and that the
	  command wouldn't be useful without it. (closes issue AST-467)
	  Reported by: Malcolm Davenport

2012-09-28 12:14 +0000 [r373989]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Update documentation to make it explicit that
	  "stream file" will not restart musiconhold. (issue
	  ASTERISK-17367) Reported by: oej

2012-09-27 22:08 +0000 [r373945]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
	  leak using channel name parameter. The SendDTMF channel name
	  parameter has two issues. 1) Crashes if the channel name does not
	  exist. 2) Leaks a channel reference if the channel is the current
	  channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
	  documentation. * Renamed app to senddtmf_name and tweaked the
	  type.

2012-09-27 16:49 +0000 [r373878-373909]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: loader: Ensure dependent modules are properly
	  initialized. If an Asterisk module specifies a dependency in
	  ast_module_info.nonoptreq, it is possible for Asterisk to skip
	  calling the modules's .load function. Asterisk was loading and
	  linking the module via load_dynamic_module() but was not adding
	  the module to the resource_heap. Therefore the module was not
	  initialized based on it's priority along with the other modules
	  in the heap. Now use load_resource() instead of
	  load_dynamic_module() for non-optional requirement. This will add
	  the module to the resource_heap so the module can be properly
	  initialized in the correct order. This is required if there are
	  any module global data structures initialized in the .load()
	  callback for the module on platforms which do not support weak
	  references. (issue ASTERISK-20439) Reported by: sruffell Patches:
	  0001-loader-Ensure-dependent-modules-are-properly-initial.patch
	  uploaded by sruffell (license 5417)

	* channels/chan_local.c: Fix an issue where Local channels dialed
	  by app_queue are considered in use immediately. The chan_local
	  channel driver returns a device state of in use even if a created
	  Local channel has not yet been dialed. This fix changes the logic
	  to return a state of not in use until the channel itself has been
	  dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach
	  Review: https://reviewboard.asterisk.org/r/2116/

2012-09-26 21:11 +0000 [r373848]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Move handling of 408 response so there is no
	  misleading warning message. (closes issue ASTERISK-20060)
	  Reported by: Walter Doekes

2012-09-26 18:04 +0000 [r373815]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_meetme.c: Fixed meetme tab completion and command
	  documentation. * Removed unnecessary case sensitivity in meetme
	  list, lock, unlock, mute, unmute, and kick commands. * Separated
	  meetme lock/unlock, mute/unmute, and kick commands into their own
	  registered commands to simplify tab completion and parameter
	  checking. meetme_lock_cmd(), meetme_mute_cmd(), and
	  meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
	  AST-1006) Reported by: John Bigelow Tested by: rmudgett

2012-09-25 23:07 +0000 [r373735-373773]  Mark Michelson <mmichelson@digium.com>

	* main/say.c: Fix saying of date in Dutch. The Dutch say the date
	  before the month. (closes issue ASTERISK-20353) Reported by: Teun
	  Ouwehand

	* configs/agents.conf.sample, channels/chan_agent.c: Remove dead
	  code and documentation for nonexistent feature. multiplelogin was
	  removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
	  was removed. (closes issue AST-948) reported by Steve Pitts

	* apps/app_voicemail.c: Fix error where improper IMAP greetings
	  would be deleted. (closes issue ASTERISK-20435) Reported by:
	  fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
	  uploaded by Michael L. Young (License #5026) (with suggested
	  modification made by me)

2012-09-25 20:10 +0000 [r373705]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Fix T.38 support when used with chan_local
	  in between. Users of the T.38 API can indicate
	  AST_T38_REQUEST_PARMS on a channel to request that the channel
	  indicate a T.38 negotiation with the parameters present on the
	  channel. The return value of this indication is expected to be
	  AST_T38_REQUEST_PARMS upon success but with chan_local involved
	  this could never occur. This fix changes chan_local to always
	  return AST_T38_REQUEST_PARMS for this situation. If the
	  underlying channel technology on the other side does not support
	  T.38 this would have been determined ahead of time using
	  ast_channel_get_t38_state and an indication would not occur.
	  (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
	  ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
	  https://reviewboard.asterisk.org/r/2070/

2012-09-25 19:32 +0000 [r373666-373702]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c: Fix an issue where media would not flow
	  for situations where the legacy STUN code is in use. The STUN
	  packets should *not* be blocked by strict RTP. (closes issue
	  ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
	  Colp (trunk r369817)

	* apps/app_queue.c: "show" completion option for "queue" shouldn't
	  appear twice When tab-completing CLI commands starting with
	  "queue", "show" appeared twice in the list due to the way that
	  Asterisk's tab completion functions and the order in which the
	  commands were registered. The registration order has been altered
	  to resolve this issue. (closes issue AST-940) Reported-by: Steve
	  Pitts

2012-09-25 17:21 +0000 [r373652]  Terry Wilson <twilson@digium.com>

	* configs/sip.conf.sample, channels/sip/include/sip.h,
	  channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
	  session timers The SIP session timer mechanism contains a
	  mandatory 'refresher' parameter (included in the Session-Expires
	  header) which is used in the session timer offer/answer signaling
	  within a SIP Invite dialog. It looks like asterisk is
	  interpreting the uac resp. uas role only as the initial role of
	  client and server (caller is uac, callee is uas). The standard
	  rfc 4028 however assigns the client role to the ((RE)-Invite)
	  requester, the server role to the ((RE)-Invite) responder. This
	  patch has Asterisk track the actual refresher as "us" or "them"
	  as opposed to relying on just the configured "uas" or "uac"
	  properties. (closes issue AST-922) Reported by: Thomas Airmont
	  Review: https://reviewboard.asterisk.org/r/2118/

2012-09-25 17:18 +0000 [r373618-373640]  Richard Mudgett <rmudgett@digium.com>

	* codecs/ilbc/iLBC_decode.c, codecs/ilbc/iLBC_encode.c: Fix
	  valgrind found memcpy issues in codec_ilbc. Valgrind found
	  codec_ilbc using memcpy instead of memmove for overlapping memory
	  blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
	  Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
	  #5674) patch uploaded by Walter Doekes

	* codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if the
	  respective sources change.

2012-09-25 16:15 +0000 [r373617]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Set Quality of Service for video
	  rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog
	  Patches: chan_sip.c.diff uploaded by ddkprog (license 6008)

2012-09-25 13:27 +0000 [r373578]  Kinsey Moore <kmoore@digium.com>

	* configs/res_odbc.conf.sample: Fix documentation for default
	  username in res_odbc This was previously stated to be "root", but
	  is actually the name of the context if unspecified. (closes issue
	  ASTERISK-20258) Reported by: Stefan x

2012-09-25 11:58 +0000 [r373532-373550]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_multicast.c: Fix an issue where a caller to ast_write
	  on a MulticastRTP channel would determine it failed when in
	  reality it did not. When sending RTP packets via multicast the
	  amount of data sent is stored in a variable and returned from the
	  write function. This is incorrect as any non-zero value returned
	  is considered a failure while a return value of 0 is success. For
	  callers (such as ast_streamfile) that checked the return value
	  they would have considered it a failure when in reality nothing
	  went wrong and it was actually a success. The write function for
	  the multicast RTP engine now returns -1 on failure and 0 on
	  success, as it should. (closes issue ASTERISK-17254) Reported by:
	  wybecom

	* channels/chan_sip.c: Add missing checks that I neglected. The SIP
	  technology and SIP info technology should be considered equal.
	  (closes issue ASTERISK-20409) Reported by: michele cicciotti
	  privatewave

2012-09-24 22:15 +0000 [r373504]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c: Revert change to res_rtp_asterisk
	  committed in r373236 (1.8) The change committed in r373236
	  attempted to account for endpoints that increased their RTP
	  timestamp in DTMF end of event re-transmissions. This change
	  attempted to make Asterisk continue to work with endpoints that
	  failed to follow the RFC while maintaining the fix that allowed
	  for out of order DTMF to be handled. Unfortunately, there is no
	  free lunch, and this patch broke any system that sent DTMF
	  immediately after an RTP session was established or when an SSRC
	  is updated. As such, that patch is being reverted for the
	  previous behavior. Endpoints that erroneously increase the RTP
	  timestamp in DTMF end of event packets will not work properly
	  with Asterisk. (issue ASTERISK-20424)

2012-09-24 22:09 +0000 [r373500]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Be consistent, send From: "Anonymous"
	  <sip:anonymous@anonymous.invalid> When setting
	  CALLERID(pres)=unavailable in the dialplan, the From header in
	  the SIP message contains "Anonymous"
	  <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
	  should use a lowercase a in the userpart of the URI. * Make the
	  From header use a lowercase A in the userpart of the anonymous
	  URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
	  Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
	  patch uploaded by Antti Yrjola

2012-09-24 20:57 +0000 [r373467]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c, funcs/func_audiohookinherit.c:
	  func_audiohookinherit: Document some missed sources. This patch
	  also mentions that AUDIOHOOK_INHERIT can be used to transfer
	  MixMonitor audiohooks. There is also wiki that addresses
	  audiohooks and the use of AUDIOHOOK_INHERIT at the following
	  link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
	  (closes issue ASTERISK-18220) Reported by: Ishfaq Malik

2012-09-24 19:15 +0000 [r373438]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a deadlock caused by a race condition
	  between removing a hint and reloading the dialplan and
	  subscribing to the removed hint. If conditions were right it was
	  possible for both the PBX core and chan_sip to deadlock by both
	  having a lock that the other wants. In the case of the PBX core
	  it had the contexts lock and wanted a SIP dialog lock, while in
	  the case of chan_sip it had the SIP dialog lock and wanted the
	  contexts lock. This fix unlocks the SIP dialog before getting the
	  extension state so that the other thread will not block on trying
	  to lock it. Once the extension state is retrieved the SIP dialog
	  is locked again and life carries on. As the SIP dialog is
	  reference counted it is not possible for it to go away after
	  unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins

2012-09-24 15:40 +0000 [r373424]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix potential reentrancy problems in
	  chan_sip. Asterisk v1.8 and later was not as vulnerable to this
	  issue. * Made find_call() lock each private as it processes the
	  found dialogs. (Primary cause of ABE-2876) * Made the other
	  functions that traverse the dialogs container lock each private
	  as it examines them. * Fix race condition in sip_call() if the
	  thread that sent the INVITE is held up long enough for a response
	  to be processed. The p->initid for the INVITE retransmission
	  could be added after it was canceled by the response processing.
	  * Made __sip_destroy() clean up resource pointers after freeing.
	  This is primarily defensive in case someone has a stale private
	  pointer. * Removed redundant memset() in reqprep(). The call to
	  init_req() already does the memset() and is the first reference
	  to req in reqprep(). * Removed useless set of req.method in
	  transmit_invite(). The calls to initreqprep() and reqprep() have
	  to do this because they memset() the req. JIRA ABE-2876
	  .......... Merged -r373423 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

2012-09-21 19:00 +0000 [r373298-373342]  Jonathan Rose <jrose@digium.com>

	* channels/iax2-provision.c: iax2-provision: Fix improper return on
	  failed cache retrieval (closes issue ASTERISK-20337) reported by:
	  John Covert Patches: iax2-provision.c.patch uploaded by John
	  Covert (license 5512)

	* apps/app_queue.c: app_queue: Make queue reload members and
	  variants of that work Prior to this patch, 'queue reload members'
	  cli command did not work at all. This also affects the manager
	  function 'QueueReload' when supplied with the 'members: yes'
	  field. (closes issue AST-956) Reported by: John Bigelow

2012-09-20 19:12 +0000 [r373242]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Fix incorrect MeetME conference bridge
	  reference count decrementing and sometimes premature destruction.
	  When using the 'e' or 'E' option to MeetMe the configured
	  conference bridges are loaded and examined to see if any are
	  empty. If no conference bridges are empty the caller is prompted
	  to enter the number of one. This operation left around a pointer
	  to the last created conference bridge still containing
	  participants. When the caller that was not able to find any empty
	  conference bridge hung up this pointer was disposed of and the
	  reference count of the conference bridge decremented. If there
	  was only a single participant in the conference bridge it was
	  ultimately destroyed prematurely. (closes issue AST-994) Reported
	  by: John Bigelow

2012-09-20 18:41 +0000 [r373236]  Matthew Jordan <mjordan@digium.com>

	* res/res_rtp_asterisk.c: When processing RFC 2833 DTMF, accomodate
	  increasing timestamps in End events While endpoints should not be
	  changing the source timestamp between DTMF event packets, the
	  fact is there exists those endpoints that do exactly that. To
	  work around this, we absorb timestamps within the expected
	  re-transmit period. Note that this period only affects End of
	  Event packets, so it should not prevent the detection of new DTMF
	  digits that happen to arrive right on top of each other. (closes
	  issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by:
	  mjordan, Vladimir Mikhelson Review:
	  https://reviewboard.asterisk.org/r/2124

2012-09-19 16:02 +0000 [r373165]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a regression where direct media was not
	  permitted for calls using SIP INFO DTMF. A change was committed
	  to fix direct media ACL support. This change wrongly assumed that
	  only a single channel technology structure exists for chan_sip.
	  This is in fact false as a second exists for calls using SIP INFO
	  DTMF. The code which performs direct media ACL checking now
	  checks for both the non-INFO DTMF and INFO DTMF channel
	  technology structures. (closes issue ASTERISK-20409) Reported by:
	  michele cicciotti privatewave

2012-09-18 20:12 +0000 [r373131]  Sean Bright <sean@malleable.com>

	* main/manager.c: Don't crash when passing a NULL message to
	  __astman_get_header. Before this commit, __astman_get_header
	  would blindly dereference the passed in 'struct message *' to
	  traverse the header list. There are cases, however, such as
	  '*CLI> sip qualify peer foo' where the message pointer is NULL,
	  so we need to check for that.

2012-09-15 00:13 +0000 [r373090]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c: Made companding law for SS7 calls only
	  determined by SS7 signaling type. For SS7, the companding law for
	  a call was chosen inconsistently depending upon ss7type (ITU vs
	  ANSI) and the DAHDI companding default (T1 vs E1). For incoming
	  calls, the companding law was determined by ss7type. For outgoing
	  calls, the companding law was determined by the DAHDI default.
	  With the wrong combination you would get A-law/u-law conflicts.
	  An A-law/u-law conflict sounds like bad static on the line. SS7
	  ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
	  noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
	  with T1 line: ok * Fix the companding law used to be determined
	  by the SS7 signaling type only.

2012-09-14 19:07 +0000 [r373061]  Matthew Jordan <mjordan@digium.com>

	* main/ssl.c, main/tcptls.c, channels/chan_sip.c: Resolve memory
	  leaks in TLS initialization and TLS client connections This patch
	  resolves two sources of memory leaks when using TLS in Asterisk:
	  1) It removes improper initialization (and multiple
	  re-initializations) of portions of the SSL library. Asterisk
	  calls SSL_library_init and SSL_load_error_strings during SSL
	  initialization; collectively this obviates the need for calling
	  any of the following during initialization or client connection
	  handling: * ERR_load_crypto_strings (handled by
	  SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
	  SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
	  SSL_library_init) 2) Failure to completely clean up all memory
	  allocated by Asterisk and by the SSL library for TLS clients.
	  This included not freeing the SSL_CTX object in the SIP channel
	  driver, as well as not clearing the error stack when the TLS
	  client exited. Note that these memory leaks were found by Thomas
	  Arimont, and this patch was essentially written by him with some
	  minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
	  Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
	  Arimont (license 5525) Review:
	  https://reviewboard.asterisk.org/r/2105

2012-09-13 18:39 +0000 [r373024]  dlee <dlee@localhost>:

	* include/asterisk/channel.h, main/channel.c: Fix timeouts for
	  ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
	  its timeout to ast_waitfor_nandfds, expecting it to decrement the
	  timeout by however many milliseconds were waited. This is a
	  problem if it consistently waits less than 1ms. The timeout will
	  never be decremented, and we wait... FOREVER! This patch makes
	  ast_waitfordigit_full manage the timeout itself. It maintains the
	  previously undocumented behavior that negative timeouts wait
	  forever. (closes issue ASTERISK-20375) Reported by: Mark
	  Michelson Tested by: Mark Michelson Review:
	  https://reviewboard.asterisk.org/r/2109/

2012-09-13  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.17.0-rc1 Released.

2012-09-12 15:42 +0000 [r372959]  Matthew Jordan <mjordan@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h: Constify
	  __ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain
	  files - most notably ccss.c - the 'tag' parameter passed to
	  __ao2_ref_debug will be a const char *. The function currently
	  expects that parameter to not be const. This causes a warning
	  when compiling, as the const qualifier is being discarded. With
	  dev-mode enabled, this prevents compiling Asterisk. This patch
	  makes __ao2_ref_debug's tag and file parameters const. (closes
	  issue ASTERISK-20408) Reported by: mjordan

2012-09-12 14:51 +0000 [r372932]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add channel name to a warning to make
	  debugging easier. The "autodestruct with owner in place" message
	  is typically indicative of a channel reference leak. Printing out
	  the name of the channel in the message may be helpful when trying
	  to debug the issue.

2012-09-11 22:11 +0000 [r372902]  Jonathan Rose <jrose@digium.com>

	* channels/chan_local.c: chan_local: Switch from using a random 4
	  digit hex identifier to unique id Changes chan_local channels to
	  use an 8 digit hex identifier generated atomically and
	  sequentially in order to eliminate the chance of having multiple
	  channels with the same name during high call volume situations.
	  (issue ASTERISK-20318) Reported by: Dan Cropp Review:
	  https://reviewboard.asterisk.org/r/2104/

2012-09-11 15:26 +0000 [r372840]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Fix bad channel application data reference. When
	  channels get bridged due to an AMI bridge action or a DTMF
	  attended transfer, the two channels that get bridged have their
	  application data pointing to the other channel's name. This means
	  that if one channel is hung up but the other moves on, it means
	  that the channel that moves on will have its application data
	  pointing at freed memory. (issue ASTERISK-20335) Reported by:
	  aragon

2012-09-10 20:53 +0000 [r372804]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_iax2.c: Ensure iax2 debug output is displayed when
	  expected When IAX2 debug was changed from iax_showframe to
	  iax_outputframe, some instances were missed (or added afterward).
	  This was causing debug output to not be displayed when expected.
	  (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
	  John Covert

2012-09-10 18:35 +0000 [r372765]  Jonathan Rose <jrose@digium.com>

	* apps/app_meetme.c: app_meetme: Document that 'p' option will
	  continue in dialplan. (closes issue AST-991) Reported by John
	  Bigelow

2012-09-10 18:31 +0000 [r372763]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Warn on CLI when UDPTL init fails This adds
	  a CLI warning when a SDP offer is rejected due to UDPTL
	  initialization failure. Previously, there was no indication of
	  the reason for offer rejection in this case. (closes issue
	  ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi

2012-09-10 17:07 +0000 [r372736]  Jonathan Rose <jrose@digium.com>

	* main/channel.c: Masquerade: Retain parkinglot settings made by
	  CHANNEL function. Prior to this patch, the user would have a
	  parkinglot set on a channel that was parked and when the channel
	  was retrieved, any attempt by that channel to park would simply
	  use the default. This patch makes parkinglot values set in this
	  way be retained through the masquerade. (closes issue AST-990)
	  Reported by: Nick Huskinson Patches:
	  masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
	  (license 6182)

2012-09-09 01:19 +0000 [r372709]  Matthew Jordan <mjordan@digium.com>

	* channels/sip/sdp_crypto.c: Only re-create an SRTP session when
	  needed; respond with correct crypto policy In r356604, SRTP
	  handling was fixed to accomodate multiple crypto keys in an SDP
	  offer and the ability to re-create an SRTP session when the
	  crypto keys changed. In certain circumstances - most notably when
	  a phone is put on hold after having been bridged for a
	  significant amount of time - the act of re-creating the SRTP
	  session causes problems for certain models of phones. The patch
	  committed in r356604 always re-created the SRTP session
	  regardless of whether or not the cryptographic keys changed.
	  Since this is technically not necessary, this patch modifies the
	  behavior to only re-create the SRTP session if Asterisk detects
	  that the remote key has changed. This allows models of phones
	  that do not handle the SRTP session changing to continue to work,
	  while also providing the behavior needed for those phones that do
	  re-negotiate cryptographic keys. In addition, in Asterisk 1.8
	  only, it was found that phones that offer AES_CM_128_HMAC_SHA1_32
	  will end up with no audio if the phone is the initiator of the
	  call. The phone will send an INVITE request specifying that
	  AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy;
	  Asterisk will set its policy to that value. Unfortunately, when
	  the call is Answered and a 200 OK is sent back to the UA, the
	  policy sent in the response's SDP will be the hard coded value
	  AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk
	  using the INVITE request's policy of AES_CM_128_HMAC_SHA1_32,
	  while the phone uses Asterisk's response of
	  AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think
	  the other is crazy. This patch fixes that by caching the policy
	  from the request and responding with it. Note that this is not a
	  problem in Asterisk 10 and later, as the ability to configure the
	  policy was added in that version. (issue ASTERISK-20194) Reported
	  by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
	  https://reviewboard.asterisk.org/r/2099

2012-09-08 03:54 +0000 [r372682]  dlee <dlee@localhost>:

	* main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
	  tcptls.c. Without this flag, those files will compile with the
	  system installed OpenSSL headers (if they exist). This is a real
	  bummer if a different path was specified using --with-ssl=
	  (closes issue ASTERISK-20392)

2012-09-07 23:05 +0000 [r372620-372655]  Richard Mudgett <rmudgett@digium.com>

	* main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). (closes
	  issue ASTERISK-20349) Reported by: Brent Eagles

	* funcs/func_math.c: Remove annoying unconditional debug message
	  from INC/DEC functions. (closes issue AST-1001) Reported by:
	  Guenther Kelleter

	* apps/app_queue.c: Fix exception path typo in app_queue.c
	  try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
	  Pepper Patches: fix-local-channel-locking.patch (license #6350)
	  patch uploaded by Jeremy Pepper

	* apps/app_voicemail.c: Fix VoicemailUserEntry event headers
	  ServerEmail and MailCommand reported values. The AMI action
	  VoicemailUsersList VoicemailUserEntry event headers ServerEmail
	  and MailCommand did not report the global values if they were not
	  overridden. The VoicemailUserEntry event header ServerEmail was
	  not populated with the global value if the voicemail user did not
	  override it. The VoicemailUserEntry event header MailCommand was
	  never populated with a value. * Removed unused struct ast_vm_user
	  member mailcmd[]. (closes issue AST-973) Reported by: John
	  Bigelow Tested by: rmudgett

2012-09-07 02:24 +0000 [r372554-372581]  Matthew Jordan <mjordan@digium.com>

	* apps/app_minivm.c: Free ast_str objects when temp file fails to
	  be created in MiniVM The previous commit (r372554) was from a
	  patch that was written before r366880, which ensured that ast_str
	  objects allocated in the sendmail routine were free'd in off
	  nominal paths. This commit frees the string objects in the off
	  nominal path introduced in r372554. (issue ASTERISK-17133)
	  Reported by: Tzafrir Cohen

	* apps/app_minivm.c: Fix file descriptor leak and pointer scope
	  issue in MiniVM when sending mail When MiniVM sends an e-mail and
	  it has the volgain option set, it will spawn sox in a separate
	  process to handle the manipulation of the sound file. In doing
	  so, it creates a temporary file. There are two problems here: 1)
	  The file descriptor returned from mkstemp is leaked 2) The
	  finalfilename character pointer points to a buffer that loses
	  scope once volgain processing is finished. Note that in r316265,
	  Russell fixed some gcc warnings by using the return value of the
	  mkstemp call. A warning was placed in minivm that the file
	  descriptor was going to be leaked. This patch reverts that
	  change, as it handles the leak and 'uses' the file descriptor
	  returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
	  Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
	  Cohen (license #5035)

2012-09-06 21:38 +0000 [r372517]  Kinsey Moore <kmoore@digium.com>

	* apps/app_queue.c: Ensure listed queues are not offered for
	  completion When using tab-completion for the list of queues on
	  "queue reset stats" or "queue reload
	  {all|members|parameters|rules}", the tab-completion listing for
	  further queues erroneously listed queues that had already been
	  added to the list. The tab-completion listing now only displays
	  queues that are not already in the list. (closes issue AST-963)
	  Reported-by: John Bigelow

2012-09-06 18:54 +0000 [r372498]  dsessions <dsessions@localhost>:

	* configs/res_ldap.conf.sample, channels/chan_sip.c: LDAP Realtime
	  Peers Cannot Register Prior to 1.8, it was not necessary for an
	  explicit "type" to be set for an asterisk LDAP realtime peer. Now
	  the routine find_peer actually checks the type field during
	  registration and fails to find the peer if it is not set. The
	  attached patches make the realtime type equal whatever type is
	  being searched for if the type is 0 upon return from routine
	  build_peer. (closes issue ASTERISK-17222) Reported by: John
	  Covert Patch by: David Vossel Tested by: Darren Sessions Review:
	  https://reviewboard.asterisk.org/r/2095/

2012-09-06 15:52 +0000 [r372471]  Jonathan Rose <jrose@digium.com>

	* UPGRADE.txt: chan_sip: Note change in behavior to how
	  directmediapermit/deny ACL works r366547 introduced a change to
	  the directmedia ACL for chan_sip which modified the behavior
	  significantly. Prior to the patch, this option would bridge peers
	  with directmedia if a peer's IP address matched its own
	  directmedia ACL. After that patch, the peer would check the
	  bridged peer's ACL instead. This change has been present since
	  1.8.14.0. That patched failed to document the change in
	  Upgrade.txt, so this patch adds mention of that change to
	  UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)

2012-09-06 14:28 +0000 [r372444]  Kinsey Moore <kmoore@digium.com>

	* apps/app_queue.c: Ensure "rules" is tab-completable for "queue
	  show" Previously, tabbing at the end of "queue show" produced a
	  list of available queues about which information could be shown,
	  but did not include an alternative command, "rules", to access
	  information about queue rules. The "rules" item should now be
	  shown in the list of tab-completable items. (closes issue
	  AST-958) Reported-by: John Bigelow

2012-09-06 02:48 +0000 [r372390-372417]  Matthew Jordan <mjordan@digium.com>

	* pbx/pbx_dundi.c: Fix DUNDi message routing bug when neighboring
	  peer is unreachable Consider a scenario where DUNDi peer PBX1 has
	  two peers that are its neighbors, PBX2 and PBX3, and where PBX2
	  and PBX3 are also neighbors. If the connection is temporarily
	  broken between PBX1 and PBX3, PBX1 should not include PBX3 in the
	  list of peers it sends to PBX2 in a DPDISCOVER message, as it
	  cannot send messages to PBX3. If it does, PBX2 will assume that
	  PBX3 already received the message and fail to forward the message
	  on to PBX3 itself. This patch fixes this by only including peers
	  in a DPDISCOVER message that are reachable by the sending node.
	  This includes all peers with an empty address (00:00:00:00:00:00)
	  and that are have been reached by a qualify message. This patch
	  also prevents attempting to qualify a dynamic peer with an empty
	  address until that peer registers. (closes issue ASTERISK-19309)
	  Reported by: Peter Racz patches: dundi_routing.patch uploaded by
	  Peter Racz (license 6290) The patch uploaded by Peter was
	  modified slightly for this commit.

	* apps/app_followme.c: Allow configured numbers for FollowMe to be
	  greater than 90 characters When parsing a 'number' defined in
	  followme.conf, FollowMe previously parsed the number in the
	  configuration file into a buffer with a length of 90 characters.
	  This can artificially limit some parallel dial scenarios. This
	  patch allows for numbers of any length to be defined in the
	  configuration file. Note that Clod Patry originally wrote a patch
	  to fix this problem and received a Ship It! on the JIRA issue.
	  The patch originally expanded the buffer to 256 characters.
	  Instead, the patch being committed duplicates the string in the
	  config file on the stack before parsing it for consumption by the
	  application. (closes issue ASTERISK-16879) Reported by: Clod
	  Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
	  by Clod Patry (license #5138) Slightly modified for this commit.

2012-09-05 19:20 +0000 [r372354]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c: Correct documentation for ModuleLoad AMI action
	  The documentation incorrectly listed 'rtp' as a reloadable
	  subsystem and left out many other reloadable subsystems. It is
	  now also documented that subsystems may only be reloaded, not
	  loaded or unloaded. (closes issue AST-977) Reported-by: John
	  Bigelow

2012-09-05 18:34 +0000 [r372339]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c: dsp.c: in ast_mf_detect_init incorrectly sets
	  goertzel samples to 160, should be MF_GSIZE Related
	  https://reviewboard.asterisk.org/r/2097/

2012-09-05 18:29 +0000 [r372337]  Kinsey Moore <kmoore@digium.com>

	* main/pbx.c: Ensure counts generated in
	  manager_show_dialplan_helper are correct When
	  manager_show_dialplan_helper was written, the counter increment
	  for the total number of contexts was placed with the extensions
	  increment instead of in the enclosing loop. This function should
	  now generate correct context counts. (closes issue AST-970)
	  Reported-by: John Bigelow

2012-09-05 13:13 +0000 [r372268]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Fix memory leaks in app_voicemail when
	  using IMAP storage or realtime config This patch fixes two memory
	  leaks: 1. When find_user is called with NULL as its first
	  parameter, the voicemail user returned is allocated on the heap.
	  The inboxcount2 function uses find_user in such a fashion when
	  counting new messages, and fails to free the resulting voicemail
	  user object. 2. When populate_defaults is called on a voicemail
	  user, it wipes whatever flags have been set on the object by
	  copying over the global flags object. If the VM_ALLOCED flag was
	  ste on the voicemail user prior to doing so, that flag is
	  removed. This leaks the voicemail user when free_user is later
	  called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
	  patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
	  Patch slightly modified for this commit. Review:
	  https://reviewboard.asterisk.org/r/2096

2012-09-05 07:35 +0000 [r372212-372239]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c: dsp.c: Fix multiple issues when no-interdigit delay
	  is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss detector
	  to original -r349249 method with some changes, remove
	  unnecessary; 1. reseting of hits=0, when no signal, only need to
	  set it once. 2. incrementing of hits, when the hit is the same as
	  the current hit. 3. setting of lasthit, when it's the same as
	  before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
	  spelling mistakes (closes issue ASTERISK-19610) alecdavis
	  (license 585) Reported by: Jean-Philippe Lord Tested by:
	  alecdavis Review: https://reviewboard.asterisk.org/r/2085/

	* main/dsp.c: dsp.c: optimize goerztzel sample loops, in
	  dtmf_detect, mf_detect and tone_detect use a temporary short int
	  when repeatedly used to call goertzel_sample. alecdavis (license
	  585) Reported by: alecdavis Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/2093/

2012-09-05 03:45 +0000 [r372185]  Michael L. Young <elgueromexicano@gmail.com>

	* res/res_rtp_asterisk.c: Fix Incrementing Sequence Number For
	  Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
	  place to increment the sequence number for retransmitted DTMF end
	  packets. With the introduction of the RTP engine API in 1.8, the
	  sequence number was no longer being incremented. This patch fixes
	  this regression as well as cleans up a few lines that were not
	  doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
	  Bansal Tested by: Michael L. Young Patches:
	  01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
	  6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/2083/

2012-09-05 02:16 +0000 [r372158]  Matthew Jordan <mjordan@digium.com>

	* cel/cel_pgsql.c: Fix memory leak when CEL is successfully written
	  to PostgreSQL database PQClear is not called when the result
	  object of a call to PQExec has a status of PGRES_COMMAND_OK.
	  Interestingly enough, the off nominal case was handled properly,
	  so this memory leak only occurred when CEL records were
	  successfully written. This patch properly clears the result in
	  the nominal code path. (closes issue ASTERISK-19991) Reported by:
	  Etienne Lessard Tested by: Etienne Lessard patches:
	  mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
	  #6394)

2012-08-30 20:51 +0000 [r372048-372089]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Prevent crash on shutdown due to refcount error
	  on queues container. When app_queue is unloaded, the queues
	  container has its refcount decremented, potentially to 0. Then
	  the taskprocessor responsible for handling device state changes
	  is unreferenced. If the taskprocessor happens to be just about to
	  run its task, then it will create and destroy an iterator on the
	  queues container. This can cause the refcount on the queues
	  container to increase to 1 and then back to 0. Going back to 0 a
	  second time results in double frees. This failure was seen
	  periodically in the testsuite when Asterisk would shut down.

	* apps/app_queue.c: Help prevent ringing queue members from being
	  rung when ringinuse set to no. Queue member status would not
	  always get updated properly when the member was called, thus
	  resulting in the member getting multiple calls. With this change,
	  we update the member's status at the time of calling, and we also
	  check to make sure the member is still available to take the call
	  before placing an outbound call. (closes issue ASTERISK-16115)
	  reported by nik600 Patches: app_queue.c-svn-r370418.patch
	  uploaded by Italo Rossi (license #6409)

2012-08-30 16:21 +0000 [r371961-372015]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
	  ignored during calls by some IAX2 peers When an IAX2 call is made
	  using the credentials of a peer defined in a dynamic Asterisk
	  Realtime Architecture (ARA) backend, the ACL rules for that peer
	  are not applied to the call attempt. This allows for a remote
	  attacker who is aware of a peer's credentials to bypass the ACL
	  rules set for that peer. This patch ensures that the ACLs are
	  applied for all peers, regardless of their storage mechanism.
	  (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
	  mjordan, Alan Frisch

	* main/manager.c, README-SERIOUSLY.bestpractices.txt: AST-2012-012:
	  Resolve AMI User Unauthorized Shell Access through ExternalIVR
	  The AMI Originate action can allow a remote user to specify
	  information that can be used to execute shell commands on the
	  system hosting Asterisk. This can result in an unwanted
	  escalation of permissions, as the Originate action, which
	  requires the "originate" class authorization, can be used to
	  perform actions that would typically require the "system" class
	  authorization. Previous attempts to prevent this permission
	  escalation (AST-2011-006, AST-2012-004) have sought to do so by
	  inspecting the names of applications and functions passed in with
	  the Originate action and, if those applications/functions matched
	  a predefined set of values, rejecting the command if the user
	  lacked the "system" class authorization. As noted by IBM X-Force
	  Research, the "ExternalIVR" application is not listed in the
	  predefined set of values. The solution for this particular
	  vulnerability is to include the "ExternalIVR" application in the
	  set of defined applications/functions that require "system" class
	  authorization. Unfortunately, the approach of inspecting fields
	  in the Originate action against known applications/functions has
	  a significant flaw. The predefined set of values can be bypassed
	  by creative use of the Originate action or by certain dialplan
	  configurations, which is beyond the ability of Asterisk to
	  analyze at run-time. Attempting to work around these scenarios
	  would result in severely restricting the applications or
	  functions and prevent their usage for legitimate means. As such,
	  any additional security vulnerabilities, where an
	  application/function that would normally require the "system"
	  class authorization can be executed by users with the "originate"
	  class authorization, will not be addressed. Instead, the
	  README-SERIOUSLY.bestpractices.txt file has been updated to
	  reflect that the AMI Originate action can result in commands
	  requiring the "system" class authorization to be executed. Proper
	  system configuration can limit the impact of such scenarios.
	  (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
	  X-Force Research

	* doc/CODING-GUIDELINES (added): Restore CODING-GUIDELINES to doc
	  folder In r294740, the CODING-GUIDELINES was removed from the doc
	  folder in favor of the content on the Asterisk wiki. Some folks
	  still look in the doc folder initially for coding guideline
	  suggestions; as such, this patch adds a CODING-GUIDELINES file
	  back into the doc folder. The content of the file merely points
	  to the correct page on the Asterisk wiki where the coding
	  guidelines currently live. (closes issue ASTERISK-20279) Reported
	  by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
	  Andrew Latham (license 5985)

2012-08-29 20:42 +0000 [r371919]  Jonathan Rose <jrose@digium.com>

	* apps/app_meetme.c: app_meetme: Adding test events for following
	  activity in MeetMe.

2012-08-29 19:38 +0000 [r371860-371888]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Initialize file descriptors for dummy channels to
	  -1. Dummy channels usually aren't read from, but functions like
	  SHELL and CURL use autoservice on the channel. (closes issue
	  ASTERISK-20283) Reported by: Gareth Palmer Patches:
	  svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
	  (modified)

	* apps/app_dial.c: Fix hangup cause passthrough regression. The
	  v1.8 -r369258 change to fix the F and F(x) action logic
	  introduced a regression in passing the hangup cause from the
	  called channel to the caller channel. (closes issue
	  ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
	  app_dial_hangupcause.patch (license #6421) patch uploaded by
	  Konstantin Suvorov (modified) Tested by: rmudgett

2012-08-29 16:59 +0000 [r371824]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
	  instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
	  Doekes

2012-08-27 21:47 +0000 [r371747-371787]  Mark Michelson <mmichelson@digium.com>

	* configs/agents.conf.sample: Fix misleading documentation in
	  agents.conf.sample regarding ackcall usage. The documentation
	  made it sound as if the DTMF acknowledgment was needed at the
	  time the agent logs in, rather than when the agent is called.
	  This is likely a relic from the days when there were multiple
	  ways of logging in agents. (closes issue AST-962) reported by
	  Steve Pitts

	* main/manager.c: Fix incorrect documentation of the MailboxStatus
	  manager command. The "Waiting" field was misdocumented as
	  reporting the number of messages waiting. In reality, it simply
	  indicated the presence or absence of waiting messages. (closes
	  issue AST-975) reported by John Bigelow

	* configs/queues.conf.sample: Fix incorrectly documented option in
	  queues.conf sharedlastcall defaults to "no" not "yes" (closes
	  issue AST-979) reported by Steve Pitts

2012-08-27 16:40 +0000 [r371718]  dlee <dlee@localhost>:

	* main/lock.c: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
	  variants. The original implementations simply wrap pthread
	  functions, which take absolute time as an argument. The spinlock
	  version for systems without those functions treated the argument
	  as a delta. This patch fixes the spinlock version to be
	  consistent with the pthread version. (closes issue
	  ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
	  uploaded by Egor Gorlin (license 6416)

2012-08-27 13:43 +0000 [r371690]  Kinsey Moore <kmoore@digium.com>

	* main/utils.c: Implement workaround for BETTER_BACKTRACES crash
	  When compiling with BETTER_BACKTRACES enabled, Asterisk will
	  sometimes crash when "core show locks" is run. This happens
	  regularly in the testsuite since several tests run "core show
	  locks" to help with debugging. This seems to be a fault with
	  libraries on certain operating systems (notably CentOS 6.2/6.3)
	  running on virtual machines and utilizing gcc 4.4.6. (closes
	  issue ASTERISK-20090)

2012-08-26 23:03 +0000 [r371662]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of
	  MF_GSIZE

2012-08-21 20:35 +0000 [r371590]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
	  res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
	  res/res_config_sqlite.c, cdr/cdr_tds.c, main/xmldoc.c,
	  apps/app_dial.c, channels/chan_dahdi.c, channels/chan_sip.c,
	  funcs/func_odbc.c, main/file.c: Fix misuses of asprintf
	  throughout the code. This fixes three main issues * Change
	  asprintf() uses to ast_asprintf() so that it pairs properly with
	  ast_free() and no longer causes MALLOC_DEBUG to freak out. * When
	  ast_asprintf() fails, set the pointer NULL if it will be
	  referenced later. * Fix some memory leaks that were spotted while
	  taking care of the first two points. (Closes issue
	  ASTERISK-20135) reported by Richard Mudgett Review:
	  https://reviewboard.asterisk.org/r/2071

2012-08-20 15:25 +0000 [r371544]  Kinsey Moore <kmoore@digium.com>

	* main/udptl.c: Ignore recovered zero-length secondary UDPTL
	  packets In some cases, recovering lost packets using the
	  secondary packet recovery mechanism with UDPTL/T.38 can result in
	  the recovery of zero-length packets. These must be ignored or the
	  frame generated from them can cause segfaults and allocation
	  failures. (closes issue ASTERISK-19762) (closes issue
	  ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
	  Gagnon (rgagnon)

2012-08-17 18:51 +0000 [r371469]  Matthew Jordan <mjordan@digium.com>

	* main/xmldoc.c: Fix memory leak in XML documentation When
	  formatting documentation fields, the XML documentation parser
	  calls xmldoc_get_formatted. This function allocates a string
	  buffer at the beginning of its routine. Unfortunately, on certain
	  code paths, it also calls xmldoc_string_cleanup, which assumes
	  that it will create the string buffer. The previously allocated
	  string buffer is then leaked by the xmldoc_string_cleanup
	  routine. Now: we don't do that. (closes issue AST-932) Reported
	  by: Alexander Homig

2012-08-17 15:49 +0000 [r371393-371436]  Kinsey Moore <kmoore@digium.com>

	* main/loader.c: Add instrumentation to subsystem reloads When
	  Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
	  generate TestEvent AMI events on subsystem reloads such as cdr,
	  dnsmgr, extconfig, etc. (issue PQ-1126)

	* main/loader.c: Add module reload instrumentation for
	  TEST_FRAMEWORK This adds AMI events for module reloads when
	  Asterisk is built with TEST_FRAMEWORK enabled and corrects
	  generation of the module load AMI event. (issue PQ-1126)

2012-08-16 22:30 +0000 [r371392]  Terry Wilson <twilson@digium.com>

	* main/config.c: Handle integer over/under-flow in ast_parse_args
	  The strtol family of functions will return *_MIN/*_MAX on
	  overflow. To detect when an overflow has happened, errno must be
	  set to 0 before calling the function, then checked afterward.
	  (closes issue ASTERISK-20120) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/2073/

2012-08-16 18:57 +0000 [r371337-371357]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable to
	  set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag
	  was used instead, which will frequently flip during reinvites.
	  (closes issue AST-897) Reported by: Thomas Arimont

	* channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP answer
	  is included in the SIP ACK Under certain conditions, a SIP
	  transaction involving directmedia wouldn't trigger a re-invite
	  because the SDP answer was included in an ACK instead of in a
	  message that we would have triggered the invite with. This patch
	  just queues a source change control frame if the dialog is using
	  directmedia when we find sdp for an ACK. (closes issue AST-913)
	  Reported by: Thomas Arimont

2012-08-15 23:10 +0000 [r371306]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix bug where final queue member would not be
	  removed from memory. If a static queue had realtime members, then
	  there could be a potential for those realtime members not to be
	  properly deleted from memory. If the queue's members were loaded
	  from realtime and then all the members were deleted from the
	  backend, then the queue would still think these members existed.
	  The reason was that there was a short- circuit in code such that
	  if there were no members found in the backend, then the queue
	  would not be updated to reflect this. Note that this only
	  affected static queues with realtime members. Realtime queues
	  with realtime members were unaffected by this issue. (closes
	  issue ASTERISK-19793) reported by Marcus Haas

2012-08-15 20:14 +0000 [r371270]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
	  relatedpeer on SIP dialog destruction The other instance of this
	  bug was fixed by jcolp/file in r121496. If we are destroying a
	  dialog only set the MWI dialog pointer on the related peer to
	  NULL if it is the dialog currently being destroyed. (closes issue
	  ASTERISK-20119) Patch-by: Misha Vodsedalek

2012-08-13 20:00 +0000 [r371201]  Kinsey Moore <kmoore@digium.com>

	* main/loader.c, apps/app_meetme.c: Add test instrumentation This
	  adds test instrumentation for loading and unloading of modules
	  and for certain actions in MeetMe to be used in the testsuite or
	  any other consumer of AMI events. These will only be generated
	  when Asterisk is built with TEST_FRAMEWORK enabled. (issue
	  PQ-1131) (issue PQ-1133)

2012-08-13 19:49 +0000 [r371198]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix problem where incorrect pointer was
	  checked for nullity.

2012-08-10 21:21 +0000 [r371141]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a couple of documentation problems in
	  app_queue.c * The RemoveQueueMember app made mention of options
	  that could be passed in, but no options are supported. I have
	  removed the listing of options from the documentation. * The
	  RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
	  that could be set. (closes issue AST-949) reported by Steve Pitts
	  (closes issue AST-954) reported by Steve Pitts

2012-08-10 16:40 +0000 [r371060-371089]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: remove ALREADYGONE flag on ooh323 call data
	  by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
	  there really. This indication arrive from asterisk core not h.323
	  stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
	  Patches: ASTERISK-19308.patch

	* addons/ooh323c/src/ooGkClient.c: Send re-register packets by GRQ
	  (gatekeeper request) interval (close issue ASTERISK-20094)
	  Patches: ASTERISK-20094-2.patch

2012-08-09 18:58 +0000 [r371012]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
	  configure, include/asterisk/autoconfig.h.in, configure.ac: Use
	  better libss7 detection test and move libpri compile test.

2012-08-09 18:58 +0000 [r370988-371011]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c: Fix to resend GRQ/RRQ if RRJ
	  (registration reject) is received (close issue ASTERISK-20094)
	  Patches: ASTERISK-20094.patch

	* addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with
	  append mode instead of overwrite

2012-08-09 17:39 +0000 [r370985]  Kinsey Moore <kmoore@digium.com>

	* apps/app_meetme.c: Correct documentation for the MeetMe x flag
	  The documentation for the x flag for MeetMe incorrectly described
	  its function as closing down the conference when the last marked
	  user left. It actually causes the users with that flag to leave
	  the conference when the last marked user exits. The functionality
	  of this flag is not changing.

2012-08-08 22:40 +0000 [r370952]  Michael L. Young <elgueromexicano@gmail.com>

	* apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a
	  channel hangs up while being spied upon and the option to exit
	  the ChanSpy application when the spied on channel hangs up is
	  set, ast_autochan_destroy is not being called and therefore a
	  reference to the spied upon channel is not removed. The symptom
	  being reported was that when using func_group in the dialplan and
	  calling "group show channels" at the cli, the spied upon channel
	  was still being shown while "core show channels" showed that the
	  channel was not up. This patch calls ast_autochan_destroy when a
	  spied upon channel hangs up and the option to exit the ChanSpy
	  application is set, removing the reference to the channel
	  allowing the count for the group that the spied channel was part
	  of to be decremented. (closes issue ASTERISK-17515) Reported by:
	  Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
	  Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
	  L. Young (license 5026)

2012-08-08 20:28 +0000 [r370923]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c: Do not define a cause that doesn't actually exist
	  AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no
	  cause information. As such, it should not be defined and
	  translatable as a cause.

2012-08-08 19:58 +0000 [r370900]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Fix the analog dial *0 flash-hook of
	  bridged peer feature. The flash-hook the bridged peer feature now
	  correctly determines if the bridged peer is another chan_dahdi
	  channel, that it is an analog channel, and that it has the
	  correct signaling for an FXO port. It now also flash-hooks the
	  correct channel.

2012-08-07 19:19 +0000 [r370856]  Kinsey Moore <kmoore@digium.com>

	* main/channel.c: Add missing AST_CAUSE_* -> text translations

2012-08-06 15:00 +0000 [r370797]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Improve debug message for temporary outbound
	  proxies. Thanks to Paul Belanger for pointing this out.

2012-08-03 21:43 +0000 [r370769-370771]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/config_parser.c: Seriously? Another compilation
	  error fixed. Somebody beat me.

	* channels/chan_sip.c: Remove unused variable.

	* channels/sip/config_parser.c, channels/sip/include/sip.h,
	  channels/chan_sip.c: Fix error in the "IPorHost" section of a SIP
	  dialstring. This is based on the review request posted by Walter
	  Doekes (referenced lower in the commit message) The main fix here
	  is to treat the IPorHost portion of the dial string as a
	  temporary outbound proxy. This ensures requests get sent to the
	  proper location. Due to the age of the request, some parts were
	  no longer relevant. For instance, the request moved outbound
	  proxy parsing code into a single method. This is done in a
	  previous commit, so it was not necessary to do again. Also, the
	  review request fixed some errors with regards to request routing
	  for CANCEL and ACK requests. This has also been fixed in more
	  recent commits. (closes issue ASTERISK-19677) reported by Walter
	  Doekes Review https://reviewboard.asterisk.org/r/1859

2012-08-01 02:25 +0000 [r370697]  Kinsey Moore <kmoore@digium.com>

	* utils/extconf.c: Revert alloca changes for utils These changes
	  were a tad overzealous in the utils directory. Unfortunately,
	  these don't compile with a "make".

2012-07-31 20:54 +0000 [r370666]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Schedule pokes of registered SIP peers
	  within a given timespan after SIP reload With a large number of
	  SIP peers registered, performing a SIP reload causes a flood of
	  SIP OPTIONS request packets. These are immediately sent out, and,
	  as responses come back, can cause peers to be flagged as 'lagged'
	  due to handling of the many response messages. This fix prevents
	  this "packet storm" and schedules the pokes for a random time.
	  That time varies between 1 ms and the peer's qualify time, or, if
	  the qualify time is unknown, the global qualifyfreq setting. The
	  committed patch has some very small modifications to the patch
	  schmidts wrote for the review. (closes issue ASTERISK-19154)
	  Reported by: Nicolo Mazzon patches: issue19154.patch license
	  #6034 uploaded by schmidts Review:
	  https://reviewboard.asterisk.org/r/1652

2012-07-31 19:31 +0000 [r370642]  Kinsey Moore <kmoore@digium.com>

	* main/utils.c, funcs/func_logic.c, channels/chan_gtalk.c,
	  cdr/cdr_pgsql.c, channels/chan_iax2.c, res/res_jabber.c,
	  main/config.c, main/channel.c, res/ael/pval.c,
	  apps/app_osplookup.c, main/manager.c, pbx/pbx_spool.c,
	  main/strcompat.c, apps/app_minivm.c, main/features.c,
	  res/res_agi.c, main/http.c, main/logger.c, pbx/pbx_ael.c,
	  main/app.c, channels/chan_alsa.c, pbx/pbx_realtime.c,
	  addons/chan_mobile.c, apps/app_while.c, include/asterisk/utils.h,
	  main/pbx.c, res/res_config_pgsql.c, channels/chan_sip.c,
	  apps/app_festival.c, pbx/pbx_lua.c, funcs/func_cut.c,
	  tests/test_linkedlists.c, apps/app_getcpeid.c,
	  funcs/func_global.c, channels/chan_jingle.c, main/tcptls.c,
	  funcs/func_channel.c, apps/app_directed_pickup.c,
	  main/callerid.c, main/file.c, apps/app_macro.c, main/astmm.c,
	  apps/app_sms.c, main/event.c, pbx/pbx_dundi.c,
	  include/asterisk/strings.h, utils/extconf.c,
	  apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c,
	  addons/res_config_mysql.c, apps/app_voicemail.c,
	  addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
	  main/say.c, main/threadstorage.c, funcs/func_strings.c: Clean up
	  and ensure proper usage of alloca() This replaces all calls to
	  alloca() with ast_alloca() which calls gcc's __builtin_alloca()
	  to avoid BSD semantics and removes all NULL checks on memory
	  allocated via ast_alloca() and ast_strdupa(). (closes issue
	  ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/
	  Patch-by: Walter Doekes (wdoekes)

2012-07-31 15:26 +0000 [r370618]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample, channels/sip/include/sip.h,
	  channels/chan_sip.c: Help mitigate potential reinvite glare
	  scenarios. When Asterisk servers are set up back-to-back, and
	  direct media is to be used betweeen endpoints, it is fairly
	  common for the two Asterisk servers to send direct media
	  reinvites to each other simultaneously. This results in 491s and
	  ACKs being exchanged between the servers. While the media
	  eventually gets set up properly, the problem is that there can be
	  a noticeable delay for the streams to stabilize. This patch adds
	  a new directmedia option called "outgoing". With this set, an
	  immediate direct media reinvite will only be sent if the call
	  direction is outgoing. For incoming dialogs, an immediate direct
	  media reinvite will not be sent, but further "reactionary" direct
	  media reinvites may be sent. For those who are having some deja
	  vu, that's because this patch was originally committed to trunk
	  since there is a new configuration option added. After seeing a
	  bug report about audio being slow to set up on SIP calls, it
	  became apparent that this patch would be the best solution for
	  resolving the issue. The patch is unintrusive and will have no
	  effect unless the option is explicitly enabled. (closes issue
	  AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
	  reported by Matt Jordan

2012-09-13  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.16.0 Released.

2012-09-11  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.16.0-rc2 Released.

	* AST-2012-013: Resolve ACL rules being ignored during calls by some
	  IAX2 peers

	* AST-2012-012: Resolve AMI User Unauthorized Shell Access through
	  ExternalIVR

	* r371860: Fix hangup cause passthrough regression.

	  The v1.8 -r369258 change to fix the F and F(x) action logic
	  introduced a regression in passing the hangup cause from the called
	  channel to the caller channel.

	  (closes issue ASTERISK-20287)
	  Reported by: Konstantin Suvorov
	  Patches:
	      app_dial_hangupcause.patch (license #6421) patch uploaded by
	      Konstantin Suvorov (modified)
	      Tested by: rmudgett

	* r372709: Only re-create an SRTP session when needed; respond with
	  correct crypto policy

	  In r356604, SRTP handling was fixed to accomodate multiple crypto
	  keys in an SDP offer and the ability to re-create an SRTP session
	  when the crypto keys changed. In certain circumstances - most
	  notably when a phone is put on hold after having been bridged for a
	  significant amount of time - the act of re-creating the SRTP session
	  causes problems for certain models of	phones.	The patch committed in
	  r356604 always re-created the SRTP session regardless	of whether or
	  not the cryptographic keys changed. Since this is technically
	  not necessary, this patch modifies the behavior to only re-create the
	  SRTP session if Asterisk detects that the remote key has changed.
	  This allows models of phones that do not handle the SRTP session
	  changing to continue to work, while also providing the behavior
	  needed for those phones that do re-negotiate cryptographic keys.

	  In addition, in Asterisk 1.8 only, it was found that phones that
	  offer	AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone
	  is the initiator of the call. The phone will send an INVITE request
	  specifying that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic
	  policy; Asterisk will set its policy to that value. Unfortunately,
	  when the call is Answered and a 200 OK is sent back to the UA, the
	  policy sent in the response's SDP will be the hard coded value
	  AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk using
	  the INVITE request's policy of AES_CM_128_HMAC_SHA1_32, while the
	  phone uses Asterisk's response of AES_CM_128_HMAC_SHA1_80. Hilarity
	  ensues as both endpoints think the other is crazy.

	  This patch fixes that by caching the policy from the request and
	  responding with it. Note that this is not a problem in Asterisk 10
	  and later, as	the ability to configure the policy was added in that
	  version.

	  (issue ASTERISK-20194)
	  Reported by: Nicolo Mazzon
	  Tested by: Nicolo Mazzon

	  Review: https://reviewboard.asterisk.org/r/2099

	* r372840: Fix bad channel application data reference.

	  When channels get bridged due to an AMI bridge action
	  or a DTMF attended transfer, the two channels that
	  get bridged have their application data pointing to
	  the other channel's name. This means that if one channel
	  is hung up but the other moves on, it means that the
	  channel that moves on will have its application data
	  pointing at freed memory.

	  (issue ASTERISK-20335)

2012-07-31  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.16.0-rc1 Released.

2012-07-30 16:47 +0000 [r370563]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Release B channel allocation on error path
	  in chan_misdn.

2012-07-25 21:00 +0000 [r370494]  Jonathan Rose <jrose@digium.com>

	* res/res_agi.c: res_agi: Add message indicating need for \n
	  character in verbose message The while loop responsible for
	  reading AGI messages from a fastAGI service can end up looping
	  indefinitely when an AGI script fails to indicate the end of a
	  message with a \n character. This patch adds an indication that
	  we are expecting a \n character to end the message to make it
	  more clear to users that this is necessary if they are receiving
	  this warning over and over. (issue ASTERISK-20061) Reported by:
	  Eike Kuiper

2012-07-24 16:53 +0000 [r370429]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: Rewrite a comment that didn't adequately explain
	  the code it was documenting.

2012-07-24 16:49 +0000 [r370428]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_oss.c: chan_oss: fix "sample rate" error message

2012-07-23 21:09 +0000 [r370360-370383]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs/func_shell.c: Improve documentation for the SHELL()
	  dialplan function.

	* main/channel.c: Free any datastores attached to dummy channels.
	  Revision 370205 added the use of a datastore attached to a dummy
	  channel to resolve a memory leak, but
	  ast_dummy_channel_destructor() in this branch did not free
	  datastores, resulting in a continued (but slightly smaller)
	  memory leak. This patch backports the change to free said
	  datastores from the Asterisk trunk. (related to issue AST-916)

2012-07-19 22:07 +0000 [r370275]  Richard Mudgett <rmudgett@digium.com>

	* main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems
	  casting away constness.

2012-07-19 22:00 +0000 [r370252-370273]  Matthew Jordan <mjordan@digium.com>

	* main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled To
	  fix a memory leak in CEL, a channel datastore was introduced
	  whose destruction function pointer was pointed to the ast_free
	  macro. Without MALLOC_DEBUG enabled this compiles as fine, as
	  ast_free is defined as free. With MALLOC_DEBUG enabled, however,
	  ast_free takes on a definition from a different place then
	  utils.h, and became undefined. This patch resolves this by using
	  a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
	  calls ast_free; when MALLOC_DEBUG is not enabled, this is defined
	  to be ast_free, which is defined to be free. (issue AST-916)
	  Reported by: Thomas Arimont

	* res/res_rtp_asterisk.c: Handle extremely out of order RFC 2833
	  DTMF The current implementation of RFC 2833 DTMF handling in
	  res_rtp_asterisk will, if a packet arrives out of order, drop the
	  packet. This is to prevent duplicate ton generation in the
	  Asterisk core. Since the RTP layer does not buffer data itself,
	  this is the only option the RTP layer currently has for handling
	  packets that arrive out of order. For the most part, this doesn't
	  matter. For a particular digit, so long as a BEGIN packet arrives
	  before the first END packet, the digit will be produced. If
	  subsequent BEGIN packets arrive interleaved with the ENDs, they
	  will be dropped; likewise, if the BEGIN or END packets themselves
	  are out of order, those packets are dropped but sufficient
	  information is conveyed to the Asterisk core to produce the
	  appropriate digit. For certain sequences of DTMF packets - most
	  notably when, for a particular digit, an END packet arrives
	  before any BEGIN packet for that digit - this is a real problem.
	  When an END arrives before any BEGINs, the END packet is dropped
	  - but at the same time, it causes subsequent BEGIN packets for
	  that digit to be ignored. When the next in order END packet
	  arrives, it too is dropped - Asterisk believes that there was no
	  initial BEGIN. The solution this patch provides is to trust the
	  END packet to convey the information needed for the Asterisk core
	  to produce the DTMF digit. If we receive an END packet, and it: *
	  Has a timestamp greater then the last timestamp received from an
	  END packet * Does not have the same sequence number as the last
	  received sequence number (and is thus not an END packet
	  retransmission) Then we send the END frame up to the Asterisk
	  core. It contains enough DTMF information for Asterisk to produce
	  the digit. On the other hand, if we receive a BEGIN or
	  continuation packet that occurs with a timestamp equal to or less
	  then the last END timestamp, then we've received something out of
	  order - but we already have received enough information to
	  produce the digit. These packets are dropped. Much thanks goes to
	  Olle Johansson (oej) for providing the idea for this solution.
	  Review: https://reviewboard.asterisk.org/r/2033/ (issue
	  ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
	  Jordan

2012-07-18 19:12 +0000 [r370183-370205]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cel.c: Resolve severe memory leak in CEL logging modules. A
	  customer reported a significant memory leak using Asterisk 1.8.
	  They have tracked it down to
	  ast_cel_fabricate_channel_from_event() in main/cel.c, which is
	  called by both in-tree CEL logging modules (cel_custom.c and
	  cel_sqlite3_custom.c) for each and every CEL event that they log.
	  The cause was an incorrect assumption about how data attached to
	  an ast_channel would be handled when the channel is destroyed;
	  the data is now stored in a datastore attached to the channel,
	  which is destroyed along with the channel at the proper time.
	  (closes issue AST-916) Review:
	  https://reviewboard.asterisk.org/r/2053/

	* apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c,
	  apps/app_stack.c, funcs/func_global.c, res/res_odbc.c,
	  main/channel.c, addons/app_mysql.c, main/pbx.c,
	  funcs/func_curl.c, main/ccss.c, funcs/func_odbc.c,
	  funcs/func_lock.c: Ensure that all ast_datastore_info structures
	  are 'const'. While addressing a bug, I came across a instance of
	  'struct ast_datastore_info' that was not declared 'const'. Since
	  the API already expects them to be 'const', this patch changes
	  the declarations of all existing instances that were not already
	  declared that way.

2012-07-16 19:50 +0000 [r370131]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Code cleanup and bugfix in chan_sip
	  outboundproxy parsing. The bug was clearing the global
	  outboundproxy when a peer-specific outboundproxy was bad. The
	  cleanup reduces duplicate code. Review:
	  https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
	  Michelson

2012-07-16 13:44 +0000 [r370081]  Kinsey Moore <kmoore@digium.com>

	* UPGRADE.txt, CHANGES: Add comments about the BUILD_NATIVE change
	  This is a significant change and mention of it should have gone
	  into UPGRADE.txt and CHANGES.

2012-07-12 20:15 +0000 [r370017]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Add missing
	  ast_hangup() calls on some analog exception paths. Make starting
	  analog_ss_thread() or __analog_ss_thread() failure paths hangup
	  the channel.

2012-07-12 20:05 +0000 [r369993-370014]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Include Expires header for SIP PUBLISH
	  requests RFC3903 requres SIP PUBLISH requests to have Expires
	  headers, so add them. Review:
	  https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth

	* channels/chan_sip.c: Prevent double uri_escaping in chan_sip when
	  pedantic is enabled If pedantic mode is enabled, outbound invites
	  will have double-escaped contacts. This avoids setting an
	  already-escaped string into a field where it is expected to be
	  unescaped. (closes issue ASTERISK-20023) Reported-by: Walter
	  Doekes

2012-07-12 14:23 +0000 [r369970]  Michael L. Young <elgueromexicano@gmail.com>

	* funcs/func_math.c: Correct Documentation For DEC Function The
	  documentation for DEC in func_math.c was incorrect. Looks like a
	  copy and paste error. (Closes issue ASTERISK-20095) Reported by:
	  Billy Chia Tested by: Michael L. Young Patches: func_math.patch
	  uploaded by Billy Chia (license 6381)

2012-07-11 17:08 +0000 [r369937]  Tilghman Lesher <tilghman@meg.abyt.es>

	* funcs/func_realtime.c: Allow the REALTIME() function to report
	  errors back to the caller. Also, do more error checking on the
	  arguments specified to the REALTIME() function and clarify the
	  documentation. While I was editing the file, a few coding
	  guidelines fixups, as well. Review:
	  https://reviewboard.asterisk.org/r/2031/

2012-07-30  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.15.0 Released.

2012-07-11  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.15.0-rc1 Released.

2012-07-10 13:33 +0000 [r369869]  Kinsey Moore <kmoore@digium.com>

	* apps/app_stack.c, main/pbx.c: Improve Goto and GotoIf related
	  documentation Correct documentation on labeliftrue and
	  labeliffalse parameters of GotoIf() and update several other
	  locations that use the same syntax. (closes issue ASTERISK-20007)
	  Patch-by: Leif Madsen Reported-by: WIMPy

2012-07-09 17:05 +0000 [r369818]  Jason Parker <jparker@digium.com>

	* configs/sip_notify.conf.sample: Add Digium phones context to
	  sip_notify sample config. This makes it so that they can be
	  reconfigured remotely. (closes issue ASTERISK-19910)

2012-07-09 14:38 +0000 [r369792]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Fix small behavioral change
	  accidentally introduced in r369750 When removing the warning for
	  AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
	  the return value, which would likely make the indication not be
	  sent in audio. This fixes that while still removing the warning
	  message.

2012-07-06 20:54 +0000 [r369750]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Add case for FLASH control frames
	  so that we don't display a warning. chan_sip channels can receive
	  flash control frames when connected to analog phones and possibly
	  for other reasons. There really isn't a reason to warn when these
	  frames are received, we can safely ignore them. Patches:
	  dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)

2012-07-06 18:40 +0000 [r369708-369731]  Mark Michelson <mmichelson@digium.com>

	* main/tcptls.c: Remove a superfluous and dangerous freeing of an
	  SSL_CTX. The problem here is that multiple server sessions share
	  a SSL_CTX. When one session ended, the SSL_CTX would be freed and
	  set NULL, leaving the other sessions unable to function. The code
	  being removed is superfluous because the SSL_CTX structures for
	  servers will be properly freed when ast_ssl_teardown is called.
	  (closes issue ASTERISK-20074) Reported by Trevor Helmsley
	  Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license
	  #5049) Testers: Trevor Helmsley

	* main/bridging.c: Fix bridging thread leak. The bridge thread was
	  exiting but was never being reaped using pthread_join(). This has
	  been fixed now by calling pthread_join() in ast_bridge_destroy().
	  (closes issue ASTERISK-19834) Reported by Marcus Hunger Review:
	  https://reviewboard.asterisk.org/r/2012

2012-07-05 19:01 +0000 [r369652]  Kinsey Moore <kmoore@digium.com>

	* apps/app_voicemail.c: AST-2012-011: Resolve heap corruption issue
	  with voicemail The heard and deleted arrays in the voicemail
	  state structure were not handled properly following the memory
	  leak fix in r354890 and a fix for an invalid free in r356797.
	  This could result in accessing and writing into freed memory. The
	  allocation for these arrays has been reworked to avoid the
	  possibility of invalid frees, access of freed memory, and crashes
	  that were occurring as a result of this. Locking around accesses
	  and modifications of the voicemail state structure members
	  dh_arraysize, heard, and deleted has been added to prevent
	  simultaneous modification and access when IMAP storage is in use.
	  If IMAP storage is not in use, this locking is not compiled in.
	  Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
	  ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney,
	  Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license
	  6273)

2012-07-05 17:01 +0000 [r369626]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Do not send a BYE when a provisional
	  response arrives during a re-INVITE Commits r369557 and r369579
	  were done to improve handling of re-INVITEs when the UA that was
	  supposed to receive the re-INVITE fails to respond. A limitation
	  of those patches occurred when a UA sent a provisional response
	  to the re-INVITE. This triggered a sending of a BYE in
	  check_pending. This patch tweaks the handling of the re-INVITE
	  such that a BYE is not sent in response to those messages. (issue
	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
	  patches: (reinvite_tweak.diff license #5012 by Steve Davies)

2012-07-03 16:58 +0000 [r369557-369579]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: More improvements to re-INVITEs timing out
	  after a provisional response There is no need to call
	  check_pendings() on a final response to an INVITE when destroying
	  the scheduler entry as it will be done later during normal
	  processing. (issue ASTERISK-19992)

	* channels/sip/include/sip.h, channels/chan_sip.c: Better handle
	  re-INVITEs with provisional but no final repsonses A previous
	  attempt at fixing this issue had negative side effects related to
	  attended transfers which this patch should resolve. Many thanks
	  to Steve Davies for all of the good suggestions and testing.
	  (closes issue ASTERISK-19992) Reported by: Steve Davies Tested
	  by: Steve Davies, Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/2009/

2012-06-29 16:52 +0000 [r369471-369490]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: With some configurations a transport is not
	  actually specified so assume UDP in these cases.

	* channels/chan_sip.c: Make the address family filter specific to
	  the transport. (closes issue ASTERISK-16618) Reported by: Leif
	  Madsen Review: https://reviewboard.asterisk.org/r/1667/

2012-06-27 20:58 +0000 [r369436]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: AST-2012-010:
	  Clean up after a reinvite that never gets a final response The
	  basic problem is that if a re-INVITE is sent by Asterisk and it
	  receives a provisional response, but no final response, then the
	  dialog is never torn down. In addition to leaking memory, this
	  also leaks file descriptors and will eventually lead to Asterisk
	  no longer being able to process calls. This patch just keeps
	  track of whether there is an outstanding re-INVITE, and if there
	  is goes ahead and cleans up everything as though there was no
	  outstanding reinvite. Review:
	  https://reviewboard.asterisk.org/r/2009/ (closes issue
	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
	  Davies, Terry Wilson

2012-06-26 13:21 +0000 [r369366-369390]  Matthew Jordan <mjordan@digium.com>

	* main/adsi.c: Fix crash in unloading of res_adsi module When
	  res_adsi is unloaded, it removes the ADSI functions that it
	  previously installed by passing a NULL adsi_funcs pointer to
	  ast_adsi_install_funcs. This function was not checking whether or
	  not the adsi_funcs pointer passed in was NULL before
	  dereferencing it to check whether or not the version of the
	  functions matches what the core was expecting it. This patch
	  makes it so that the version is only checked if a potentially
	  valid adsi_funcs pointer was passed in. Passing in NULL removes
	  the installed functions, bypassing the version check.

	* main/cdr.c: Tweak CDR change in r369351 As Tilghman pointed out
	  on review 1996, the check to see if a CDR end time has been set
	  is sufficient to know whether or not the duration value can be
	  used. The check-in done for r369351 forgot to include this
	  change.

2012-06-25 19:13 +0000 [r369352]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: Re-fix how local
	  tag is generated when sending a 481 to an INVITE. Match our local
	  tag to whatever to-tag was sent in the initial INVITE. Because
	  the size of the to-tag may not fit in the buffer in the sip_pvt,
	  it has been changed to a string field. (closes issue
	  ASTERISK-19892) reported by Walter Doekes Review:
	  https://reviewboard.asterisk.org/r/1977

2012-06-25 19:12 +0000 [r369351]  Matthew Jordan <mjordan@digium.com>

	* main/cdr.c: Fix incorrect duration reporting in CDRs created in
	  batch mode Certain places in core/cdr.c would, if the duration
	  value were 0, calculate the duration as being the delta between
	  the current time and the time at which the CDR record was
	  started. While this does not typically cause a problem in
	  non-batch mode, this can cause an issue in batch mode where CDR
	  records are gathered and written long after those calls have
	  ended. In particular, this affects calls that were never
	  answered, as those are expected to have a duration of 0. Often,
	  this would result in CDR logs with a significant number of calls
	  with lengthy durations, but dispositions of "BUSY". Note that
	  this does not affect cdr_csv, as that backend does not use
	  ast_cdr_getvar and instead directly reports the duration value.
	  The affected core backends include cdr_apative_odbc and
	  cdr_custom; other extended or deprecated CDR backends may
	  potentially still directly manipulate the duration values. (issue
	  ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
	  Reported by: Thomas Arimont Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1996/

2012-06-25 15:57 +0000 [r369327]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix Bridge application occasionally returning to
	  the wrong location. * Fix do_bridge_masquerade() getting the
	  resume location from the zombie channel. The code must not touch
	  a clone channel after it has masqueraded it. The clone channel
	  has become a zombie and is starting to hangup. (closes issue
	  ASTERISK-19985) Reported by: jamicque Patches:
	  jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: jamicque

2012-06-25 15:50 +0000 [r369302-369324]  Mark Michelson <mmichelson@digium.com>

	* main/adsi.c (added): Forgot to svn add this file in my last
	  commit.

	* res/res_adsi.exports.in (removed), include/asterisk/adsi.h,
	  main/Makefile, res/res_adsi.c: Eliminate embedding of res_adsi.so
	  module. The way this is done is to stop using the optional API.
	  Instead, res_adsi.so, when loaded fills in a table of function
	  pointers. Review: https://reviewboard.asterisk.org/r/1991

	* channels/chan_sip.c: Be more consistent with the return code for
	  requests received from invalid domain. When Asterisk receives an
	  INVITE from an external domain when allowexternaldomains=no send
	  a 403 instead of a 404. This is consistent with Asterisk's
	  behavior when receiving a REGISTER in this situation. (Closes
	  issue ASTERISK-19601) Reported by Matthew Jordan Patches:
	  ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
	  #5049)

2012-06-23 00:04 +0000 [r369235-369282]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix Bridge application and AMI Bridge action
	  error handling. * Fix AMI Bridge action disconnecting the AMI
	  link on error. * Fix AMI Bridge action and Bridge application not
	  checking if their masquerades were successful. * Fix Bridge
	  application running the h-exten when it should not. * Made
	  do_bridge_masquerade() return if the masquerade was successful so
	  the Bridge application and AMI Bridge action could deal with it
	  correctly. * Made bridge_call_thread_launch() hangup the passed
	  in channels if the bridge_call_thread fails to start. Those
	  channels would have been orphaned. * Made builtin_atxfer() check
	  the success of the transfer masquerade setup.

	* apps/app_queue.c: Explicitly check caller hangup in app Queue
	  rather than a polluted res2 value.

	* apps/app_dial.c: Check if PBX was started and fix F and F(x)
	  action logic in Dial application.

	* main/ccss.c: Check if PBX was started for generic CCSS recall.

	* channels/chan_sip.c: Change incorrect chan_sip zombie hangup
	  debug message. They are all zombies now.

2012-06-22 19:28 +0000 [r369214]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't crash on a guest directmedia call A
	  sip_pvt may not have relatedpeer set if a call doesn't match up
	  with a peer. If there is no relatedpeer, there is no direct media
	  ACL to apply, so just return that it is allowed. (closes issue
	  ASTERISK-20040) Reported by: Terry Wilson

2012-06-22 17:14 +0000 [r369195]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Don't parse media stream state for SIP video
	  streams The sendonly/recvonly/sendrecv/inactive media stream
	  attributes were parsed for video, but nothing was ever done with
	  them. With this code removed, an UNSUPPORTED message is produced
	  when these attributes are used in conjunction with a video stream
	  which is the better behavior since they were never really
	  supported in the first place.

2012-06-20 17:33 +0000 [r369130-369146]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
	  locking issue on empty callList (issue ASTERISK-19298) Reported
	  by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch

	* addons/chan_ooh323.c: fix compile error (1.8 don't have
	  ast_channel_name macro)

2012-06-20 02:03 +0000 [r369108]  Michael L. Young <elgueromexicano@gmail.com>

	* include/asterisk/netsock2.h, main/netsock2.c: Fix NULL pointer
	  segfault in ast_sockaddr_parse() While working with
	  ast_parse_arg() to perform a validity check, a segfault occurred.
	  The segfault occurred due to passing a NULL pointer to
	  ast_sockaddr_parse() from ast_parse_arg(). According to the
	  documentation in config.h, "result pointer to the result. NULL is
	  valid here, and can be used to perform only the validity checks."
	  This patch fixes the segfault by checking for a NULL pointer.
	  This patch also adds documentation to netsock2.h about why it is
	  necessary to check for a NULL pointer. (Closes issue
	  ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
	  L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
	  by Michael L. Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1990/

2012-06-19 23:28 +0000 [r369090]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
	  config file (rtp voice, video, udptl except rtcp) (closes issue
	  ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
	  19179-ooh323-2.patch

2012-06-19 15:30 +0000 [r369066]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix request routing issue when outboundproxy
	  is used. Asterisk was incorrectly setting the destination of
	  CANCELs and ACKs for error responses to the URI of the initial
	  INVITE. This resulted in further requests, such as INVITEs with
	  authentication credentials, to be routed incorrectly. Instead,
	  when these CANCEL or ACKs are to be sent, we should simply keep
	  the destination the same as what it previously was. There is no
	  need to alter it any. (closes issue ASTERISK-20008) Reported by
	  Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark
	  Michelson (license #5049)

2012-06-18 18:07 +0000 [r369043]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix monitoring calls put in a parking lot. * Fix
	  a regression that was introduced by -r366167 which effectively
	  disabled monitoring parked calls. (closes issue ASTERISK-20012)
	  Reported by: sdolloff Tested by: rmudgett

2012-06-15 15:57 +0000 [r369001-369002]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/find_missing_support_level (added): Add a script to
	  enable finding source files without support-levels defined.

	* main/devicestate.c, main/astfd.c, main/ssl.c,
	  main/taskprocessor.c, main/astobj2.c, main/indications.c,
	  main/config.c, main/loader.c, main/term.c, main/cli.c,
	  channels/sig_analog.c, main/framehook.c, main/strcompat.c,
	  main/plc.c, res/ais/evt.c, main/fskmodem_int.c, main/syslog.c,
	  main/stdtime/localtime.c, main/db.c, main/bridging.c,
	  channels/sig_ss7.c, main/datastore.c, main/sched.c,
	  channels/sip/sdp_crypto.c, main/pbx.c, main/strings.c,
	  channels/vcodecs.c, channels/iax2-provision.c, main/aoc.c,
	  pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c,
	  main/chanvars.c, main/netsock.c, main/data.c, main/srv.c,
	  channels/chan_misdn.c, main/privacy.c,
	  channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
	  main/test.c, main/audiohook.c, main/alaw.c, main/asterisk.c,
	  main/timing.c, main/global_datastores.c, main/fskmodem_float.c,
	  main/ccss.c, channels/sip/reqresp_parser.c,
	  channels/misdn/isdn_msg_parser.c, main/utils.c, main/xml.c,
	  main/autochan.c, main/enum.c, channels/misdn/isdn_lib.c,
	  main/fskmodem.c, channels/misdn_config.c, main/io.c,
	  res/ael/pval.c, main/channel.c, main/cdr.c, main/ulaw.c,
	  main/dial.c, main/tdd.c, main/heap.c, channels/console_gui.c,
	  channels/misdn/ie.c, main/logger.c, channels/console_board.c,
	  main/app.c, main/image.c, main/dns.c, main/lock.c, main/stun.c,
	  main/dnsmgr.c, channels/sip/srtp.c, main/translate.c,
	  main/slinfactory.c, main/jitterbuf.c, main/acl.c,
	  channels/sig_pri.c, main/tcptls.c, main/hashtab.c,
	  main/abstract_jb.c, main/callerid.c, main/file.c,
	  res/snmp/agent.c, main/astmm.c, channels/misdn/portinfo.c,
	  main/event.c, channels/sip/config_parser.c, channels/vgrabbers.c,
	  main/xmldoc.c, main/dsp.c, main/udptl.c, main/netsock2.c,
	  main/autoservice.c, main/rtp_engine.c, main/frame.c,
	  main/security_events.c, res/ais/clm.c, main/threadstorage.c,
	  main/say.c, channels/console_video.c: Add support-level
	  indications to many more source files. Since we now have tools
	  that scan through the source tree looking for files with specific
	  support levels, we need to ensure that every file that is a
	  component of a 'core' or 'extended' module (or the main Asterisk
	  binary) is explicitly marked with its support level. This patch
	  adds support-level indications to many more source files in tree,
	  but avoids adding them to third-party libraries that are included
	  in the tree and to source files that don't end up involved in
	  Asterisk itself.

2012-06-14 15:23 +0000 [r368898-368927]  Mark Michelson <mmichelson@digium.com>

	* main/Makefile: Revert Makefile change to remove embedding
	  res_adsi.so The change has resulted in a linking error for
	  certain versions of GCC. This is much worse than the original
	  issue, so for now, temporarily revert the change. A more thorough
	  change will be sought out.

	* funcs/func_volume.c: Fix a deadlock that occurs when func_volume
	  is used on a local channel. This was discovered by trying to
	  perform a call forward to an extension that makes use of
	  func_volume. When the local channel is optimized away, the
	  datastore on the local;2 channel would have its audiohook
	  destroyed rather than detaching the audiohook from the channel
	  and then destroying it. With this patch, func_volume's datastore
	  destructor takes the proper route of detaching the audiohook and
	  then destroying it. (closes issue ASTERISK-19611) reported by
	  Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
	  Michelson (license #5049)

2012-06-13 20:26 +0000 [r368894]  Matthew Jordan <mjordan@digium.com>

	* res/res_smdi.c, res/res_adsi.c: Mark res_smdi/res_adsi as 'core'
	  supported modules Recently, various issues surrounding weak
	  symbols have caused problems with modules that rely on that
	  feature to be enabled in menuselect. This includes app_voicemail
	  and chan_dahdi, as they both rely upon res_smdi and res_adsi,
	  which, in certain circumstances, may not be enabled by default in
	  menuselect. Because res_smdi/res_adsi are dependencies for
	  chan_dahdi/app_voicemail, this patch marks both as 'core'
	  supported modules. This will allow both app_voicemail and
	  chan_dahdi to be enabled as well, regardless of whether or not
	  that system supports weak symbols. (issue AST-900) Reported by:
	  Thomas Arimont (issue AST-885) Reported by: Denis Alberto
	  Martinez

2012-06-13 19:00 +0000 [r368873]  Mark Michelson <mmichelson@digium.com>

	* main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
	  the result is that Asterisk has a phantom module loaded at
	  startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
	  reported by Leif Madsen

2012-06-13 14:27 +0000 [r368830-368852]  Matthew Jordan <mjordan@digium.com>

	* Makefile: Do not install empty directories; add ASTLIBDIR r368830
	  modified the installation script to only create a directory if
	  that directory does not exist. If some directory variable was
	  empty, it would attempt to create the empty location. It also
	  failed to create the ASTLIBDIR directory. This patch fixes it
	  such that the correct directories are made and only created if a
	  value specifying them actually exists.

	* Makefile: Do not perform install on existing directories If a
	  directory already exists, performing a 'make install' will remove
	  the permissions associated with the current directory and replace
	  them with the permissions of the user executing the install. This
	  patch changes this behavior to only perform an install on the
	  directory if the directory does not exist. Thus, if a user later
	  changes the permissions on that directory, those permissions will
	  be preserved in subsequent installs. Review:
	  https://reviewboard.asterisk.org/r/1986 Review:
	  https://reviewboard.asterisk.org/r/1864 (closes issue
	  ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
	  Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
	  by mjordan)

2012-06-12 15:36 +0000 [r368807]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Set the Caller ID "tag" on peers even if
	  remote party information is present. On incoming calls, we were
	  setting the cid_tag on the dialog only if there was no remote
	  party information (Remote-Party-ID or P-Asserted-Identity)
	  present. The Caller ID tag is an invented parameter, though, and
	  should be set no matter the circumstance. (closes issue
	  ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
	  Reported by Trey Blancher

2012-06-11 17:03 +0000 [r368759]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, channels/chan_iax2.c, main/channel.c,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_sip.c: Fix deadlock potential with
	  ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
	  the channel lock held can result in a deadlock because the
	  function also locks the bridged channel. (issue ASTERISK-19537)
	  (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
	  Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
	  Davis

2012-06-11 15:13 +0000 [r368719-368738]  Kinsey Moore <kmoore@digium.com>

	* apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
	  res/res_config_odbc.c, channels/sip/dialplan_functions.c,
	  pbx/pbx_config.c, apps/app_directory.c, res/res_odbc.c,
	  res/res_speech.c, apps/app_voicemail.c, main/udptl.c,
	  channels/sip/sdp_crypto.c, channels/chan_sip.c, res/res_fax.c,
	  main/say.c, funcs/func_strings.c, channels/sip/reqresp_parser.c:
	  Fix coverity UNUSED_VALUE findings in core support level files
	  Most of these were just saving returned values without using them
	  and in some cases the variable being saved to could be removed as
	  well. (issue ASTERISK-19672)

	* main/md5.c: Fix compilation in dev-mode Backport a compilation
	  fix in md5.c from trunk that only showed up in dev-mode under
	  certain compiler versions.

2012-07-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.14.0 Released.

2012-07-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.14.0-rc2 Released.

	* AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
  	  transactions

	* AST-2012-011: Remote Crash Vulnerability in VoiceMail Application

	* Fix crash on a guest directmedia call

	  A sip_pvt may not have relatedpeer set if a call doesn't match up
	  with a peer.  If there is no relatedpeer, there is no direct media
	  ACL to apply, so just return that is is allowed.

	  (closes issue ASTERISK-20040)
	
	* Fix request routing issue when outboundproxy is used

	  Asterisk was incorrectly setting the destination of CANCELs and ACKs
	  for error responses to the URI of the initial INVITE.  This resulted
	  in further requests, such as INVITEs with authentication
	  credentials, to be routed incorrectly.  Instead when these CANCEL or
	  ACKs are to be esnt, we should simply keep the destination the same
	  as what it previously was.  There is no need to alter it any.

	  (closes issue ASTERISK-20008)
	
	* Fix monitoring calls put in a parking lot

	  Fix a regression that was introduced by r366167 which effectively
	  disabled monitoring parked calls.

	  (closes issue ASTERISK-20012)

2012-06-08  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.14.0-rc1 Released.

2012-06-06 21:27 +0000 [r368644]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
	  to orignate a second call deadlock. A deadlock can occur when a
	  POTS phone tries to flash hook to originate a second call for
	  3-way or transfer. If another process is scanning the channels
	  container when the POTS line flash hooks then a deadlock will
	  occur. * Release the channel and private locks when creating a
	  new channel as a result of a flash hook. (closes issue
	  ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett

2012-06-06 19:13 +0000 [r368625]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a specific scenario where ACKs are not
	  matched. If a dialog-starting INVITE contains a to-tag, then
	  Asterisk will respond with a 481. In this case, the resulting
	  incoming ACK would not be matched, so Asterisk would continue
	  retransmitting the 481 until the transaction times out. There
	  were two issues. Asterisk, upon creating a sip_pvt would generate
	  a local tag. However, when the time came to transmit the 481,
	  since there was a to-tag in the INVITE, Asterisk would place this
	  original to-tag in the 481 response. When the ACK came in,
	  Asterisk would attempt to match the to-tag in the ACK to the
	  generated local tag. Unfortunately, Asterisk never actually
	  transmitted a response with the generated local tag, so the
	  to-tag in the ACK would not match. The other problem was that
	  when the 481 was sent, nothing was set on the sip_pvt to indicate
	  what CSeq is expected in the ACK. To fix the first problem, we
	  zero out the to-tag seen in the incoming INVITE. This way,
	  Asterisk, when time to send a response, will send its generated
	  local tag instead. To fix the second problem, we set the
	  sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
	  481. (closes issue ASTERISK-19892) Reported by Mark Michelson

2012-06-06 17:20 +0000 [r368604]  Matthew Jordan <mjordan@digium.com>

	* build_tools/make_version: Add feature modifier to versions
	  produced from branches Certain branches, such as Certified
	  Asterisk, may have a modifier added to them that specifies the
	  features available in that branch. For branches, this modifier is
	  expected to be reflected in the location of the branch in
	  subversion. For example, a subversion of URL of
	  /certified/branches/1.8.11 would have a feature modifier of
	  'certified'. This is slightly different then how features are
	  determined for tags, where the feature is part of the actual tag
	  name, e.g., "10.5.0-digiumphones". In keeping with the
	  nomenclature used for tags, the feature specifier for branches is
	  translated and placed after the revision numbers. For the example
	  given previously, this would result in a branch version of
	  "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".

2012-06-06 16:07 +0000 [r368586]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure overlapping hold flags do not
	  conflict When changing between different modes of hold, the flags
	  were not being cleared out properly causing a failure to change
	  hold states. (closes issue ASTERISK-19919) Patch-by: Morten
	  Tryfoss Reported-by: Morten Tryfoss

2012-06-06 01:08 +0000 [r368567]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix parked call performing a DTMF blind transfer
	  after being retrieved. When a parked call was retrieved from the
	  parking lot, it could not do a blind transfer because it caused
	  the involved calls to be hung up unconditionally. * Made the
	  ParkedCall application return the ast_bridge_call() return value.
	  (closes issue ABE-2862) Reported by: Vlad Povorozniuc

2012-06-05 15:26 +0000 [r368520-368533]  Kinsey Moore <kmoore@digium.com>

	* apps/app_minivm.c: Resolve some build warnings My newly upgraded
	  compiler caught these usages of uninitialized values. They
	  weren't actually used.

	* apps/app_voicemail.c: Ensure that pages and emails are sent using
	  RFC822-compliant date format When localization was added to
	  app_voicemail, these headers were altered when they should have
	  remained in en_US format for RFC compliance. This reverts the
	  changes to those two lines. (closes issue ASTERISK-19876)

2012-06-04 21:56 +0000 [r368498]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Relay proper SIP responses on calling side.
	  Revision 351130 broke corect HANGUPCAUSE setting for the 404 case
	  in chan_sip. Other cases were also potentially broken. This patch
	  fixes the relaying of causes to be what they used to be. (closes
	  issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter
	  Doekes (via a reviewboard test to be committed later) Patches:
	  chan_sip.diff uploaded by Pavel Troller (license #6302)

2012-06-04 21:10 +0000 [r368405-368469]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
	  ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call

	* main/channel.c: Fix potential deadlock between masquerade and
	  chan_local. * Restructure ast_do_masquerade() to not hold channel
	  locks while it calls ast_indicate(). * Simplify many calls to
	  ast_do_masquerade() since it will never return a failure now. If
	  it does fail internally because a channel driver callback
	  operation failed, the only thing ast_do_masquerade() can do is
	  generate a warning message about strange things may happen and
	  press on. * Fixed the call to ast_bridged_channel() in
	  ast_do_masquerade(). This change fixes half of the deadlock
	  reported in ASTERISK-19801 between masquerades and chan_iax.
	  (closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
	  rmudgett Review: https://reviewboard.asterisk.org/r/1915/

2012-06-01 23:21 +0000 [r368308]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_stack.c: Fix deadlock when Gosub used with alternate
	  dialplan switches. Attempting to remove a channel from
	  autoservice with the channel lock held will result in deadlock. *
	  Restructured gosub_exec() to not call ast_parseable_goto() and
	  ast_exists_extension() with the channel lock held. (closes issue
	  ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett

2012-06-01 18:18 +0000 [r368218]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Improve SDP parsing warning messages *
	  'Unsupported media type' is only reported when that is in fact
	  the case, not when a supported media type is included in an 'm'
	  line that has an invalid format. * All warning messages related
	  to parsing 'm' lines now include the 'm' line contents. * (minor
	  bugfix) newline added to port-number-zero warning messages. *
	  Warning messages improved to use RFC-specified terminology for
	  various items. * Warnings for offers that include more than one
	  port for a single media type now include the media type. Review:
	  https://reviewboard.asterisk.org/r/1811/

2012-06-01 03:25 +0000 [r368092]  Michael L. Young <elgueromexicano@gmail.com>

	* funcs/func_channel.c: Add documentation to function CHANNEL for
	  options echocan_mode and buffers The ability to set
	  "echocan_mode" and "buffers" through the dialplan was added to
	  chan_dahdi some time ago. This patch adds some documentation to
	  func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
	  Noll Tested by: Michael L. Young Patches:
	  asterisk-19911-branch18.diff uploaded by Michael L. Young
	  (license 5026) Review: https://reviewboard.asterisk.org/r/1949/

2012-05-31 18:00 +0000 [r367906-368039]  Richard Mudgett <rmudgett@digium.com>

	* main/db1-ast/btree/bt_open.c, apps/app_queue.c,
	  channels/chan_iax2.c, pbx/pbx_config.c, res/ael/pval.c,
	  main/tcptls.c, main/manager.c, res/res_config_odbc.c,
	  channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c,
	  main/features.c: Coverity Report: Fix issues for error type
	  REVERSE_INULL (core modules) * Fixes findings:
	  0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt
	  Jordan

	* channels/sig_pri.c, channels/sig_ss7.c: Use the
	  DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)

	* channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
	  executing CLI "pri show channels" and "ss7 show channels"
	  commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
	  * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
	  deadlock properly. * Code ss7_grab() better. (closes issue
	  ASTERISK-19854) Reported by: Jaxon Patches:
	  jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
	  by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
	  Jaxon

	* apps/app_meetme.c: Coverity Report: Fix issues for error type
	  REVERSE_INULL (deprecated modules) * Fix only issue pointed out
	  by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
	  * Change use of %i to %d in sscanf() in find_user(). The use of
	  %i gives unexpected parsing because it can accept hex, octal, and
	  decimal integer formats. * Changed other uses of %i in
	  app_meetme() to use %d for consistency. (issue ASTERISK-19648)
	  Reported by: Matt Jordan

2012-05-29 18:30 +0000 [r367843]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_skinny.c: AST-2012-008: Fix remote crash
	  vulnerability in chan_skinny When a skinny session is
	  unregistered, the corresponding device pointer is set to NULL in
	  the channel private data. If the client was not in the on-hook
	  state at the time the connection was closed, the device pointer
	  can later be dereferenced if a message or channel event attempts
	  to use a line's pointer to said device. The patches prevent this
	  from occurring by checking the line's pointer in message handlers
	  and channel callbacks that can fire after an unregistration
	  attempt. (closes issue ASTERISK-19905) Reported by: Christoph
	  Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
	  AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
	  AST-2012-008-10.diff uploaded by mjordan (license 6283)

2012-05-25 16:28 +0000 [r367781]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
	  without suggested MOH class crash. * Made schedule_delivery() set
	  the received frame f->data.ptr to NULL if the datalen is zero. *
	  Fix queue_signalling() memcpy() size error. * Made
	  queue_signalling() not use C++ keyword variable names. (closes
	  issue ASTERISK-19597) Reported by: mgrobecker Patches:
	  jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett, Michael L. Young

2012-05-25 02:27 +0000 [r367730]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's
	  allowtransfer setting The pvt_sip allowtransfer was not being set
	  to that of the peer's setting. Therefore, the global
	  allowtransfer setting was being used instead which would lead to
	  calls not being transfered if the global setting was set to 'no'
	  despite the setting on the peer being 'yes' and vice versa, calls
	  would be allowed to transfer even if the peer's setting was 'no'
	  but the global setting was 'yes'. (Closes issue ASTERISK-19856)
	  Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
	  issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
	  Young (license 5026) Review:
	  https://reviewboard.asterisk.org/r/1923/

2012-05-24 22:21 +0000 [r367469-367678]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, apps/app_dial.c: Fix Dial I option ignored if
	  dial forked and one fork redirects. The Dial and Queue I option
	  is intended to block connected line updates and redirecting
	  updates. However, it is a feature that when a call is locally
	  redirected, the I option is disabled if the redirected call runs
	  as a local channel so the administrator can have an opportunity
	  to setup new connected line information. Unfortunately, the Dial
	  and Queue I option is disabled for *all* forked calls if one of
	  those calls is redirected. * Make the Dial and Queue I option
	  apply to each outgoing call leg independently. Now if one
	  outgoing call leg is locally redirected, the other outgoing calls
	  are not affected. * Made Dial not pass any redirecting updates
	  when forking calls. Redirecting updates do not make sense for
	  this scenario. * Made Queue not pass any redirecting updates when
	  using the ringall strategy. Redirecting updates do not make sense
	  for this scenario. * Fixed deadlock potential with chan_local
	  when Dial and Queue send redirecting updates for a local
	  redirect. * Converted the Queue stillgoing flag to a boolean
	  bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett
	  Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1920/

	* main/pbx.c: Fix WaitExten(x,m(musicclass)) string termination.
	  The AST_CONTROL_HOLD MOH class from the WaitExten application can
	  now be queued onto a channel, passed over local channels with the
	  /m option, and passed over IAX channels.

2012-05-23 20:27 +0000 [r367416]  Mark Michelson <mmichelson@digium.com>

	* main/tcptls.c: Only call SSL_CTX_free if DO_SSL is defined.
	  Thanks to Paul Belanger for pointing out this error.

2012-05-23 13:06 +0000 [r367362]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Update a peer's LastMsgsSent when the peer
	  is notified of waiting messages Previously, MWI logic utilized a
	  counter called 'lastmsgssent' to know whether or not MWI NOTIFY
	  requests had been sent to a specific peer. When MWI notifications
	  were changed to use the internal event framework, this value was
	  no longer needed for its original purpose. Hence, it was no
	  longer updated with the new/old message counts for a peer.
	  However, the value was still presented when, either by AMI or
	  CLI, a 'sip show peer [peer]' command was executed. The output of
	  the command would always display the erroneous value of
	  32767/65535 for 'LastMsgsSent'. This patch makes it so that the
	  value of lastmsgssent is updated appropriately. The value should
	  now display the new/old message counts for a particular peer.
	  (closes issue ASTERISK-17866) Reported by: Steve Davies patches
	  by: ast-17866-rb1272.patch (License #5041 by irroot) Modified
	  slightly for this commit Review:
	  https://reviewboard.asterisk.org/r/1939

2012-05-22 17:14 +0000 [r367266-367292]  Terry Wilson <twilson@digium.com>

	* include/asterisk/channel.h, main/cel.c, main/asterisk.c,
	  main/channel.c, include/asterisk/cel.h: Fix race condition for
	  CEL LINKEDID_END event This patch fixes to situations that could
	  cause the CEL LINKEDID_END event to be missed. 1) During a core
	  stop gracefully, modules are unloaded when ast_active_channels ==
	  0. The LINKDEDID_END event fires during the channel destructor.
	  This means that occasionally, the cel_* module will be unloaded
	  before the channel is destroyed. It seemed generally useful to
	  wait until the refcount of all channels == 0 before unloading, so
	  I added a channel counter and used it in the shutdown code. 2)
	  During a masquerade, ast_channel_change_linkedid is called. It
	  calls ast_cel_check_retire_linkedid which unrefs the linkedid in
	  the linkedids container in cel.c. It didn't ref the new linkedid.
	  Now it does. Review: https://reviewboard.asterisk.org/r/1900/

	* channels/chan_sip.c: Resolve crash in subscribing for MWI
	  notifications ASTOBJ_UNREF sets the variable to NULL after
	  unreffing it, so the variable should definitely not be used after
	  that. To solve this in the two cases that affect subscribing for
	  MWI notifications, we instead save the ref locally, and unref
	  them in the error conditions. (closes issue ASTERISK-19827)
	  Reported by: B. R Review:
	  https://reviewboard.asterisk.org/r/1940/

2012-05-18 17:47 +0000 [r367002-367027]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c, main/say.c: Address MISSING_BREAK static
	  analysis reports some more. This addresses core findings 4 and 6.
	  Moises Silva helped me by stating that a break could be safely
	  added to the case where it is added in chan_dahdi.c In say.c, I
	  have added a comment indicating that static analysis complains
	  but that it is currently unknown if this is correct. This fixes
	  all core findings of this type. (closes issue ASTERISK-19662)
	  reported by Matthew Jordan

	* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
	  Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
	  structures were allocated but never freed. This was a bigger
	  issue for clients than servers since new SSL_CTX structures could
	  be allocated for each connection. Servers, on the other hand,
	  typically set up a single SSL_CTX for their lifetime. This is
	  solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
	  ssl_ctx on it, it is freed so that a new one can take its place.
	  2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
	  been added so that servers can properly free their SSL_CTXs.
	  (issue ASTERISK-19278)

2012-05-18 15:42 +0000 [r366944]  Matthew Jordan <mjordan@digium.com>

	* main/cli.c, channels/chan_sip.c, funcs/func_odbc.c: Fix more
	  memory leaks This patch adds to what was fixed in r366880.
	  Specifically, it addresses the following: * chan_sip: dispose of
	  an allocated frame in off nominal code paths in sip_rtp_read *
	  func_odbc: when disposing of an allocated resultset, ensure that
	  any rows that were appended to that resultset are also disposed
	  of * cli: free the created return string buffer in another off
	  nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1922/

2012-05-18 14:16 +0000 [r366882]  Kinsey Moore <kmoore@digium.com>

	* channels/sip/config_parser.c: Reorder and renumber tests
	  appropriately It appears that a patch did not apply properly when
	  adding tests 12 and 13 and test 11 was duplicated. These tests
	  have been reordered and renumbered such that they make sense.

2012-05-18 13:58 +0000 [r366880]  Matthew Jordan <mjordan@digium.com>

	* res/res_calendar_caldav.c, res/res_musiconhold.c,
	  res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c,
	  main/enum.c, main/editline/term.c, main/config.c, res/res_srtp.c,
	  main/editline/tokenizer.c, main/cli.c, channels/chan_dahdi.c,
	  main/data.c, funcs/func_odbc.c, apps/app_minivm.c,
	  main/features.c, main/editline/readline.c,
	  channels/sip/config_parser.c, main/xmldoc.c, res/res_calendar.c,
	  apps/app_voicemail.c, res/res_rtp_asterisk.c, main/netsock2.c,
	  res/res_calendar_icalendar.c, res/res_calendar_exchange.c,
	  main/pbx.c, apps/app_page.c, channels/chan_sip.c,
	  funcs/func_dialgroup.c, apps/app_record.c: Fix a variety of
	  memory leaks This patch addresses a number of memory leaks in a
	  variety of modules that were found by a static analysis tool. A
	  brief summary of the changes: * app_minivm: free ast_str objects
	  on off nominal paths * app_page: free the ast_dial object if the
	  requested channel technology cannot be appended to the dialing
	  structure * app_queue: if a penalty rule failed to match any
	  existing rule list names, the created rule would not be inserted
	  and its memory would be leaked * app_read: dispose of the created
	  silence detector in the presence of off nominal circumstances *
	  app_voicemail: dispose of an allocated unique ID field for MWI
	  event un-subscribe requests in off nominal paths; dispose of
	  configuration objects when using the secret.conf option *
	  chan_dahdi: dispose of the allocated frame produced by
	  ast_dsp_process * chan_iax2: properly unref peer in CLI command
	  "iax2 unregister" * chan_sip: dispose of the allocated frame
	  produced by sip_rtp_read's call of ast_dsp_process; free memory
	  in parse unit tests * func_dialgroup: properly deref ao2 object
	  grhead in nominal path of dialgroup_read * func_odbc: free
	  resultset in off nominal paths of odbc_read * cli: free
	  match_list in off nominal paths of CLI match completion * config:
	  free comment_buffer/list_buffer when configuration file load is
	  unchanged; free the same buffers any time they were created and
	  config files were processed * data: free XML nodes in various
	  places * enum: free context buffer in off nominal paths *
	  features: free ast_call_feature in off nominal paths of
	  applicationmap config processing * netsock2: users of
	  ast_sockaddr_resolve pass in an ast_sockaddr struct that is
	  allocated by the method. Failures in ast_sockaddr_resolve could
	  result in the users of the method not knowing whether or not the
	  buffer was allocated. The method will now not allocate the
	  ast_sockaddr struct if it will return failure. * pbx: cleanup
	  hash table traversals in off nominal paths; free ignore pattern
	  buffer if it already exists for the specified context * xmldoc:
	  cleanup various nodes when we no longer need them *
	  main/editline: various cleanup of pointers not being freed before
	  being assigned to other memory, cleanup along off nominal paths *
	  menuselect/mxml: cleanup of value buffer for an attribute when
	  that attribute did not specify a value * res_calendar*: responses
	  are allocated via the various *_request method returns and should
	  not be allocated in the various write_event methods; ensure
	  attendee buffer is freed if no data exists in the parsed node;
	  ensure that calendar objects are de-ref'd appropriately *
	  res_jabber: free buffer in off nominal path * res_musiconhold:
	  close the DIR* object in off nominal paths * res_rtp_asterisk: if
	  we run out of ports, close the rtp socket object and free the rtp
	  object * res_srtp: if we fail to create the session in libsrtp,
	  destroy the temporary ast_srtp object (issue ASTERISK-19665)
	  Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1922

2012-05-17 14:40 +0000 [r366791]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Fix missed locking of opposing pvt
	  for directmedia acl from r366547 It also required deadlock
	  avoidance since two sip_pvts structs needed to be locked
	  simultaneously. Trunk handles it differently, so this is a 1.8
	  and 10 patch only. (issue AST-876)

2012-05-17 12:51 +0000 [r366740]  Matthew Jordan <mjordan@digium.com>

	* res/res_calendar_ews.c, channels/chan_dahdi.c: Fix checking
	  bounds of array index after using it; improper sizeof This patch
	  fixes two problems pointed out by a static analysis tool. * In
	  chan_dahdi, when an event is handled the index of the sub channel
	  is first obtained. In very off nominal cases, the method that
	  determines the index can return a negative value. In the event
	  handling code, whether or not the index returned is valid was
	  being checked after that value was used to index into an array.
	  This patch makes it so the value is checked before any indexing
	  is done. * In res_calendar_ews, sizeof was being passed a pointer
	  instead of the struct to determine the amount of memory to
	  allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
	  issue ASTERISK-19671) Reported by: Matt Jordan

2012-05-16 15:52 +0000 [r366597-366650]  Mark Michelson <mmichelson@digium.com>

	* main/http.c: Fix incorrect default port number for HTTP server.
	  Thanks to Tzafrir Cohen for bringing this up on the Asterisk
	  developers mailing list.

	* channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
	  getting a Diversion header's reason parameter. The use here was
	  assuming that the pointer would be updated, but the updated
	  string is actually returned by ast_strip_quoted() instead.

2012-05-15 20:14 +0000 [r366547]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: chan_sip: Check the right channel's host
	  address for directmediapermit/deny Prior to this patch, when
	  checking the addresses for directmediapermit and directmediadeny,
	  Asterisk would check the host address of the channel permit/deny
	  was specified, which differs from the expectations of both our
	  users and the development team. Instead, directmediapermit/deny
	  now checks against the address of the channel that the peer with
	  the ACL is connected to. (issue AST-876) Review:
	  https://reviewboard.asterisk.org/r/1899/

2012-05-14 19:57 +0000 [r366389-366409]  Mark Michelson <mmichelson@digium.com>

	* pbx/dundi-parser.c: Fix two more coverity constant expression
	  result findings. These correspond to findings 0 and 1 in the core
	  findings of ASTERISK-19649. After contacting Mark Spencer, he was
	  unsure of what the intent behind these lines of code were, so
	  they are being axed. For Asterisk 1.8 and 10, the output of
	  debugging DUNDi frames will not be changed, but for trunk the
	  "Retry" portion will be omitted since it does not properly
	  distinguish retransmissions from initial frames. (closes issue
	  ASTERISK-19649) Reported by Matthew Jordan

	* channels/chan_sip.c: Fix broken reinvite glare scenario. To make
	  a long story short, reinvite glares were broken because Asterisk
	  would invert the To and From headers when ACKing a 491 response.
	  The reason was because the initreq of the dialog was being
	  changed to the incoming glared reinvite instead of being set to
	  the outgoing glared reinvite. This change has three parts * In
	  handle_incoming, we never will reject an ACK because it has a
	  to-tag present, even if we think the request may be out of
	  dialog. * In handle_request_invite, we do not change the initreq
	  when receiving a reinvite to which we will respond with a 491. *
	  In handle_request_invite, several superflous settings up
	  pendinginvite have been removed since this is dones automatically
	  by transmit_response_reliable Review:
	  https://reviewboard.asterisk.org/r/1911

2012-05-11 23:53 +0000 [r366296]  Russell Bryant <russell@russellbryant.com>

	* addons/format_mp3.c: format_mp3: Fix a possible crash mp3_read().
	  This patch fixes a potential crash in mp3_read() by not assuming
	  that dbuf has enough data to finish filling up the output buffer.
	  The patch also makes sure that the dbuf state gets reset after we
	  know we read everything out of it already. In passing, this patch
	  includes some other cleanups of this module, including stripping
	  trailing whitespace, formatting fixes based on coding guidelines,
	  and removing a number of unused members from the private state
	  struct. (closes issue ASTERISK-19761) Reported by: Chris
	  Maciejewsk Tested by: Chris Maciejewsk

2012-05-10 23:38 +0000 [r366240]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: * Made ast_change_name() hold the channels
	  container lock while changing the channel name. * Eliminate
	  redundant list not empty check in clone_variables().

2012-05-10 20:50 +0000 [r366167]  Kinsey Moore <kmoore@digium.com>

	* main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c,
	  channels/iax2-parser.c, main/config.c, res/res_monitor.c,
	  main/channel.c, main/cdr.c, res/ael/pval.c, main/data.c,
	  channels/chan_dahdi.c, main/tcptls.c, main/manager.c,
	  main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c,
	  res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c,
	  funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c,
	  channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
	  channels/sip/reqresp_parser.c: Resolve FORWARD_NULL static
	  analysis warnings This resolves core findings from ASTERISK-19650
	  numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56,
	  82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding
	  numbers 26, 33, and 29 were already resolved. Those skipped were
	  either extended/deprecated or in areas of code that shouldn't be
	  disturbed. (Closes issue ASTERISK-19650)

2012-05-10 16:47 +0000 [r366094]  Jonathan Rose <jrose@digium.com>

	* channels/iax2-provision.c, apps/app_queue.c,
	  channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
	  main/asterisk.c, main/db.c, main/xmldoc.c, apps/app_voicemail.c,
	  main/pbx.c, channels/sig_analog.c, channels/chan_sip.c,
	  funcs/func_lock.c, main/features.c, main/acl.c: Coverity Report:
	  Fix issues for error type CHECKED_RETURN for core (issue
	  ASTERISK-19658) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1905/

2012-05-10 16:10 +0000 [r366052]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Close the proper tcptls_session when session
	  creation fails. (issue AST-998) Reported by: Thomas Arimont
	  Tested by: Thomas Arimont

2012-05-10 15:35 +0000 [r365989-366048]  Jonathan Rose <jrose@digium.com>

	* apps/app_chanspy.c, apps/app_page.c, funcs/func_cdr.c,
	  main/features.c, apps/app_disa.c: Coverity Report: Fix issues for
	  error type UNINIT in Core supported modules (issue
	  ASTERISK-19652) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1909/

	* codecs/codec_dahdi.c: Block on frameout if the hardware has
	  enough samples to complete a frame. Fixes some problems with
	  skipping audio in elaborate scenarios involving multiple codecs
	  by making codec_dahdi operate in a more synchronous fashion
	  similar to codec_g729. This change also fixes the use of file
	  conversion tools from Asterisk's CLI. This change may cause the
	  thread responsible for transcoding audio to block briefly (Shaun
	  Ruffell describes this as 'several milliseconds') while waiting
	  for the hardware transcoder. (closes issue ASTERISK-19643)
	  reported by: Shaun Ruffell Patches:
	  0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
	  uploaded by Shaun Ruffell (license 5417)

2012-05-09 16:11 +0000 [r365896]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel
	  outlasts its corresponding sip_pvt. chan_sip was coded under the
	  assumption that a SIP dialog with an owner channel will always be
	  destroyed after the owner channel has been hung up. However,
	  there are situations where the SIP dialog can time out and auto
	  destruct before the corresponding channel has hung up. A typical
	  example of this would be if the 'h' extension in the dialplan
	  takes a long time to complete. In such cases,
	  __sip_autodestruct() would complain about the dialog being auto
	  destroyed with an owner channel still in place. The problem is
	  that even once the owner channel was hung up, the sip_pvt would
	  still be linked in its ao2_container because nothing would ever
	  unlink it. The fix for this is that if __sip_autodestruct() is
	  called for a sip_pvt that still has an owner channel in place,
	  the destruction is rescheduled for 10 seconds in the future. This
	  will continue until the owner channel is finally hung up. (closes
	  issue ASTERISK-19425) reported by David Cunningham Patches:
	  ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
	  (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
	  Dean Vesvuio

2012-05-08 20:14 +0000 [r365631-365692]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_followme.c: * Fix FollowMe memory leak on error paths in
	  app_exec(). * Fix FollowMe leaving recorded caller name file on
	  error paths in app_exec(). * Use correct buffer dimension define
	  in struct call_followme.moh[] and struct fm_args.namerecloc[].
	  This fixes unexpected namerecloc filename length restriction.

	* apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite
	  in FollowMe. * Made use MAX_YN_STRING define to make all
	  accept/decline DTMF buffers the same size. Just using 20 isn't
	  good enough when someone didn't get the memo. * Fix stupid use of
	  a global variable in FollowMe. (ynlongest) * Fix bit field
	  declarations in FollowMe. * Fix FollowMe n option documentation.

2012-05-08 15:48 +0000 [r365574]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Send more accurate identification
	  information in dialog-info SIP NOTIFYs. This uses the calling
	  channel's caller ID and connected line information to populate
	  the remote and local identities in the dialog-info NOTIFY when an
	  extension is ringing. There is a bit of an oddity here, and that
	  is that we seed the remote target with the To header of the
	  outbound call rather than the from header. This is because it was
	  reported that seeding with the from header caused hints to be
	  broken with certain SNOM devices. A comment has been added to the
	  code to explain this. (closes issue ASTERISK-16735) reported by
	  Maciej Krajewski patches: local_remote_hint2.diff uploaded by
	  Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
	  Michelson (license #5049) Tested by Niccolo Belli

2012-05-07 18:40 +0000 [r365476]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_config.c: Fix type punned compiler warning in
	  test_config.c

2012-05-07 18:36 +0000 [r365474]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c, main/pbx.c: Support VoiceMail d() option
	  when extension does not exist in channel's context The VoiceMail
	  d([c]) option is documented to accept digits for a new extension
	  in context <c>, if played during the greeting. This option works
	  fine if the extension being redirected to has an extension with
	  the same initial digit in the channel's current context. If that
	  digit did not happen to exist in some extension, a dialplan match
	  would fail and the user would not be redirected. This patch fixes
	  it such that if the <c> option is used, the extensions are
	  matched in that context as opposed to the caller's original
	  context. (closes issue ASTERISK-18243) Reported by: mjordan
	  Tested by: mjordan Review:
	  https://reviewboard.asterisk.org/r/1892

2012-05-07 16:01 +0000 [r365460]  Mark Michelson <mmichelson@digium.com>

	* main/audiohook.c, res/res_speech.c, channels/sig_analog.c,
	  main/abstract_jb.c, res/res_agi.c: Fix findings 0-3, 5, and 8 for
	  Coverity MISSING_BREAK errors. (Issue ASTERISK-19662)

2012-05-04 22:12 +0000 [r365398]  Kinsey Moore <kmoore@digium.com>

	* apps/app_followme.c, channels/chan_iax2.c,
	  channels/sip/config_parser.c, pbx/pbx_config.c,
	  apps/app_chanspy.c, apps/app_stack.c, main/config.c,
	  apps/app_voicemail.c, channels/chan_sip.c, funcs/func_aes.c,
	  main/features.c: Fix many issues from the NULL_RETURNS Coverity
	  report Most of the changes here are trivial NULL checks. There
	  are a couple optimizations to remove the need to check for NULL
	  and outboundproxy parsing in chan_sip.c was rewritten to avoid
	  use of strtok. Additionally, a bug was found and fixed with the
	  parsing of outboundproxy when "outboundproxy=," was set. (Closes
	  issue ASTERISK-19654)

2012-05-04 16:24 +0000 [r365313]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c: Fix local channel chains optimizing
	  themselves out of a call. * Made chan_local.c:check_bridge()
	  check the return value of ast_channel_masquerade(). In long
	  chains of local channels, the masquerade occasionally fails to
	  get setup because there is another masquerade already setup on an
	  adjacent local channel in the chain. * Made the outgoing local
	  channel (the ;2 channel) flush one voice or video frame per
	  optimization attempt. * Made sure that the outgoing local channel
	  also does not have any frames in its queue before the masquerade.
	  * Made do the masquerade immediately to minimize the chance that
	  the outgoing channel queue does not get any new frames added and
	  thus unconditionally flushed. * Made block indication -1 (Stop
	  tones) event when the local channel is going to optimize itself
	  out. When the call is answered, a chain of local channels pass
	  down a -1 indication for each bridge. This blizzard of -1 events
	  really slows down the optimization process. (closes issue
	  ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
	  Davis Review: https://reviewboard.asterisk.org/r/1894/

2012-05-04 15:48 +0000 [r365298]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c: Fix core FINDING 2, FINDING 3, and
	  FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
	  These three all are in RTP code that attempts to print the number
	  of sequence number cycles in an RTCP RR report. The code was
	  masking out the upper 16 bits and then shifting the number right
	  by 16 bits. This led to an all zero result in all cases. The fix
	  is to do the shift without the bit masking. (issue
	  ASTERISK-19649)

2012-05-03 14:54 +0000 [r365143-365159]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/h323/H323-MESSAGES.h,
	  addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
	  addons/ooh323c/src/ooh323.c: Fix warning of Coverity Static
	  analysis, change H225ProtocolIdentifier from value to pointer per
	  functions that use this. (close issue ASTERISK-19670) Reported
	  by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415)

	* addons/ooh323c/src/ooq931.c: Fix coverity static analysis
	  warning, allocate full ie structure instead of without data
	  buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
	  Patches: ASTERISK-19674.patch (License #5415)

2012-06-04  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.13.0 Released.

2012-05-30  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.13.0-rc2 Released.

	* Resolve crash in subscribing for MWI notifications.

	  ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
	  variable should definitely not be used after that. To solve this in
	  the two cases that affect subscribing for MWI notifications, we
	  instead save the ref locally, and unref them in the error
	  conditions.

	  (closes issue ASTERISK-19827)
	  Reported by: B. R.
	  Review: https://reviewboard.asterisk.org/r/1940/

	* AST-2012-007

	* AST-2012-008


2012-05-03  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.13.0-rc1 Released.

2012-05-02 17:02 +0000 [r365006-365068]  Terry Wilson <twilson@digium.com>

	* main/cel.c, channels/chan_local.c: Don't leak a ref if out of
	  memory and can't link the linkedid If the ao2_link fails, we are
	  most likely out of memory and bad things are going to happen.
	  Before those bad things happen, make sure to clean up the
	  linkedid references. This patch also adds a comment explaining
	  why linkedid can't be passed to both local channel allocations
	  and combines two ao2_ref calls into 1. Review:
	  https://reviewboard.asterisk.org/r/1895/

	* main/cel.c, channels/chan_local.c: Fix a CEL LINKEDID_END race
	  and local channel linkedids This patch has the ;2 channel inherit
	  the linkedid of the ;1 channel and fixes the race condition by no
	  longer scanning the channel list for "other" channels with the
	  same linkedid. Instead, cel.c has an ao2 container of linkedid
	  strings and uses the refcount of the string as a counter of how
	  many channels with the linkedid exist. Not only does this
	  eliminate the race condition, but it also allows us to look up
	  the linkedid by the hashed key instead of traversing the entire
	  channel list. Review: https://reviewboard.asterisk.org/r/1895/

2012-05-01 23:11 +0000 [r364902]  Richard Mudgett <rmudgett@digium.com>

	* main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
	  (closes issue ASTERISK-19755) Reported by: Gunther Kelleter
	  Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
	  Kelleter

2012-05-01 23:08 +0000 [r364899]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
	  error. As it turned out, this wasn't a huge deal. We were calling
	  ast_app_parse_options() for a set of options of which none took
	  arguments. The proper thing to do for this case is to pass NULL
	  for the "args" parameter here. We were instead passing a
	  seemingly-randomly chosen char * from the function. While this
	  would never get written to, you can rest assured things would
	  have gotten bad had new options (which took arguments) been added
	  to func_volume. (closes issue ASTERISK-19656)

2012-05-01 21:37 +0000 [r364841]  Jason Parker <jparker@digium.com>

	* main/manager.c: Prevent a potential crash when using manager
	  hooks. Found by me while poking at DPMA-127.

2012-05-01 21:36 +0000 [r364840]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c: * Fix error path resouce leak in
	  local_request(). * Restructure local_request() to reduce
	  indentation.

2012-05-01 19:03 +0000 [r364786]  Kinsey Moore <kmoore@digium.com>

	* apps/app_confbridge.c: Play conf-placeintoconf message to the
	  correct channel Correct the code in app_confbridge to play the
	  conf-placeintoconf message to the marked user entering the bridge
	  instead of to the conference while the marked user hears silence.
	  (closes issue ASTERISK-19641) Reported-by: Mark A Walters

2012-05-01 18:16 +0000 [r364769]  Jonathan Rose <jrose@digium.com>

	* main/app.c: Fix bad check in voicemail functions for
	  ast_inboxcount2_func Check looks for ast_inboxcount_func instead
	  of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
	  issue ASTERISK-19718) Reported by: Corey Farrell Patches:
	  ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
	  (license 5909)

2012-04-30 19:39 +0000 [r364706]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Revert improved identities sent in
	  dialog-info NOTIFY requests in r360862 Revision 360862 was
	  intended to improve identities sent in dialog-info NOTIFY
	  requests. Some users reported that hint became broken once this
	  was done. It's not clear exactly what part of the patch has
	  caused this regression, but broken hints are bad. For now, this
	  revision is being reverted so that the next releases of Asterisk
	  do not have bad behavior in them. The original reported issue
	  will have to be fixed differently in the next version of
	  Asterisk. (issue ASTERISK-16735)

2012-04-30 16:37 +0000 [r364649]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323cDriver.c: Fix use freed pointer in return value
	  from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
	  Patches: ASTERISK-19663-ooh323.patch (License #5415)

2012-04-30 15:51 +0000 [r364635]  Mark Murawki <markm@intellasoft.net>

	* main/logger.c: Sanatize result from bfd_find_nearest_line
	  (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
	  to null resulting in a crash when strrchr(file) runs (closes
	  issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
	  Murawski

2012-04-29 19:31 +0000 [r364578]  Matthew Jordan <mjordan@digium.com>

	* formats/format_g723.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_sln16.c,
	  formats/format_wav_gsm.c, formats/format_siren14.c,
	  formats/format_gsm.c, formats/format_g719.c,
	  formats/format_siren7.c, formats/format_g729.c,
	  formats/format_ilbc.c, formats/format_sln.c,
	  formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
	  Fix error that caused truncate operations to fail Another very
	  inappropriate placement of a ')' (again introduced in r362151)
	  caused the various truncate operations to attempt to truncate the
	  sound file at a position of '0'. (issue ASTERISK-19655) Reported
	  by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec

2012-04-27 21:48 +0000 [r364341]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Don't attempt to make use of the
	  dynamic_exclude_static ACL if DNS lookup fails. (closes issue
	  ASTERISK-18321) Reported by Dan Lukes Patches:
	  ASTERISK-18321.patch by Mark Michelson (license #5049)

2012-04-27 21:45 +0000 [r364340]  Terry Wilson <twilson@digium.com>

	* tests/test_config.c (added), main/config.c: Fix ast_parse_arg
	  numeric type range checking and add tests ast_parse_arg wasn't
	  checking for strto* parse errors or limiting the results by the
	  actual range of the numeric types. This patch fixes that and adds
	  unit tests as well. Review:
	  https://reviewboard.asterisk.org/r/1879/

2012-04-27 19:26 +0000 [r364277]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/time.h: Prevent overflow in calculation in
	  ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
	  attempts to calculate the difference, in milliseconds, between
	  two timeval structs, and return the difference in a 64-bit
	  integer. Unfortunately, it assumes that the long tv_sec/tv_usec
	  members in the timeval struct are large enough to hold the
	  calculated values before it returns. On 64-bit machines, this
	  might be the case, as a long may be 64-bits. On 32-bit machines,
	  however, a long may be less (32-bits), in which case, the
	  calculation can overflow. This overflow caused significant
	  problems in MixMonitor, which uses the method to determine if an
	  audio factory, which has not presented audio to an audiohook, is
	  merely late in providing said audio or will never provide audio.
	  In an overflow situation, the audiohook would incorrectly
	  determine that an audio factory that will never provide audio is
	  merely late instead. This led to situations where a MixMonitor
	  never recorded any audio. Note that this happened most frequently
	  when that MixMonitor was started by the ConfBridge application
	  itself, or when the MixMonitor was attached to a Local channel.
	  (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
	  Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
	  #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
	  Murawski Tested by: Michael L. Young Patches:
	  32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
	  (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
	  feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
	  https://reviewboard.asterisk.org/r/1889/

2012-04-27 18:57 +0000 [r364258]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Allow SIP pvts involved in Replaces
	  transfers to fall out of reference sooner Unref the SIP pvt
	  stored in the refer structure as soon as it is no longer needed
	  so that the pvt and associated file descriptors can be freed
	  sooner. This change makes a reference decrement unnecessary in
	  code that handles SIP BYE/Also transfers which should not touch
	  the reference anyway. (related to issue ASTERISK-19579)

2012-04-27 14:42 +0000 [r364203]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Allow for reloading SRTP crypto keys within
	  the same SIP dialog As a continuation of the patch in r356604,
	  which allowed for the reloading of SRTP keys in re-INVITE
	  transfer scenarios, this patch addresses the more common case
	  where a new key is requested within the context of a current SIP
	  dialog. This can occur, for example, when certain phones request
	  a SIP hold. Previously, once a dialog was associated with an SRTP
	  object, any subsequent attempt to process crypto keys in any SDP
	  offer - either the current one or a new offer in a new SIP
	  request - were ignored. This patch changes this behavior to only
	  ignore subsequent crypto keys within the current SDP offer, but
	  allows future SDP offers to change the keys. (issue
	  ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
	  Arimont Review: https://reviewboard.asteriskorg/r/1885/

2012-04-26 21:10 +0000 [r364060-364108]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_directed_pickup.c: Update Pickup application
	  documentation. (With feeling this time.)

	* main/features.c: Fix DTMF atxfer running h exten after the wrong
	  bridge ends. When party B does an attended transfer of party A to
	  party C, the attending bridge between party B and C should not be
	  running an h exten when the bridge ends. Running an h exten now
	  sets a softhangup flag to ensure that an AGI will run in dead AGI
	  mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B
	  channel for the attending bridge between party B and C. (closes
	  issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario
	  (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
	  Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario

2012-04-26 19:24 +0000 [r364046]  Terry Wilson <twilson@digium.com>

	* main/asterisk.c: Add more constness to the end_buf pointer in the
	  netconsole issue ASTERISK-18308 Review:
	  https://reviewboard.asterisk.org/r/1876/

2012-04-26 13:24 +0000 [r363986]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Fix reference leaks involving SIP Replaces
	  transfers The reference held for SIP blind transfers using the
	  Replaces header in an INVITE was never freed on success and also
	  failed to be freed in some error conditions. This caused a file
	  descriptor leak since the RTP structures in use at the time of
	  the transfer were never freed. This reference leak and another
	  relating to subscriptions in the same code path have now been
	  corrected. (closes issue ASTERISK-19579)

2012-04-26 09:44 +0000 [r363934]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_sip.c: chan_sip: [general] maxforwards, not checked
	  for a value greater than 255 The peer maxforwards is checked for
	  both '< 1' and '> 255', but the default 'maxforwards' in the
	  [general] section is only checked for '< 1' alecdavis (license
	  585) Reported by: alecdavis Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1888/

2012-04-26 03:11 +0000 [r363375-363875]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_directed_pickup.c: Update Pickup application
	  documentation. (Even better)

	* apps/app_directed_pickup.c: Update Pickup application
	  documentation.

	* channels/sig_pri.c, channels/chan_dahdi.c: Make
	  DAHDISendCallreroutingFacility wait 5 seconds for a reply before
	  disconnecting the call. Some switches may not handle the
	  call-deflection/call-rerouting message if the call is
	  disconnected too soon after being sent. Asteisk was not waiting
	  for any reply before disconnecting the call. * Added a 5 second
	  delay before disconnecting the call to wait for a potential
	  response if the peer does not disconnect first. (closes issue
	  ASTERISK-19708) Reported by: mehdi Shirazi Patches:
	  jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  Clear ISDN channel resetting state if the peer continues to use
	  it. Some ISDN switches occasionally fail to send a RESTART
	  ACKNOWLEDGE in response to a RESTART request. * Made the second
	  SETUP received after sending a RESTART request clear the channel
	  resetting state as if the peer had sent the expected RESTART
	  ACKNOWLEDGE before continuing to process the SETUP. The peer may
	  not be sending the expected RESTART ACKNOWLEDGE. (issue
	  ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
	  jira_ast_815_v1.8.patch (license #5621) patch uploaded by
	  rmudgett (modified)

	* main/features.c: Fix recalled party B feature flags for a failed
	  DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to
	  C 3) B hangs up 4) C does not answer 5) B is called back 6) B
	  answers 7) B cannot initiate transfers anymore * Add dial
	  features datastore to recalled party B channel that is a copy of
	  the original party B channel's dial features datastore. *
	  Extracted add_features_datastore() from
	  add_features_datastores(). * Renamed struct ast_dial_features
	  features_caller and features_callee members to my_features and
	  peer_features respectively. These better names eliminate the need
	  for some explanatory comments. * Simplified code accessing the
	  struct ast_dial_features datastore. (closes issue ASTERISK-19383)
	  Reported by: lgfsantos

	* main/features.c: Hangup affected channel in error paths of
	  bridge_call_thread().

2012-04-23 16:02 +0000 [r363209]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/astfd.c: On some platforms, O_RDONLY is not a flag to be
	  checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
	  specification does not mandate how these 3 flags must be
	  specified, only that one of the three must be specified in every
	  call.

2012-04-23 14:33 +0000 [r363141]  Jonathan Rose <jrose@digium.com>

	* main/manager.c, /: AST-2012-004: Fix an error that allows AMI
	  users to run shell commands sans authorization. As detailed in
	  the advisory, AMI users without write authorization for SYSTEM
	  class AMI actions were able to run system commands by going
	  through other AMI commands which did not require that
	  authorization. Specifically, GetVar and Status allowed users to
	  do this by setting their variable/s options to the SHELL or EVAL
	  functions. Also, within 1.8, 10, and trunk there was a similar
	  flaw with the Originate action that allowed users with originate
	  permission to run MixMonitor and supply a shell command in the
	  Data argument. That flaw is fixed in those versions of this
	  patch. (closes issue ASTERISK-17465) Reported By: David Woolley
	  Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
	  (license 6182) ........ Merged revisions 363117 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2012-04-23 14:05 +0000 [r363102-363106]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE handling
	  when no channel owner exists If Asterisk receives a SIP UPDATE
	  request after a call has been terminated and the channel has been
	  destroyed but before the SIP dialog has been destroyed, a
	  condition exists where a connected line update would be attempted
	  on a non-existing channel. This would cause Asterisk to crash.
	  The patch resolves this by first ensuring that the SIP dialog has
	  an owning channel before attempting a connected line update. If
	  an UPDATE request is received and no channel is associated with
	  the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
	  Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
	  ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
	  6283)

	* /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
	  heap overflow in keypad button handling When handling a keypad
	  button message event, the received digit is placed into a fixed
	  length buffer that acts as a queue. When a new message event is
	  received, the length of that buffer is not checked before placing
	  the new digit on the end of the queue. The situation exists where
	  sufficient keypad button message events would occur that would
	  cause the buffer to be overrun. This patch explicitly checks that
	  there is sufficient room in the buffer before appending a new
	  digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
	  ........ Merged revisions 363100 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2012-04-21 01:44 +0000 [r362997]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Update app_dial M and U option GOTO return value
	  documentation.

2012-04-20 16:09 +0000 [r362815-362868]  Terry Wilson <twilson@digium.com>

	* main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
	  (closes issue ASTERISK-19758) Reported by: Barry Miller Tested
	  by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
	  (license 5434)

	* apps/app_speech_utils.c: Document Speech* apps hangup on failure
	  and suggest TryExec The Speech API apps return -1 on failure,
	  which will hang up the channel. This may not be desirable
	  behavior for some, but it isn't something that can be changed
	  without breaking people's dialplans or writing an option to all
	  of the Speech apps that does what TryExec already does. This
	  patch documents the hangup behavior of the apps, and suggests
	  TryExec as the solution. (closes issue AST-813)

2012-04-19 21:58 +0000 [r362729]  Walter Doekes <walter+asterisk@wjd.nu>

	* funcs/func_version.c: Fix documentation for
	  ${VERSION(ASTERISK_VERSION_NUM)}.

2012-04-19 21:05 +0000 [r362680]  Michael L. Young <elgueromexicano@gmail.com>

	* tests/test_linkedlists.c, tests/test_poll.c: Add leading and
	  trailing backslashes A couple of unit tests did not have have
	  leading or trailing backslashes when setting their test category
	  resulting in a warning message being displayed. Added the
	  backslash where needed.

2012-04-19 20:59 +0000 [r362677]  Richard Mudgett <rmudgett@digium.com>

	* configs/queues.conf.sample: Update membermacro and membergosub
	  documentation in queues.conf.sample.

2012-04-19 15:53 +0000 [r362586]  Sean Bright <sean@malleable.com>

	* apps/app_externalivr.c: Prevent a crash in ExternalIVR when the
	  'S' command is sent first. If the first command sent from an
	  ExternalIVR client is an 'S' command, we were blindly removing
	  the first element from the play list and deferencing it, even if
	  it was NULL. This corrects that and also locks appropriately in
	  one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski

2012-04-19 14:26 +0000 [r362536]  Terry Wilson <twilson@digium.com>

	* main/asterisk.c: Handle multiple commands per connection via
	  netconsole Asterisk would accept multiple NULL-delimited CLI
	  commands via the netconsole socket, but would occasionally miss a
	  command due to the command not being completely read into the
	  buffer. This patch ensures that any partial commands get moved to
	  the front of the read buffer, appended to, and properly sent.
	  (closes issue ASTERISK-18308) Review:
	  https://reviewboard.asterisk.org/r/1876/

2012-04-19 02:08 +0000 [r362485]  Matthew Jordan <mjordan@digium.com>

	* apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
	  addons/chan_mobile.c, main/asterisk.c, channels/chan_unistim.c,
	  main/frame.c, main/tdd.c, main/jitterbuf.c: Fix a variety of
	  potential buffer overflows * chan_mobile: Fixed an overrun where
	  the cind_state buffer (an integer array of size 16) would be
	  overrun due to improper bounds checking. At worst, the buffer can
	  be overrun by a total of 48 bytes (assuming 4-byte integers),
	  which would still leave it within the allocated memory of struct
	  hfp. This would corrupt other elements in that struct but not
	  necessarily cause any further issues. * app_sms: The array imsg
	  is of size 250, while the array (ud) that the data is copied into
	  is of size 160. If the size of the inbound message is greater
	  then 160, up to 90 bytes could be overrun in ud. This would
	  corrupt the user data header (array udh) adjacent to ud. *
	  chan_unistim: A number of invalid memmoves are corrected. These
	  would move data (which may or may not be valid) into the ends of
	  these buffers. * asterisk: ast_console_toggle_loglevel does not
	  check that the console log level being set is less then or equal
	  to the allowed log levels of 32. * frame: In
	  ast_codec_pref_prepend, if any occurrence of the specified codec
	  is not found, the value used to index into the array pref->order
	  would be one greater then the maximum size of the array. *
	  jitterbuf: If the element being placed into the jitter buffer
	  lands in the last available slot in the jitter history buffer,
	  the insertion sort attempts to move the last entry in the buffer
	  into one slot past the maximum length of the buffer. Note that
	  this occurred for both the min and max jitter history buffers. *
	  tdd: If a read from fsk_serial returns a character that is
	  greater then 32, an attempt to read past one of the statically
	  defined arrays containing the values that character maps to would
	  occur. * localtime: struct ast_time and tm are not the same size
	  - ast_time is larger, although it contains the elements of tm
	  within it in the same layout. Hence, when using memcpy to copy
	  the contents of tm into ast_time, the size of tm should be used,
	  as opposed to the size of ast_time. * extconf: this treats
	  ast_timing's minmask array as if it had a length of 48, when it
	  has defined the size of the array as 24. pbx.h defines minmask as
	  having a size of 48. (issue ASTERISK-19668) Reported by: Matt
	  Jordan

2012-04-18 16:20 +0000 [r362428]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample: Add ability to ignore layer 1
	  alarms for BRI PTMP lines. Several telcos bring the BRI PTMP
	  layer 1 down when the line is idle. When layer 1 goes down,
	  Asterisk cannot make outgoing calls. Incoming calls could fail as
	  well because the alarm processing is handled by a different code
	  path than the Q.931 messages. * Add the layer1_presence
	  configuration option to ignore layer 1 alarms when the telco
	  brings layer 1 down. This option can be configured by span while
	  the similar DAHDI driver teignorered=1 option is system wide.
	  This option unlike layer2_persistence does not require libpri
	  v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845

2012-04-17 21:18 +0000 [r362355-362368]  Matthew Jordan <mjordan@digium.com>

	* main/frame.c: Handle case where an unknown format is used to get
	  the preferred codec size In ast_codec_pref_getsize, if an unknown
	  format is passed to the method, no preferred codec will be
	  selected and a negative number will be used to index into the
	  format list. The method now logs an unknown format as a warning,
	  and returns an empty format list. (issue ASTERISK-19655) Reported
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/

	* res/res_musiconhold.c, res/res_rtp_asterisk.c, res/res_agi.c: Fix
	  places in resources where a negative return value could impact
	  execution This patch addresses a number of modules in resources
	  that did not handle the negative return value from function calls
	  adequately. This includes: * res_agi.c: if the result of the read
	  function is a negative number, indicating some failure, the
	  result would instead be treated as the number of bytes read. This
	  patch now treats negative results in the same manner as an end of
	  file condition, with the exception that it also logs the error
	  code indicated by the return. * res_musiconhold.c: if spawn_mp3
	  fails to assign a file descriptor to srcfd, and instead assigns a
	  negative value, that file descriptor could later be passed to
	  functions that require a valid file descriptor. If spawn_mp3
	  fails, we now immediately retry instead of continuing in the
	  logic. * res_rtp_asterisk.c: if no codec can be matched between
	  two RTP instances in a peer to peer bridge, we immediately return
	  instead of attempting to use the codec payload type as an index
	  to determine the appropriate negotiated codec. (issue
	  ASTERISK-19655) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/

	* main/asterisk.c, main/manager.c, main/translate.c: Fix places in
	  main where a negative return value could impact execution This
	  patch addresses a number of modules in main that did not handle
	  the negative return value from function calls adequately, or were
	  not sufficiently clear that the conditions leading to improper
	  handling of the return values could not occur. This includes: *
	  asterisk.c: A negative return value from the read function would
	  be used directly as an index into a buffer. We now check for
	  success of the read function prior to using its result as an
	  index. * manager.c: Check for failures in mkstemp and lseek when
	  handling the temporary file created for processing data returned
	  from a CLI command in action_command. Also check that the result
	  of an lseek is sanitized prior to using it as the size of a
	  memory map to allocate. * translate.c: Note in the appropriate
	  locations where powerof cannot return a negative value, due to
	  proper checks placed on the inputs to that function. (issue
	  ASTERISK-19655) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/

	* funcs/func_env.c: Fix places where a negative return from ftello
	  could be used as invalid input In a variety of locations in both
	  reading and writing a file, the result from the C library
	  function ftello is used as input to other functions. For the
	  parameters and functions in question, a negative value is invalid
	  input. This patch checks the return value from the ftello
	  function to determine if we were able to determine the current
	  position in the file stream and, if not, fail gracefully. (issue
	  ASTERISK-19655) Reported by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1863/

2012-04-17 20:43 +0000 [r362354]  Jonathan Rose <jrose@digium.com>

	* main/utils.c, res/res_config_curl.c, res/res_config_pgsql.c,
	  res/res_config_odbc.c: Make use of va_args more appropriate to
	  form in various res_config modules plus utils. A number of
	  va_copy operations weren't matched with a corresponding va_end in
	  res_config_odbc. Also, there was a potential for va_end to be
	  invoked twice on the same va_arg in utils, which would mean
	  invoking va_end on an undefined variable... which is bad. va_end
	  is removed from various functions in config_pgsql and config_curl
	  since they aren't making their own copy. The invokers of those
	  functions are responsible for calling va_end on them. (issue
	  ASTERISK-19451) Reported by: Walter Doekes Review:
	  https://reviewboard.asterisk.org/r/1848/

2012-04-17 18:25 +0000 [r362304]  Matthew Jordan <mjordan@digium.com>

	* formats/format_sln16.c, formats/format_wav_gsm.c,
	  formats/format_siren14.c, formats/format_gsm.c,
	  formats/format_g719.c, formats/format_siren7.c,
	  formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
	  formats/format_pcm.c: Fix error that caused seek format
	  operations to set max file size to '1' or '0' A very
	  inappropriate placement of a ')' (introduced in r362151) caused
	  the maximum size of a file to be set as the result of a
	  comparison operation, as opposed to the result of the ftello
	  operation. This resulted in seeking being restricted to the
	  beginning of the file, or 1 byte into the file. Thanks to the
	  Asterisk Test Suite for properly freaking out about this on at
	  least one test. (issue ASTERISK-19655) Reported by: Matt Jordan

2012-04-17 02:37 +0000 [r362253]  Michael L. Young <elgueromexicano@gmail.com>

	* channels/chan_sip.c: Turn off warning message when bind address
	  is set to any. When a bind address is set to an ANY address
	  (udpbindport=::), a warning message is displayed stating that
	  "Address remapping activated in sip.conf but we're using IPv6,
	  which doesn't need it. Please remove 'localnet' and/or
	  'externaddr' settings." But if one is running dual stack, we
	  shouldn't be told to turn those settings off. This patch checks
	  if the bind address is an ANY address or not. The warning message
	  will now only be displayed if the bind address is NOT an ANY
	  address and IPv6 is being used. Also, updated the copyright year.
	  (closes issue ASTERISK-19456) Reported by: Michael L. Young
	  Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
	  uploaded by Michael L. Young (license 5026)

2012-04-16 21:56 +0000 [r362151-362204]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_dahdi.c, channels/chan_agent.c: Fix negative return
	  handling in channel drivers In chan_agent, while handling a
	  channel indicate, the agent channel driver must obtain a lock on
	  both the agent channel, as well as the channel the agent channel
	  is using. To do so, it attempts to lock the other channel first,
	  then unlock the agent channel which is locked prior to entry into
	  the indicate handler. If this unlock fails with a negative return
	  value, which can occur if the object passed to agent_indicate is
	  an invalid ao2 object or is NULL, the return value is passed
	  directly to strerror, which can only accept positive integer
	  values. In chan_dahdi, the return value of dahdi_get_index is
	  used to directly index into the sub-channel array. If
	  dahd_get_index returns a negative value, it would use that value
	  to index into the array, which could cause an invalid memory
	  access. If dahdi_get_index returns a negative number, we now
	  default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863/

	* apps/app_voicemail.c: Fix handling of negative return code when
	  storing voicemails in ODBC storage When storing a voicemail
	  message using an ODBC connection to a database, the voicemail
	  message is first stored on disk. The sound file associated with
	  the message is read into memory before being transmitted to the
	  database. When this occurs, a failure in the C library's lseek
	  function would cause a negative value to be passed to the mmap as
	  the size of the memory map to create. This would almost certainly
	  cause the creation of the memory map to fail, resulting in the
	  message being lost. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863

	* formats/format_g723.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_sln16.c,
	  formats/format_wav_gsm.c, formats/format_siren14.c,
	  formats/format_gsm.c, formats/format_g719.c,
	  formats/format_siren7.c, formats/format_g729.c,
	  formats/format_ilbc.c, formats/format_sln.c,
	  formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
	  Check for IO stream failures in various format's truncate/seek
	  operations For the formats that support seek and/or truncate
	  operations, many of the C library calls used to determine or set
	  the current position indicator in the file stream were not being
	  checked. In some situations, if an error occurred, a negative
	  value would be returned from the library call. This could then be
	  interpreted inappropriately as positional data. This patch checks
	  the return values from these library calls before using them in
	  subsequent operations. (issue ASTERISK-19655) Reported by: Matt
	  Jordan Review: https://reviewboard.asterisk.org/r/1863/

2012-04-13 15:54 +0000 [r362079-362082]  Jonathan Rose <jrose@digium.com>

	* apps/app_forkcdr.c: Make ForkCDR e option not set end time of the
	  newly forked CDR log Prior to this patch, ForkCDR's e option
	  would immediately set the end time of the forked CDR to that of
	  the CDR that is being terminated. This resulted in the new CDR's
	  end time being roughly the same as it's beginning time (which is
	  in turn roughly the same as the original's end time). (closes
	  issue ASTERISK-19164) Reported by: Steve Davies Patches:
	  cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)

	* apps/app_meetme.c: Send relative path named recordings to the
	  meetme directory instead of sounds Prior to this patch, no effort
	  was made to parse the path name to determine a proper destination
	  for recordings of MeetMe's r option. This fixes that. Review:
	  https://reviewboard.asterisk.org/r/1846/

2012-04-12 16:18 +0000 [r361955-361972]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_iax2.c: Make trunkfreq take effect when set
	  Previously, setting trunkfreq had no effect on initial load or on
	  reload and only ever used the default value. This causes
	  trunkfreq to be used appropriately on initial load and reload.
	  (closes issue ASTERISK-19521) Patch-by: Jaco Kroon

	* Makefile.rules, makeopts.in, codecs/lpc10/Makefile, Makefile,
	  build_tools/cflags.xml, build_tools/menuselect-deps.in,
	  codecs/gsm/src/k6opt.s, configure, codecs/gsm/Makefile,
	  configure.ac: Simplify build system architecture optimization
	  This change to the build system rips out any usage of PROC along
	  with architecture-specific optimizations in favor of using
	  -march=native where it is supported. This fixes broken builds on
	  64bit Intel systems and results in better optimized code on
	  systems running GCC 4.2+. Review:
	  https://reviewboard.asterisk.org/r/1852/ (closes issue
	  ASTERISK-19462)

2012-04-10 21:43 +0000 [r361854]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Prevent invalid access of free'd memory if
	  DAHDI channel during an MWI event In the MWI processing loop,
	  when a valid event occurs the temporary caller ID information is
	  deallocated. If a new DAHDI channel is successfully created, the
	  event is passed up to the analog_ss_thread without error and the
	  loop exits. If, however, the DAHDI channel is not created, then
	  the caller ID struct has been free'd, and the gains reset to
	  their previous level. This will almost certainly cause an invalid
	  access to the free'd memory, either in subsequent calls to
	  callerid_free or calls to callerid_feed. * Rework the -r361705
	  patch to better manage the cs and mtd allocated resources. *
	  Fixed use of mwimonitoractive flag to be correct if the
	  mwi_thread() fails to start.

2012-04-10 19:57 +0000 [r361657-361803]  Matthew Jordan <mjordan@digium.com>

	* main/http.c: Fix crash caused by unloading or reloading of
	  res_http_post When unlinking itself from the registered HTTP
	  URIs, res_http_post could inadvertently free all URIs registered
	  with the HTTP server. This patch modifies the unregister method
	  to only free the URI that is actually being unregistered, as
	  opposed to all of them.

	* funcs/func_curl.c: Allow func_curl to exit gracefully if list
	  allocation fails during write If the global_curl_info data
	  structure could not be allocated, the datastore associated with
	  the operation would be free'd, but the function would not return.
	  This would later dereference the datastore, almost certainly
	  causing Asterisk to crash. With this patch, if the data structure
	  is not allocated the method will return an error code, and not
	  attempt any further operation.

	* channels/chan_dahdi.c: Prevent invalid access of free'd memory if
	  DAHDI channel during an MWI event In the MWI processing loop,
	  when a valid event occurs the temporary caller ID information is
	  deallocated. If a new DAHDI channel is successfully created, the
	  event is passed up to the analog_ss_thread without error and the
	  loop exits. If, however, the DAHDI channel is not created, then
	  the caller ID struct has been free'd, and the gains reset to
	  their previous level. This will almost certainly cause an invalid
	  access to the free'd memory, either in subsequent calls to
	  callerid_free or calls to callerid_feed. This patch makes it so
	  that we only free the caller ID structure if a DAHDI channel is
	  successfully created, and we bump the gains back up if we fail to
	  make a DAHDI channel.

	* funcs/func_global.c: Change SHARED function to use a safe
	  traversal when modifying a variable When the SHARED function
	  modifies a variable, it removes it from its list of variables and
	  reinserts the new value at the head of the list of variables.
	  Doing this inside a standard list traversal can be dangerous, as
	  the standard list traversal does not account for the list being
	  changed. While the code in question should not cause a use after
	  free violation due to its breaking out of the loop after freeing
	  the variable, it could lead to a maintenance issue if the loop
	  was modified. This also fixes a violation reported by a static
	  analysis tool, which also makes this code easier to maintain in
	  the future.

2012-04-06 21:50 +0000 [r361558-361606]  Matthew Jordan <mjordan@digium.com>

	* res/res_calendar_ews.c: Fix memory leak in res_calendar_ews when
	  event email address node is empty If the XML calendar data
	  returned by a Microsoft Exchange Web Service specifies an XML
	  Event E-Mail Address ("EmailAddress"), and no e-mail address is
	  provided, a condition existed where an ast_calendar_attendee
	  struct would be allocated but not appended to the list of
	  attendees. Because of that, the memory associated with the
	  attendee would never be freed. This patch frees the memory if no
	  e-mail address is provided.

	* apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
	  option with user specified A memory leak/reference counting leak
	  occurs if the MeetMeAdmin 'e' command (eject last user that
	  joined) is used in conjunction with a specified user. Regardless
	  of the command being executed, if a user is specified for the
	  command, MeetMeAdmin will look up that user. Because the 'e'
	  option kicks the last user that joined, as opposed to the one
	  specified, the reference to the user specified by the command
	  would be leaked when the user variable was assigned to the last
	  user that joined.

2012-04-06 18:09 +0000 [r361471]  Kinsey Moore <kmoore@digium.com>

	* apps/app_ices.c, channels/chan_gtalk.c, channels/chan_iax2.c,
	  res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c,
	  main/tcptls.c, funcs/func_channel.c, channels/console_gui.c,
	  apps/app_sms.c, apps/app_chanspy.c, addons/chan_mobile.c,
	  channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
	  res/res_clioriginate.c, channels/chan_unistim.c, main/pbx.c,
	  channels/chan_sip.c, res/res_fax.c, funcs/func_strings.c,
	  channels/console_video.c, formats/format_ogg_vorbis.c: Add
	  missing newlines to CLI logging

2012-04-06 16:27 +0000 [r361403-361412]  Paul Belanger <paul.belanger@polybeacon.com>

	* funcs/func_sysinfo.c: Fix typo in svn:keywords

	* bridges/bridge_multiplexed.c, bridges/bridge_builtin_features.c:
	  Fix typo in svn:keywords

2012-04-06 15:47 +0000 [r361380]  Russell Bryant <russell@russellbryant.com>

	* apps/rpt_flow.pdf (removed), configs/rpt.conf.sample (removed),
	  configs/usbradio.conf.sample (removed): Remove a few more files
	  related to chan_usbradio and app_rpt.

2012-04-06 14:01 +0000 [r361332]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Fix a typo in the warning messages for an
	  ignored media stream Added a '\n' to the warning messages when we
	  ignore a media stream due to the port number being '0'. (closes
	  issue ASTERISK-19646) Reported by: Badalian Vyacheslav

2012-04-06 13:30 +0000 [r361329]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c: Remove unnecessary error message in app_dial.c
	  The error message for failure to stop autoservice after a gosub
	  or macro call during a dial was removed for macro while Asterisk
	  1.4 was still being actively developed. The corresponding gosub
	  error message was never removed. (closes issue ASTERISK-19551)

2012-04-05 16:36 +0000 [r361201-361269]  Jonathan Rose <jrose@digium.com>

	* apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
	  uses the class if it's been defined There were a few instances of
	  restarting music on hold in meetme that would cause Asterisk to
	  revert to the default class of music on hold for no adequate
	  reason. Review: https://reviewboard.asterisk.org/r/1844/

	* addons/ooh323cDriver.c: Fix some stuff involving calls to memcpy
	  and memset The important parts of the patch were already applied
	  through other updates. (closes issue ASTERISK-19445) Reported by:
	  Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto
	  Dei (license 5027)

	* funcs/func_devstate.c: Make 'help devstate change' display
	  properly (get rid of excess comma) (closes issue ASTERISK-19444)
	  Reported by: Makoto Dei Patches:
	  devstate-change-usage-truncate.patch uploaded by Makoto Dei
	  (license 5027)

2012-05-02  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.12.0 Released.

2012-05-01  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.12.0-rc3 Released.

        * channels/chan_sip.c: Revert revision 360862

        Revision 360862 was intended to improve identities sent in
	dialog-info NOTIFY requests. Some users reported that hint became
	broken once this was done. It's not clear exactly what part of
	the patch has caused this regression, but broken hints are bad.

        For now, this revision is being reverted so that the next releases of
        Asterisk do not have bad behavior in them.  The original reported
 	issue will have to be fixed differently in the next version of
	Asterisk.

	(issue ASTERISK-16735)

2012-04-24  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.12.0-rc2 Released.

	* AST-2012-004

	* AST-2012-005

	* AST-2012-006

2012-04-04  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.12.0-rc1 Released.

2012-04-04 16:29 +0000 [r361090-361142]  Jonathan Rose <jrose@digium.com>

	* main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
	  channels/chan_iax2.c, apps/app_milliwatt.c, main/channel.c,
	  pbx/pbx_loopback.c, addons/chan_ooh323.c, channels/chan_sip.c:
	  Replace GNU old-style field designator extensions to fix clang
	  warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
	  clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)

	* apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
	  nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
	  by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/

2012-04-03 20:08 +0000 [r360987-361040]  Kinsey Moore <kmoore@digium.com>

	* apps/app_transfer.c: Fix the display of documentation for
	  Transfer This came up while fixing documentation generation for
	  many other cases where the argument separator was not being
	  displayed properly. Now that it is displayed properly, it shows
	  up in the wrong place for Transfer since the '/' is only required
	  if Tech is present. (related to issue ASTERISK-18168)

	* channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
	  This change prevents Asterisk from sending RTCP receiver reports
	  during a remote bridge since it is no longer receiving media and
	  should not be reporting anything. (related to ASTERISK-19366)

2012-03-30 21:26 +0000 [r360933]  Richard Mudgett <rmudgett@digium.com>

	* main/logger.c: Fix logger deadlock on Asterisk shutdown. The
	  logger_thread() had an exit path that failed to release the
	  logmsgs list lock. * Make logger_thread() exit path unlock the
	  logmsgs list lock. * Made ast_log() not queue any messages to the
	  logmsgs list if the close_logger_thread flag is set. (issue
	  ASTERISK-19463) Reported by: Matt Jordan

2012-03-29 23:32 +0000 [r360862-360884]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Fix potential race condition during call pickup.
	  Prior to this patch, a connected line update was queued during
	  call pickup and then an answer frame was queued. The original
	  caller would presumably then have his connected line updated and
	  then the call would be answered. In actuality, the answer frame
	  was not how the call ended up being answered. Rather, an odd
	  section in app_dial that checks if the called channel's state is
	  up. The result is that the order of the connected line update and
	  the answer were variable. In most cases, this wasn't actually a
	  bad thing. However, if the 'I' option was passed to dial, the
	  connected line update would be inhibited. The fix is to queued
	  the connected line after the answer frame is queued. This way the
	  race in app_dial is between two conditions resulting in an
	  answer. This way the connected line update occurs after the
	  answer every time. (closes issue ASTERISK-19183) Reported by:
	  Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches:
	  ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)

	* channels/chan_sip.c: Improve accuracy of identifying information
	  sent in dialog-info SIP NOTIFY requests. This change makes use of
	  connected party information in addition to caller ID in order to
	  populate local and remote XML elements in the dialog-info
	  NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej
	  Krajewski Tested by: Maciej Krajewski Patches:
	  local_remote_hint2.diff uploaded by Mark Michelson (license 5049)

2012-03-28 19:06 +0000 [r360712]  Terry Wilson <twilson@digium.com>

	* cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
	  channels/chan_gtalk.c, channels/chan_jingle.c,
	  addons/chan_ooh323.c: Destroy configs when they are no longer
	  used https://reviewboard.asterisk.org/r/1834/

2012-03-27 16:59 +0000 [r360625]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Make a debug message regarding subscription
	  changes more accurate. I was getting confused during some testing
	  why Asterisk was saying that a subscription was being added when
	  it was clearly being removed. This fixes that confusion.

2012-03-27 14:32 +0000 [r360488-360574]  Jonathan Rose <jrose@digium.com>

	* configure: Updates config with bootstrap where I changed
	  configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
	  Clark

	* configure.ac: Fix BETTER_BACKTRACES library detection for
	  Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
	  Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
	  Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
	  uploaded by Bryon Clark (license 6157)

2012-03-26 18:37 +0000 [r360471-360474]  Paul Belanger <pabelanger@digium.com>

	* CHANGES: Update CHANGES for r360471

	* CHANGES: Fix Asterisk version typo

	* main/dnsmgr.c: Increase verbosity level for ast_verb messages
	  While this does not fix the issue of the CLI being flooded by
	  'doing dnsmgr_lookup' messages, increasing the verbosity level
	  above 5 should help minimize it.

2012-03-24 23:46 +0000 [r360356-360413]  Russell Bryant <russell@russellbryant.com>

	* funcs/func_curl.c: func_curl: Fix leak of an ast_str in error
	  handling code path.

	* apps/app_page.c: app_page: Fix a memory leak on every Page().
	  dial_list is a dynamically allocated array that is allocated at
	  the beginning of Page() based on how many devices will be dialed.
	  This was never being freed.

	* apps/app_jack.c: app_jack: fix datastore memory leak in error
	  handling path.

	* res/ael/ael.tab.h, main/ast_expr2.c, main/ast_expr2.h,
	  res/ael/ael.tab.c, main/ast_expr2f.c, res/ael/ael_lex.c: Rebuild
	  parsers. This is needed to include the last fix to
	  main/ast_expr2.y. The changes look much bigger as this
	  regeneration of the code was done with newer versions of flex and
	  bison.

	* main/ast_expr2.y: expression parser: Fix (theoretical) memory
	  leak. Fix a memory leak that is very unlikely to actually happen.
	  If a malloc() succeeded, but the following strdup() failed, the
	  memory from the original malloc() would be leaked.

2012-03-24 00:35 +0000 [r360262-360309]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, main/channel.c: Make number not available
	  presentation also set screening to network provided. Q.951
	  indicates that when the presentation indicator is "Number not
	  available due to interworking" for a number then the screening
	  indicator field should be "Network provided". * Made
	  ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
	  when the presentation is "Number not available due to
	  interworking". This fix makes Asterisk consistent and it also
	  makes it consistent with earlier branches as far as this
	  presentation value is concerned. * Made pri_to_ast_presentation()
	  and ast_to_pri_presentation() conversions handle the "Number not
	  available due to interworking" case better in sig_pri.c. This
	  change is possible because the minimum required libpri version
	  (v1.4.11) has the necessary defines in libpri.h.

	* channels/chan_sip.c: Add missing initialization of
	  update_redirecting in chan_sip.c

2012-03-21 14:51 +0000 [r360138]  Jonathan Rose <jrose@digium.com>

	* contrib/scripts/install_prereq: Update install_prereq script to
	  include missing GSM library for debian amd move SQLite3. (closes
	  issue ASTERISK-19367) Reported by: Andrew Latham Patches:
	  debian_install_prereq.diff uploaded by Andrew Latham (license
	  5985)

2012-03-21 13:19 +0000 [r360087]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configure, configure.ac: Also detect gmime 2.6 Also detect gmime
	  version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License
	  #5035) <tzafrir.cohen@xorcom.com>

2012-03-21 13:19 +0000 [r360086]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on
	  the final response to a re-INVITE When Asterisk detects a hangup
	  and cannot send a BYE due to a pending INVITE, it sets the
	  pendingbye flag and waits for the final response to that INVITE.
	  When the response is received, it transmits the BYE. If, however,
	  that INVITE request is a pending re-INVITE, it needs to first
	  send a CANCEL request to terminate the pending re-INVITE. In that
	  circumstance, Asterisk was, in some scenarios, clearing the
	  pendingbye flag after processing the CANCEL request and not
	  checking for a pending BYE when receiving the final 487 response
	  to the INVITE. This patch ensures that if the pendingbye flag is
	  set, it is honored regardless of the nature of the INVITE request
	  currently in flight. (closes issue ASTERISK-19365) Reported by:
	  Thomas Arimont Tested by: Thomas Arimont Patches:
	  bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
	  6283) Review: https://reviewboard.asterisk.org/r/1807

2012-03-20 20:32 +0000 [r360033]  Kinsey Moore <kmoore@digium.com>

	* apps/app_echo.c: Prevent Echo() from relaying control, null, and
	  modem frames Echo()'s description states that it echoes audio,
	  video, and DTMF except for # while it actually echoes any frame
	  that it receives other than DTMF #. This was causing frame storms
	  in the test suite in some circumstances where Echo() was attached
	  to both ends of a pair of local channels and control frames were
	  being periodically generated. Echo()'s behavior and description
	  have been modifed so that it only echoes media and non-# DTMF
	  frames.

2012-03-20 17:21 +0000 [r359979]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/manager.h, main/manager.c: Allow AMI action
	  callback to be reentrant. Fix AMI module reload deadlock
	  regression from ASTERISK-18479 when it tried to fix the race
	  between calling an AMI action callback and unregistering that
	  action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
	  Locking the ao2 object guaranteed that there were no active
	  callbacks that mattered when ast_manager_unregister() was called.
	  Unfortunately, this causes the deadlock situation. The patch
	  stops locking the ao2 object to allow multiple threads to invoke
	  the callback re-entrantly. There is no way to guarantee a module
	  unload will not crash because of an active callback. The code
	  attempts to minimize the chance with the registered flag and the
	  maximum 5 second delay before ast_manager_unregister() returns.
	  The trunk version of the patch changes the API to fix the race
	  condition correctly to prevent the module code from unloading
	  from memory while an action callback is active. * Don't hold the
	  lock while calling the AMI action callback. (closes issue
	  ASTERISK-19487) Reported by: Philippe Lindheimer Review:
	  https://reviewboard.asterisk.org/r/1818/ Review:
	  https://reviewboard.asterisk.org/r/1820/

2012-03-16 20:13 +0000 [r359892]  Jonathan Rose <jrose@digium.com>

	* apps/app_chanspy.c: Prevent chanspy from binding to zombie
	  channels This patch addresses a bug with chanspy on local
	  channels which roughly 50% of the time would create a situation
	  where chanspy can latch onto a zombie channel, keeping the zombie
	  alive forever and causing the channel doing the spying to never
	  be able to hang up. (closes issue ASTERISK-19493) Reported by:
	  lvl Review: https://reviewboard.asterisk.org/r/1819/

2012-03-16 08:22 +0000 [r359809]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sip/include/sip.h: Missed lastinvite CSeq int to
	  uint32_t change from Review:
	  https://reviewboard.asterisk.org/r/1699/

2012-03-15 19:01 +0000 [r359656-359706]  Matthew Jordan <mjordan@digium.com>

	* main/utils.c: Fix remotely exploitable stack overflow in HTTP
	  manager There exists a remotely exploitable stack buffer overflow
	  in HTTP digest authentication handling in Asterisk. The
	  particular method in question is only utilized by HTTP AMI. When
	  parsing the digest information, the length of the string is not
	  checked when it is copied into temporary buffers allocated on the
	  stack. This patch fixes this behavior by parsing out pre-defined
	  key/value pairs and avoiding unnecessary copies to the stack.
	  (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
	  by: Matt Jordan

	* apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun
	  in Milliwatt Milliwatt is vulnerable to a remotely exploitable
	  stack overrun when using the 'o' option. This occurs due to the
	  milliwatt_generate function not accounting for
	  AST_FRIENDLY_OFFSET when calculating the maximum number of
	  samples it can put in the output buffer. This patch resolves this
	  issue by taking into account AST_FRIENDLY_OFFSET when determining
	  the maximum number of samples allowed. Note that at no point is
	  remote code execution possible. The data that is written into the
	  buffer is the pre-defined Milliwatt data, and not custom data.
	  (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
	  by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
	  Russell Bryant (license 6283) Note that this patch was written by
	  Russell, even though Matt uploaded it ........ Merged revisions
	  359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2012-03-15 18:17 +0000 [r359609]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, apps/app_dial.c: Add missing connected line
	  macro calls to initial dial for Dial and Queue apps. The
	  connected line interception macros do not get executed when the
	  outgoing channel is initially created and that channel's
	  caller-id is implicitly imported into the incoming channel's
	  connected line data. If you are using the interception macros,
	  you would expect that they get run for every change to a
	  channel's connected line information outside of normal dialplan
	  execution. Review: https://reviewboard.asterisk.org/r/1817/

2012-03-15 00:52 +0000 [r359452-359558]  Russell Bryant <russell@russellbryant.com>

	* channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
	  sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
	  try_transfer() so that the code isn't (potentially) trying to
	  read from it while uninitialized.

	* channels/chan_gtalk.c: chan_gtalk: Fix use of uninitialized vars
	  in config handling. Fix potential use of context, parkinglot, and
	  prefs before they are initialized.

	* channels/chan_gtalk.c: chan_gtalk: Fix potential use of
	  uninitialized variable. Avoid potential use of idroster in
	  gtalk_alloc() before it has been initialized.

	* apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized
	  variable. Ensure that status is set before it is used by
	  resetting it during each loop iteration. This could have resulted
	  in incorrect results from this app.

	* main/udptl.c: udptl: Ensure fec[] in udptl_build_packet() is
	  initialized. Scan results indicated that this array could be used
	  uninitialized. At a quick look, it looks correct. In any case,
	  initializing it is a Good Thing (tm).

	* include/asterisk/app.h: app.h: Always initialize
	  AST_DECLARE_APP_ARGS(). This patch ensures that the struct
	  defined by AST_DECLARE_APP_ARGS() is always fully initialized.
	  I'm not sure if this fixes any real bugs, but it silences a bunch
	  of warnings from coverity, and is generally a good thing to do
	  anyway.

2012-03-14 22:20 +0000 [r359451]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, main/channel.c,
	  channels/chan_agent.c: Fix deadlock potential with some
	  ast_indicate/ast_indicate_data calls. Calling
	  ast_indicate()/ast_indicate_data() with the channel lock held can
	  result in a deadlock with a local channel because of how local
	  channels need to avoid deadlock.

2012-03-14 17:32 +0000 [r359356]  Matthew Jordan <mjordan@digium.com>

	* main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
	  missed resynchronizations When a change in time occurs, such that
	  the timestamps associated with frames being placed into an
	  adaptive jitter buffer (implemented in jitterbuf.c) are
	  significantly different then the previously inserted frames, the
	  jitter buffer checks to see if it needs to be resynched to the
	  new time frame. If three consecutive packets break the threshold,
	  the jitter buffer resynchs itself to the new timestamps. This
	  currently only occurs when history is calculated, and hence only
	  on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
	  hand, are never passed to the history calculations. Because of
	  this, if the jump in time is greater then the maximum allowed
	  length of the jitter buffer, the JB_TYPE_CONTROL frames are
	  dropped and no resynchronization occurs. Alterntively, if the
	  overfill logic is not triggered, the JB_TYPE_CONTROL frame will
	  be placed into the buffer, but with a time reference that is not
	  applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
	  the overflow logic until reads from the jitter buffer reach the
	  errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
	  frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
	  are unlikely to occur in multiples, it perform the
	  resynchronization on any JB_TYPE_CONTROL frame that breaks the
	  resynch threshold. Note that this only impacts chan_iax2, as
	  other consumers of the adaptive jitter buffer use the abstract
	  jitter buffer API, which does not use JB_TYPE_CONTROL frames.
	  Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
	  ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
	  Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
	  (license 5722)

2012-03-14 17:17 +0000 [r359344]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, main/channel.c: Fix Dial m and r options and
	  forked calls generating warnings for voice frames. When connected
	  line support was added, the wait_for_answer() variable single
	  changed its meaning slightly. Unfortunately, the places where
	  single was used did not necessarily get updated to reflect that
	  change. Also audio/video frames were sent to all forked calls
	  when the endpoints were never made compatible. * Don't pass
	  audio/video media frames when the channels have not been made
	  compatible. * Added handling of AST_CONTROL_SRCCHANGE to
	  app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
	  because that frame can also pass a requested MOH class. (closes
	  issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
	  ASTERISK-17541) Reported by: clint Review:
	  https://reviewboard.asterisk.org/r/1805/

2012-03-14 10:52 +0000 [r359050-359259]  Russell Bryant <russell@russellbryant.com>

	* include/asterisk/logger.h, main/logger.c: Fix bogus reads/writes
	  of console log levels in asterisk.c This patch updates the
	  NUMLOGLEVELS define in logger.h to 32, to match the fact that
	  logger.c implements 32 log levels (because of the custom log
	  level stuff). asterisk.c uses this define to size an array of
	  levels per remote console. This array is modified in
	  ast_console_toggle_loglevel(), which is called by the "logger set
	  level" CLI command. While the documentation for the CLI command
	  doesn't make it terribly obvious, you can use this CLI command to
	  toggle a custom log level on a remote console, as well. However,
	  doing so led to an invalid array index in asterisk.c. This array
	  is read from any time a log message is written to a console. So,
	  all custom log level messages resulted in a bogus read if a
	  remote console was connected.

	* apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
	  reads/writes due to incorrect sizeof(). These few places in the
	  code used sizeof() on h_addr in struct hostent. This is
	  sizeof(char *). The correct way to get the size of this address
	  is to use h_length. This error would result in reads/writes of 8
	  bytes instead of 4 on 64-bit machines.

	* main/sched.c: Fix inaccurate sizeof() in sched.c. This code just
	  needed sizeof(int), not sizeof(int *).

	* utils/astman.c: Fix incorrect sizeof() in astman.

	* res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto.
	  In this case, just remove the memset(). There was a redundant
	  memset that is done correctly just 2 lines later.

	* res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.

	* main/features.c: Fix incorrect sizeof() usage in features.c. This
	  didn't actually result in a bug anywhere, luckily. The only place
	  where the result of these memcpys was used is in app_dial, and
	  the only field that it read out of ast_call_feature was the first
	  one, which is an int, so these memcpys always copied just enough
	  to avoid a problem.

	* main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().

	* main/pbx.c: Don't use a buffer after it goes out of scope. 's' is
	  set to 'workspace'. Make sure 'workspace' doesn't go out of scope
	  while the reference to it via 's' is still used.

	* res/ais/ais.h, res/res_ais.c, res/ais/clm.c, res/ais/evt.c: Dump
	  cache of published events when a node joins the cluster. Also use
	  a more reliable method for stopping the poll() thread.

	* makeopts.in, apps/app_rpt.c (removed), channels/chan_usbradio.c
	  (removed), channels/xpmr (removed),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Remove
	  chan_usbradio and app_rpt. These modules are being maintained
	  outside of the tree and have been for a long time now, so it
	  doesn't make sense to keep them here. Review:
	  https://reviewboard.asterisk.org/r/1764/

2012-03-13 20:31 +0000 [r358943-358978]  Terry Wilson <twilson@digium.com>

	* main/features.c: Fix setting CDR variables in the hangup
	  extension A previous CDR fix for setting CDR variables during a
	  bridge via custom dialplan features broke setting CDR variables
	  in the hangup extension. This patch fixes the issue. Review:
	  https://reviewboard.asterisk.org/r/1794/

	* main/devicestate.c, include/asterisk/devicestate.h,
	  channels/chan_sip.c, tests/test_devicestate.c: Make hints for
	  invalid SIP devices return Unavail, not idle This patch
	  drastically simplifies the device state aggegation code. The old
	  method was not only overly complex, but also made it impossible
	  to return AST_DEVICE_INVALID from the aggregation code. The unit
	  test update is as a result of fixing that bug. The SIP change
	  stems from a bug introduced by removing a DNS lookup for
	  hostname-based SIP channels. (closes issue ASTERISK-16702)
	  Review: https://reviewboard.asterisk.org/r/1808/

2012-03-13 16:54 +0000 [r358810-358859]  Tilghman Lesher <tilghman@meg.abyt.es>

	* UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL
	  functionality.

	* utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c,
	  res/ael/pval.c, funcs/func_dialplan.c, tests/test_gosub.c: Enable
	  macros in 1.8 to find the next highest "h" extension in a
	  context, like in 1.4. This change restores functionality that was
	  present in 1.4, when AEL macros were implemented with the Macro
	  dialplan application. Macros are fraught with functionality
	  issues, because they consume a large portion of the underlying
	  application stack. This limits the ability of AEL users to call
	  many layers of subroutines, an issue which Gosub does not have
	  (originally tested to 100,000 levels deep). Therefore, starting
	  in 1.6.0, AEL macros were implemented with Gosub. However, there
	  were some implicit behaviors of Macro, which were not replicated
	  at the same time as with the transition to Gosub, one of which is
	  documented in the related issue. In particular, the "h" extension
	  is designed to execute not in the Macro context, but in the
	  topmost calling context. Due to legacy issues with a misapplied
	  bugfix many years ago, when a macro exited in 1.4, it looks in
	  all calling contexts, bubbling up from the deepest level until it
	  finds an "h" extension. Since AEL hides the complexity of the
	  underlying dialplan logic from the AEL programmer, it's
	  reasonable to assume that this behavior should not change in the
	  transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
	  break working AEL configurations in the transition to Asterisk
	  1.8 LTS. This fix is the result, which implements a search for
	  the "h" extension in all calling Gosub contexts. Fixes
	  ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
	  (License #5003) by Tilghman Lesher (with slight modifications for
	  1.8) Tested by: Johan Wilfer Review:
	  https://reviewboard.asterisk.org/r/1776/

2012-03-08 16:39 +0000 [r358643]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Make transfer not ignore port information
	  with SIP. Attempting to transfer with SIP to an address like
	  1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
	  the host string and ignored. This simply keeps chan_sip from
	  cutting off the port number during these kinds of transfers.
	  (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
	  https://reviewboard.asterisk.org/r/1790/diff/#index_header

2012-03-07 18:25 +0000 [r358530]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c: Change directly setting _softhangup in
	  sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
	  ASTERISK-19372)

2012-03-07 16:11 +0000 [r358484]  Sean Bright <sean@malleable.com>

	* codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
	  number of samples set properly. If the wctc4xxp returns more than
	  a single packet, we need to update the number of samples in the
	  returned frame accordingly. Acked-by: Shaun Ruffell
	  <sruffell@digium.com>

2012-03-07 15:16 +0000 [r358435-358438]  Terry Wilson <twilson@digium.com>

	* configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
	  cdr_adaptive_odbc.conf.sample

	* cdr/cdr_adaptive_odbc.c, cel/cel_odbc.c: Add detection for ODBC
	  WCHAR fields Without detecting these types, cel_odbc blows up
	  when the character set for the table is utf8. This also wraps
	  cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
	  #ifdef seen in other parts of the code.

2012-03-06 17:44 +0000 [r358260-358377]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix ring cadance setup for outgoing calls
	  on FXS ports. * Fix referencing the wrong variable in
	  chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
	  compiling with -Wshadow and finding this bug.

	* channels/sig_ss7.c: Drop SS7 call if not connected yet when
	  INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
	  clear a failed call as soon as possible. * Made SS7 hangup a call
	  immediately if it has not connected yet for
	  INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
	  inband tone. (closes issue ASTERISK-19372) Reported by: Igor
	  Nikolaev

	* channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
	  Setup DSP when SS7 call is connected or early media is available.
	  Outgoing SS7 calls fail to detect incoming DTMF so any bridged
	  channel that requires out-of-band DTMF will not work. * Added
	  sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
	  The new call converts conditionaled out unconverted code and
	  shows that the code really did something useful. * Improved some
	  chan_dahdi DTMF debug messages to help track DTMF handling.
	  (closes issue ASTERISK-19312) Reported by: Igor Nikolaev

2012-03-05 18:49 +0000 [r358214]  Jonathan Rose <jrose@digium.com>

	* main/manager.c: Eliminate double close of file descriptor in
	  manager.c The process_output function in manager.c attempted to
	  call fclose and close immediately afterwards. Since fclose
	  implies close, this resulted in a potential double free on file
	  descriptors. This patch changes that behavior and also adds error
	  checking to fclose and close depending on which was deemed
	  necessary. Also error messages. Thanks to Rosen Iliev for
	  pointing out the location of the problem. (closes issue
	  ASTERISK-18453) Reported By: Jaco Kroon Review:
	  https://reviewboard.asterisk.org/r/1793/

2012-03-05 16:41 +0000 [r358162]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Defer sending the connected line reinvite if
	  a reinvite is already in progress. (issue ASTERISK-19355)
	  Reported by: tomaso (closes issue AST-825)

2012-03-05 15:54 +0000 [r358115]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on
	  Replaces errors Asterisk was not setting pendinginvite in the
	  upper half of handle_request_invite such that the 4xx was
	  retransmitted repeatedly even though an ack was received for
	  every retransmission. (closes issue ASTERISK-19303) Patch-by:
	  Jeremiah Gowdy

2012-03-02 23:27 +0000 [r357986-358029]  Terry Wilson <twilson@digium.com>

	* channels/xpmr/xpmr.c, channels/chan_usbradio.c: Fix
	  unused-but-set-variable warnings All of these were pretty
	  obviously unused. Some were unused because the code that used
	  them was #if 0'd. In those cases, I just commented out the
	  unused-but-set variables.

	* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
	  channels/chan_misdn.c: Correct some set-but-unused variable
	  warnings in the mISDN library. (from kpfleming's commit to trunk
	  r356292)

	* channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode
	  x=++x and x=x=1? Really?

2012-03-02 21:02 +0000 [r357940]  Kinsey Moore <kmoore@digium.com>

	* main/event.c, include/asterisk/strings.h, main/ccss.c,
	  tests/test_event.c: Fix case-sensitivity for device-specific
	  event subscriptions and CCSS This change fixes case-sensitivity
	  for device-specific subscriptions such that the technology
	  identifier is case-insensitive while the remainder of the device
	  string is still case-sensitive. This should also preserve the
	  original case of the device string as passed in to the event
	  system. CCSS is the only feature affected as it is the only
	  consumer of device-specific event subscriptions. The second part
	  of this patch addresses similar case-sensitivity issues within
	  CCSS itself that prevented it from functioning correctly after
	  the fix to the events system. This adds a unit test to verify
	  that the event system works as expected. (closes issue
	  ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/

2012-03-02 18:34 +0000 [r357894]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, main/channel.c: Remove ISDN hold restriction
	  for non-bridged calls. The check if an ISDN call is bridged
	  before it could be placed on hold is not necessary and is overly
	  restrictive. The check was originally done to prevent problems
	  with call transfers in case a user tried to transfer a call
	  connected to an application to another call connected to an
	  application. The ISDN transfer code has not required this
	  restriction for quite some time because ECT could transfer any
	  two active calls to each other. * Remove ISDN hold restriction
	  for calls connected to applications. * Made
	  ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
	  AST_CONTROL_UNHOLD instead of generating a warning message.
	  (closes issue ASTERISK-19388) Reported by: Birger Harzenetter
	  Tested by: rmudgett

2012-03-02 15:58 +0000 [r357811]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: The default value for mohinterpret is the
	  empty string, so when resetting to default values don't
	  explicitly set the value to "default."

2012-03-02 15:45 +0000 [r357809]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix
	  next_channel() channel reference leak in ChanSpy. (closes issue
	  ASTERISK-19461) Reported by: Irontec Patches:
	  app_chanspy_iteartor_next_unref.patch (license #6213) patch
	  uploaded by Irontec (issue ASTERISK-17515)

2012-03-02 00:59 +0000 [r357760-357761]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Fix race condition that can cause important
	  control frames (such as a hangup) to be missed. This takes two
	  actions. 1. Move the reading of the alertpipe in __ast_read() to
	  immediately before the removal of frames from the readq. This
	  means we won't do something silly like read from the alertpipe,
	  then ignore the fact that there's a frame to get from the readq
	  since channel's fdno is the AST_TIMING_FD. 2. When
	  ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
	  if the channel's fdno is the AST_TIMING_FD, then set the fdno to
	  -1. This is because if the rate is 0 and the timingfunc is NULL,
	  it means that the channel's timing fd is being invalidated, so
	  any pending reads should not occur. This may actually solve more
	  issues than the referenced one below, but it's not known at this
	  time for sure. (closes issue ASTERISK-19223) reported by
	  Frank-Michael Wittig Review:
	  https://reviewboard.asterisk.org/r/1779

	* main/translate.c: Second attempt to get optimal translation paths
	  when codec_resample is used. This borrows code heavily from
	  changes made in translation code in Asterisk 10. This uses the
	  quality and sample rate change of translation in order to pick
	  paths rather than the computational cost of translations.
	  Computational cost is used solely in determining if a single
	  translation step from a specific translator is better than the
	  same translation step provided by a different translator. (closes
	  issue ASTERISK-16821) reported by Andrew Lindh Review:
	  https://reviewboard.asterisk.org/r/1772

2012-03-01 14:18 +0000 [r357665]  Kinsey Moore <kmoore@digium.com>

	* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a
	  port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was
	  introduced which changed the behavior of ast_find_ourip such that
	  port number was wiped out. This caused the port in internip
	  (which is used for Contact and Call-ID on NOTIFYs) to be 0. This
	  change causes ast_find_ourip to be port-preserving again. (closes
	  issue ASTERISK-19430)

2012-02-29 19:41 +0000 [r357575]  Walter Doekes <walter+asterisk@wjd.nu>

	* apps/app_dial.c: Fix copying of CDR(accountcode) to local
	  channels. In r203638, during the addition of the Channel Event
	  Logging, in mid-2009, this got broken in trunk and ended up in
	  asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
	  the calling channel is available to dialed channels again as well
	  as showing up properly in the CDR's. (closes issue
	  ASTERISK-19384) Patches: accountcode.patch (License #6033) by
	  jamicque Review: https://reviewboard.asterisk.org/r/1775/
	  Reviewed by: Richard Mudgett

2012-02-28 22:27 +0000 [r357455-357490]  Jonathan Rose <jrose@digium.com>

	* UPGRADE.txt, configs/sip.conf.sample: Adding transport=udp to
	  sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
	  (issue ASTERISK-19352) Reported by: jamicque Patches:
	  asterisk-19352-transport-warning-message-v1.patch uploaded by
	  Michael L. Young (license 5026)

	* cdr/cdr_adaptive_odbc.c: Add additional character type types to
	  supported data types for cdr_adaptive_odbc The reporter was uable
	  to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this
	  patch adds those along with some other character types to the
	  list of types cdr_adaptive_odbc will work using the varchar
	  conditions. The problem wasn't really UTF8 characters as much as
	  it was a failure to respond to the exact type that was
	  declared/in use on that database. (closes issue ASTERISK-19334)
	  Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
	  uploaded by Igor Nikolaev (license 6236)

2012-02-28 21:19 +0000 [r357416]  Tilghman Lesher <tilghman@meg.abyt.es>

	* apps/app_stack.c: Correctly reset the dialplan priority. When the
	  stack frame is allocated, we save the address to which we should
	  return, when the Gosub returns. However, if we just want to
	  restore the priority, then we need to subtract 1 before setting
	  it. Otherwise, when a Gosub goes to a nonexistent address, it
	  will skip a priority in the dialplan. This is because when we
	  return from an application, the PBX increments the priority for
	  us.

2012-02-28 20:57 +0000 [r357407]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Use more reasonable cause code when rejecting
	  incoming call waiting calls. (closes issue ASTERISK-19397)
	  Reported by: Birger Harzenetter Patches: nochannel-cause.patch
	  (license #5870) patch uploaded by Birger Harzenetter

2012-02-28 20:26 +0000 [r357356-357386]  Jonathan Rose <jrose@digium.com>

	* UPGRADE.txt: Moves UPGRADE.txt notes from r357356 to a new
	  section specific to 1.8.12 (issue ASTERISK-19352) reported by:
	  jamicque

	* UPGRADE.txt: Adds UPGRADE.txt notes to r357266 indicating changes
	  to transport option (issue ASTERISK-19352) Reported by: jamicque

2012-02-28 19:32 +0000 [r357352]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_page.c: Remove dupliate 'i' option table entry in
	  app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
	  Patches: app_page-duplicate-i-option.patch (license #5027) patch
	  uploaded by Makoto Dei

2012-02-28 18:00 +0000 [r357266]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Changes transport option in sip.conf so that
	  using multiple instances doesn't stack. Prior to this patch,
	  Using "transport=" multiple times would cause them to add to one
	  another like allow/deny. This patch changes that behavior to
	  simply use the transport option specified last. Also, if no
	  transport option is applied now, the default will automatically
	  be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
	  asterisk-19352-transport-warning-message-v1.patch uploaded by
	  Michael L. Young (license 5026)
	  issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
	  (license 5674) Review:
	  https://reviewboard.asterisk.org/r/1745/diff/#index_header

2012-02-28 14:45 +0000 [r357212]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
	  build system has some special magic to ensure that if Asterisk is
	  built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
	  source is still compiled with the optimizer enabled (even though
	  the result will be thrown away), because the compiler is able to
	  find a great deal of coding errors and bugs as a result of
	  running its optimizers. Unfortunately at some point this mode got
	  broken, and the 'throwaway' compile of the code was no longer
	  done with the optimizer enabled. This patch corrects that
	  problem.

2012-02-27 23:34 +0000 [r357093]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix callerid of Originated calls. Thanks to Matt
	  Riddell for tracking this down. (closes issue ASTERISK-19385)
	  Reported by: ornix

2012-03-29  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.11.0 Released.

2012-03-26  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.11.0-rc3 Released.

	* AST-2012-003

	* AST-2012-002

	* /main/manager.c, /include/asterisk/manager.h: Fix AMI deadlock
	  regression by allowing AMI action callback to be reentrant

	  Fix AMI module reload deadlock from ASTERISK-18479 when it tried
	  to fix the race between calling an AMI action callback and
	  unregistering that action.  Refixes ASTERISK-13784 broken by
	  ASTERISK-17785 change.

	  Locking the ao2 object guaranteed that there were no active
	  callbacks that mattered when ast_manager_unregister() was called.
	  Unfortunately, this causes the deadlock situation.  The patch stops
	  locking the ao2 object to allow multiple threads to invoke the
	  callback re-entrantly.  There is no way to guarantee a module unload
	  will not crash because of an active callback.  The code attempts to
	  minimize the chance with the registered flag and the maximum 5
	  second delay before ast_manager_unregister() returns.

	  The trunk version of the patch changes the API to fix the race
	  condition correctly to prevent the module code from unloading from
	  memory while an action callback is active.

	  * Don't hold the lock while calling the AMI action callback.

	  (closes issue ASTERISK-19487)
	  Reported by: Philippe Lindheimer

	  Review: https://reviewboard.asterisk.org/r/1818/

2012-03-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.11.0-rc2 Released.

	* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
	  a port of 0.

	  In the change from 1.6.2 to 1.8, ast_sockaddr was
	  introduced which changed the behavior of ast_find_ourip such
	  that port number was  wiped out.  This caused the port in
	  internip (which is used for Contact and Call-ID on NOTIFYs) to be
	  0.  This change causes ast_find_ourip to be port-preserving again.

2012-01-30 21:57 +0000 [r353368-353320]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sip/include/sip.h, channels/sip/include/dialog.h,
	  channels/chan_sip.c: RFC3261 Section 8.1.1.5. The sequence number
	  value MUST be expressible as a 32-bit unsigned integer * fix: use
	  %u instead of %d when dealing with CSeq numbers - to remove
	  possibility of -ve numbers. * fix: change all uses of seqno and
	  friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
	  Summary of CSeq numbers. An initial CSeq number must be less than
	  2^31 A CSeq number can increase in value up to 2^32-1 An
	  incrementing CSeq number must not wrap around to 0. Tested with
	  Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
	  Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1699/

	* channels/chan_sip.c: prevent debug messsges displaying -ve Cseq
	  numbers. Missed in R353320

2012-01-30 23:17 +0000 [r353371]  Terry Wilson <twilson@digium.com>

	* include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
	  Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
	  currently takes a pointer to an ast_sockaddr and updates it
	  anytime an address resolves to something different. There are a
	  couple of issues with this. First, the ast_sockaddr is usually
	  the address of an ast_sockaddr inside a refcounted struct and we
	  never bump the refcount of those structs when using dnsmgr. This
	  makes it possible that a refresh could happen after the
	  destructor for that object is called (despite ast_dnsmgr_release
	  being called in that destructor). Second, the module using dnsmgr
	  cannot be aware of an address changing without polling for it in
	  the code. If an action needs to be taken on address update (like
	  re-linking a SIP peer in the peers_by_ip table), then polling for
	  this change negates many of the benefits of having dnsmgr in the
	  first place. This patch adds a function to the dnsmgr API that
	  calls an update callback instead of blindly updating the address
	  itself. It also moves calls to ast_dnsmgr_release outside of the
	  destructor functions and into cleanup functions that are called
	  when we no longer need the objects and increments the refcount of
	  the objects using dnsmgr since those objects are stored on the
	  ast_dnsmgr_entry struct. A helper function for returning the
	  proper default SIP port (non-tls vs tls) is also added and used.
	  This patch also incorporates changes from a patch posted by Timo
	  Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
	  ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/

2012-01-31 16:51 +0000 [r353454]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, main/manager.c: Fix memory leak in
	  error paths for action_originate(). * Fix memory leak of vars in
	  error paths for action_originate(). * Moved struct
	  fast_originate_helper tech and data members to stringfields. *
	  Simplified ActionID header handling for fast_originate(). * Added
	  doxygen note to ast_request() and ast_call() and the associated
	  channel callbacks that the data/addr parameters should be treated
	  as const char *. Review: https://reviewboard.asterisk.org/r/1690/

2012-01-31 23:41 +0000 [r353502]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: Allow res_calendar to be unloaded The
	  calendaring tech modules depend on res_calendar and initially
	  res_calendar just bumped the use count so that it couldn't be
	  unloaded. res_calendar can potentially create many threads and
	  I've seen issues where the Asterisk shutdown has failed where it
	  looked like these threads could be the culprit. This patch adds
	  unload support for res_calendar. Unloading res_calendar will also
	  unload the dependant tech modules as well. (closes issue
	  ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/

2012-02-01 15:02 +0000 [r353550]  Matthew Jordan <mjordan@digium.com>

	* contrib/init.d/etc_default_asterisk: Added clarification for the
	  VERBOSITY setting to etc_default_asterisk Clarified that using
	  the VERBOSITY setting in etc_default_asterisk is the same as
	  using the -v command line switch, which causes Asterisk to launch
	  in console mode. (closes issue ASTERISK-17030) Reported by: Jonas

2012-02-01 15:50 +0000 [r353598]  Sean Bright <sean@malleable.com>

	* include/asterisk/audiohook.h: Resolve an overlap in the
	  ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
	  AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
	  unintended side effects. This patch moves
	  AST_AUDIOHOOK_TRIGGER_WRITE, and updates
	  AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
	  This will affect existing modules that use these flags, so be
	  sure to recompile as necessary. (closes issue ASTERISK-19246)
	  Reported by: feyfre

2012-02-01 21:05 +0000 [r353769-353720]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for
	  various functions in chan_sip There are a number of cleaner
	  looking wrappers for ast_sockaddr_stringify_fmt available which
	  are slightly more readable than using a direct call to
	  ast_sockaddr_stringify_fmt. This patch switches a number of those
	  calls in chan_sip to use those wrappers and is generally
	  harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
	  Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
	  Michael L. Young (license 5026)

	* channels/chan_sip.c: Fix sip show peers port output, align
	  columns, and fix ami port output. A previous patch I committed
	  from ASTERISK-16930 unexpectedly changed some output for the AMI
	  action "sippeers" which this patch changes back. Also, this
	  aligns the output for the cli command "sip show peers" and fixes
	  another issue that patch introduced by using
	  ast_sockaddr_stringify calls multiple times without immediately
	  using the pointer. I also went ahead and did a little janitorial
	  work to clean up whitespace in _sip_show_peers. (issue
	  ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
	  Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
	  Walter Doekes (license 5674)

2012-02-02 16:58 +0000 [r353770]  Mark Michelson <mmichelson@digium.com>

	* UPGRADE.txt, configs/manager.conf.sample,
	  include/asterisk/manager.h, configs/http.conf.sample,
	  main/manager.c, main/http.c: Fix TLS port binding behavior as
	  well as reload behavior: * Removes references to tlsbindport from
	  http.conf.sample and manager.conf.sample * Properly bind to port
	  specified in tlsbindaddr, using the default port if specified. *
	  On a reload, properly close socket if the service has been
	  disabled. A note has been added to UPGRADE.txt to indicate how
	  ports must be set for TLS. (closes issue ASTERISK-16959) reported
	  by Olaf Holthausen (closes issue ASTERISK-19201) reported by
	  Chris Mylonas (closes issue ASTERISK-19204) reported by Chris
	  Mylonas Review: https://reviewboard.asterisk.org/r/1709

2012-02-02 18:31 +0000 [r353818]  Jonathan Rose <jrose@digium.com>

	* funcs/func_curl.c: Backports some documentation for func_curl
	  from 10 to 1.8 For some reason this function was completely
	  undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
	  references to an enumerator that was added in the Asterisk 10
	  version of func_curl. That was the only change I noted. (closes
	  issue ASTERISK-19186) Reported by: Olivier Krief

2012-02-02 20:01 +0000 [r353867]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  Restore the 'w' modifier support for ISDN spans.
	  Dial(DAHDI/g0/1234w888) This feature also causes the sending
	  complete ie to be sent for switch types that do not automatically
	  send the ie. (EuroISDN/ETSI) The main difference between dialing
	  Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
	  sending of the sending complete ie. (closes issue ASTERISK-19176)
	  Reported by: rmudgett Tested by: rmudgett

2012-02-02 22:26 +0000 [r353915]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure entering T.38 passthrough does not
	  cause an infinite loop After R340970 Asterisk was still polling
	  the RTCP file descriptor after RTCP is shut down and removed. If
	  the descriptor happened to have data ready when the removal
	  occured then Asterisk would go into an infinite loop trying to
	  read data that it can never actually access. This change disables
	  the audio RTCP file descriptor for the duration of the T.38
	  transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
	  Vrban

2012-02-03 21:24 +0000 [r353999]  Jonathan Rose <jrose@digium.com>

	* channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due
	  to r335976 Bad locking order was added to chan_agent to prevent
	  segfaults from having no locking in a patch by irroot. This patch
	  addresses the bad locking order by releasing locks before getting
	  the right locking order to stop deadlocks from occuring when
	  doing multiple interactions with agents. (closes issue
	  ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
	  https://reviewboard.asterisk.org/r/1708/

2012-02-06 17:28 +0000 [r354216-354116]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Add missing headers to AMI UnParkedCall event to
	  uniquely identify the call. The AMI UnParkedCall event was
	  missing the Parkinglot and Uniqueid headers that the AMI
	  ParkedCall event contains. (closes issue ASTERISK-19240) Reported
	  by: Michael Yara

	* pbx/pbx_config.c: Improved documentation of CLI "dialplan add
	  extension" command. * Documented dialplan add extension
	  <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
	  of command without the app-data value. There are many
	  applications that do no need any parameters so it is silly to
	  require that field for all commands. * Fixed a couple
	  ast_malloc/ast_free mismatches with ast_add_extension2() calls.
	  (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
	  by: rmudgett

2012-02-07 15:04 +0000 [r354263]  Jonathan Rose <jrose@digium.com>

	* cdr/cdr_pgsql.c: Fix column duplication bug in module reload for
	  cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so
	  would cause the column list to keep its current data and then add
	  a second copy during the reload. This would cause attempts to log
	  the CDR to the database to fail. This patch also cleans up some
	  unnecessary null checks for ast_free and deals with a few
	  potential locking problems. (closes issue ASTERISK-19216)
	  Reported by: Jacek Konieczny Review:
	  https://reviewboard.asterisk.org/r/1711/

2012-02-07 20:53 +0000 [r354348]  Terry Wilson <twilson@digium.com>

	* contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
	  Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
	  instead of "" 2. Don't set ipaddr or port to the string "(null)"
	  when they are empty 3. Add missing required fields, set default
	  for lastms to 0, and modify the length of the ipaddr field to 45
	  in the Postgresql realtime.sql file. (closes issue
	  ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/

2012-02-09 02:23 +0000 [r354492]  Russell Bryant <russell@russellbryant.com>

	* main/channel.c: Remove some unnecessary locking from
	  ast_hangup(). This patch removes some unnecessary locking of the
	  channels container in ast_hangup(). The reason this came up is
	  that this lock can very quickly block the entire system. If any
	  of the channel cleanup code decides to block, it causes a problem
	  for the whole system. For example, when audiohooks get destroyed,
	  if that blocks for a while waiting on the mixmonitor thread to
	  exit because it's busy blocking on some I/O, it causes a problem
	  for many other threads in the meantime. Review:
	  https://reviewboard.asterisk.org/r/1712/

2012-02-09 02:52 +0000 [r354495]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well,
	  thats embarrasing. I forgot to initialize the caller_id storage.
	  (closes issue ASTERISK-19311) Reported by: tootai Tested by:
	  rmudgett

2012-02-09 16:30 +0000 [r354542]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
	  codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to
	  account for both lowercase alphatbetic DTMF events, as well as
	  uppercase alphabetic DTMF events. When this occurred, the
	  comparison of the character buffer containing the event code was
	  changed such that the buffer was first compared again '0' and '9'
	  to determine if it was numeric. Unfortunately, since the first
	  character in the buffer will typically be '1' in the case of
	  non-numeric event codes (10-16), this caused those codes to be
	  converted to a DTMF event of '1'. This patch fixes that, and
	  cleans up handling of both application/dtmf-relay and
	  application/dtmf content types. Review:
	  https://reviewboard.asterisk.org/r/1722/ (closes issue
	  ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan

2012-02-09 16:56 +0000 [r354545]  Mark Michelson <mmichelson@digium.com>

	* CHANGES, res/res_fax.c: Adding reload support to res_fax.so
	  (closes issue ASTERISK-16712) reported by Frank DiGennaro Review:
	  https://reviewboard.asterisk.org/r/1713

2012-02-09 17:07 +0000 [r354547]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Clean-up of minor formatting issues in
	  r354542/3/4 rmudgett pointed out some formatting issues in the
	  check-in for ASTERISK-19290. This cleans those up. Review:
	  https://reviewboards.asterisk.org/r/1722/

2012-02-09 17:32 +0000 [r354640-354594]  Mark Michelson <mmichelson@digium.com>

	* main/translate.c: Fix translation path choices. This change makes
	  it so computational cost is not taken into account when deciding
	  if a multistep path is better than a single-step path. This means
	  that the only time a multistep path will be chosen is if no
	  single-step path exists. This ensures a better quality
	  translation even if it turns out to be slightly slower. (closes
	  issue ASTERISK-16821) reported by Andrew Lindh Review:
	  https://reviewboard.asterisk.org/r/1715

	* main/translate.c: Remove outdated comment.

2012-02-09 19:52 +0000 [r354702-354655]  Kinsey Moore <kmoore@digium.com>

	* main/config.c: Make the config parser remove escaping backslashes
	  The config parser in Asterisk does not currently remove a
	  backslash that is used to escape a semicolon which would
	  otherwise be interpreted as the start of a comment. The change
	  here causes that backslash to be removed, but does not create a
	  real escape system in the config parser. The biggest complication
	  with a real escape system would be breaking existing configs
	  everywhere (parsing \\ as \ and breaking on escaped non-semicolon
	  characters) even though it would be the "right" way to do things.
	  (closes issue ASTERISK-17121) Review:
	  https://reviewboard.asterisk.org/r/1724/

	* channels/chan_sip.c: Fix parsing of SIP headers where compact and
	  non-compact headers are mixed Change parsing of SIP headers so
	  that compactness of the header no longer influences which header
	  will be chosen. Previously, a non-compact header would be chosen
	  instead of a preceeding compact-form header. (closes issue
	  ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/

2012-02-09 22:01 +0000 [r354749]  Terry Wilson <twilson@digium.com>

	* funcs/func_cdr.c: Note that CDRs are immutable once a bridge is
	  torn down CDRs cannot be modified after a bridge is torn down,
	  (e.g. after Dial() returns) even though the CDR() function may be
	  called. Since modifying the CDR code to change this behavior
	  could very easily break all kinds of things, this patch just
	  documents this limitation. (closes issues ASTERISK-16923) Review:
	  https://reviewboard.asterisk.org/r/1720/

2012-02-10 18:03 +0000 [r354835]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c: Fix AMI Redirect ExtraChannel not redirecting to
	  the same exten and context. The astman_get_header() never returns
	  NULL so the check by the code for NULL would never fail. (closes
	  issue ASTERISK-16974) Reported by: Nuno Borges Patches:
	  0018325.patch (license #6116) patch uploaded by Nuno Borges
	  (modified)

2012-02-10 21:45 +0000 [r354889]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix a voicemail memory leak with
	  heard/deleted messages. open_mailbox() was changed quite a long
	  time ago to allocate this memory. close_mailbox() should have
	  been changed to be responsible for freeing it.

2012-02-13 17:22 +0000 [r354953]  Richard Mudgett <rmudgett@digium.com>

	* res/res_config_pgsql.c, configs/extconfig.conf.sample: Fix
	  reconnecting to pgsql database after connection loss. There can
	  only be one database connection in res_config_pgsql just like
	  res_config_sqlite. If the connection is lost, the connection may
	  not get reestablished to the same database if the res_pgsql.conf
	  and extconfig.conf files are inconsistent. * Made only use the
	  configured database from res_pgsql.conf. * Fixed potential buffer
	  overwrite of last[] in config_pgsql(). (closes issue
	  ASTERISK-16982) Reported by: german aracil boned Review:
	  https://reviewboard.asterisk.org/r/1731/

2012-02-13 19:49 +0000 [r355009]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at
	  a time as otherwise they would share the same common local
	  context list. (closes issue AST-758)

2012-02-13 22:02 +0000 [r355056]  Richard Mudgett <rmudgett@digium.com>

	* pbx/pbx_spool.c: Fix occasional incorrectly delayed call-file
	  execution. Since the dir timestamp is available at one second
	  resolution, we cannot know if it was updated within the same
	  second after we scanned it. Therefore, we will force another scan
	  if the dir was just modified. * Changed to force another scan if
	  the directory was just modified. (closes issue ASTERISK-19081)
	  Reported by: Knut Bakke Review:
	  https://reviewboard.asterisk.org/r/1688/

2012-02-14 09:41 +0000 [r355136]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: call manager_event only if there is not
	  null channel structure (Closes issue ASTERISK-19298) Reported by:
	  robinfood Patches: issue19298.patch uploaded by may213 (License
	  #5415)

2012-02-14 13:33 +0000 [r355182]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: Clear the high order bit from the
	  destination call number before sending. send_apathetic_reply
	  takes the incoming frame's source call number as the destination
	  call number for the outgoing frame. If the incoming frame was a
	  full frame, then the high order bit of the source call number is
	  set and will be interpreted as a retransmit when sent back out as
	  the destination call number.

2012-02-14 15:50 +0000 [r355228]  Jason Parker <jparker@digium.com>

	* configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs
	  by default in sample configs.

2012-02-14 16:26 +0000 [r355268]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Properly invert the return of a strncmp
	  call. This was causing identification that should have been made
	  private to be public. (closes issue AST-814) reported by Patrick
	  Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
	  (license 5430)

2012-02-14 18:12 +0000 [r355365-355319]  Richard Mudgett <rmudgett@digium.com>

	* cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in
	  cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported
	  by: Alex Villacis Lasso Patches:
	  asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
	  (license #5617) patch uploaded by Alex Villacis Lasso Review:
	  https://reviewboard.asterisk.org/r/1740/

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  formats/format_ogg_vorbis.c: Fix voicemail problems when using
	  ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
	  format because it did not implement the seek and tell format
	  callbacks among other problems. Since we were already using the
	  libvorbis and libvorbisenc libraries we can use libvorbisfile as
	  it is also part of the vorbis library package. * Made use the
	  libvorbisfile to handle the ogg/vorbis file stream. The
	  format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
	  (closes issue ASTERISK-16926) Reported by: sque Patches:
	  ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
	  by sque

2012-02-15 17:24 +0000 [r355529-355448]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: Use TRUNK_CALL_START as originally
	  intended. Back in r646, TRUNK_CALL_START was added and defined as
	  0x4000. That same value was also hard-coded in one part of the
	  IAX2 code instead of using the #define. TRUNK_CALL_START has
	  changed over the years (for dealing with LOW_MEMORY), but the
	  hard-coded usage was never updated to match. This patch fixes
	  that.

	* channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall
	  in chan_iax2 if IAX_OLD_FIND is specified. These variables are
	  only accessed from the IAX_OLD_FIND path, so there is no reason
	  to keep them updated otherwise.

	* channels/chan_iax2.c: When IAX2 debugging is enabled, make sure
	  to log 'apathetic' messages too.

2012-02-16 18:26 +0000 [r355608-355574]  Richard Mudgett <rmudgett@digium.com>

	* res/res_monitor.c: Fix AMI Monitor action without File header
	  converting channel name into filename. * Fix potential Solaris
	  crash if Monitor application has a urlbase and no fname_base
	  option.

	* configure, include/asterisk/autoconfig.h.in,
	  autoconf/ast_c_declare_check.m4 (added), configure.ac,
	  formats/format_ogg_vorbis.c: Fix compile problem when old version
	  of libvorbisfile v1.1.2 is used. The principle difference between
	  libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
	  of the predefined callbacks OV_CALLBACKS_xxx in
	  vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
	  configure script to detect if libvorbisfile.h declares
	  OV_CALLBACKS_NOCLOSE. * Copied the declaration of
	  OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
	  (closes issue ASTERISK-19370) Reported by: Jonn Taylor

2012-02-16 20:01 +0000 [r355622]  Sean Bright <sean@malleable.com>

	* main/audiohook.c: Revert a change to audio_audiohook_write_list
	  that had no affect. When I made this change initially, I was
	  under the false impression that the audiohooks structure remained
	  on the channel after all of the hooks had been detached. This is
	  not the case, ast ast_read takes care of removing the audiohooks
	  structure if the lists are empty.

2012-02-16 23:53 +0000 [r355711-355700]  Paul Belanger <pabelanger@digium.com>

	* addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
	  Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)

	* addons/ooh323c/src/ooSocket.c: Missed a variable

	* addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
	  Revert 355700 and 355701

2012-02-17 16:04 +0000 [r355732-355721]  Mark Michelson <mmichelson@digium.com>

	* main/translate.c: Revert change to translate.c as it has caused
	  an infinite loop to occur in circumstances.

	* channels/chan_sip.c: Fix regressions with regards to route-set
	  creation on early dialogs. This fixes two main issues: 1.
	  Asterisk would send a CANCEL to the route created by the
	  provisional response instead of using the same destination it did
	  in the initial INVITE. 2. If a new route set arrives in a 200 OK
	  than was in the 1XX response (perfectly possible if our outbound
	  INVITE gets forked), then the route set in the 200 OK needs to
	  overwrite the route set in the 1XX response. (closes issue
	  ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
	  Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
	  (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
	  (license 6034) Review: https://reviewboard.asterisk.org/r/1749

2012-02-17 19:32 +0000 [r355793-355746]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: Pass the correct value to
	  ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
	  variable to determine how often to send trunk packets, but this
	  value is in milliseconds while ast_timer_set_rate() expects the
	  rate argument to be ticks per second. So we divide 1000 by
	  trunkfreq and pass that in instead. With a default of 20ms, this
	  change makes IAX2 send trunk packets every 20ms instead of every
	  50ms. Tracked down by myself and Bob Wienholt.

	* channels/chan_iax2.c, configs/iax.conf.sample: Don't allow
	  trunkfreq to be greater than 1000ms.

2012-02-18 03:59 +0000 [r355839]  Paul Belanger <pabelanger@digium.com>

	* res/res_pktccops.c: Fix -Werror=unused-but-set-variable compiler
	  error (gcc 4.6.2)

2012-02-18 07:55 +0000 [r355850]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
	  channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h,
	  channels/sig_analog.h: push 'outgoing' flag from sig_XXX up to
	  chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
	  in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
	  flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
	  Now provides a callback for all the low level sig_XXX modules.
	  (issue ASTERISK-19316) alecdavis (license 585) Reported by:
	  Jeremy Pepper Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1747/

2012-02-19 17:49 +0000 [r356107-355901]  Sean Bright <sean@malleable.com>

	* channels/chan_iax2.c: Set the length of the ast_sockaddr, so that
	  we can set it's port later. Without this, the call to
	  ast_sockaddr_set_port a few lines later is a noop.

	* channels/chan_iax2.c: Add some boilerplate documentation for
	  IAXVAR and IAXPEER.

	* channels/chan_dahdi.c: Change some debug messages from LOG_DEBUG
	  to ast_debug.

	* channels/chan_dahdi.c: This was a LOG_NOTICE, so roll it back.

	* channels/chan_iax2.c: Remove spurious warning when
	  'qualifyfreqnotok' is set successfully. (closes issue
	  ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
	  Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
	  Covert (license 5512)

	* channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
	  make sense when an IP is passed in.

2012-02-22 14:50 +0000 [r356214]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Fix potential buffer overrun and memory leak
	  when executing "sip show peers" The "sip show peers" command uses
	  a fix sized array to sort the current peers in the peers
	  ao2_container. The size of the array is based on the current
	  number of peers in the container. However, once the size of the
	  array is determined, the number of peers in the container can
	  change, as the peers container is not locked. This could cause a
	  buffer overrun when populating the array, if peers were added to
	  the container after the array was created. Additionally, a memory
	  leak of the allocated array would occur if a user caused the
	  _show_peers method to return CLI_SHOWUSAGE. We now create a
	  snapshot of the current peers using an ao2_callback with the
	  OBJ_MULTIPLE flag. This size of the array is set to the number of
	  peers that the iterator will iterate over; hence, if peers are
	  added or removed from the peers container it will not affect the
	  execution of the "sip show peers" command. Review:
	  https://reviewboard.asterisk.org/r/1738/ (closes issue
	  ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
	  Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
	  Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
	  (license 6283)

2012-02-22 20:20 +0000 [r356290]  Paul Belanger <pabelanger@digium.com>

	* apps/app_rpt.c: Fix -Werror=unused-but-set-variable compiler
	  error (gcc 4.6.2) Review:
	  https://reviewboard.asterisk.org/r/1763/

2012-02-22 21:08 +0000 [r356291]  Terry Wilson <twilson@digium.com>

	* include/asterisk/calendar.h, main/loader.c, res/res_calendar.c:
	  Track module use count for res_calendar If the res_calendar
	  module was followed immediately by one of the calendar tech
	  modules and "core stop gracefully" was run, Asterisk would crash.
	  This patch adds use count tracking for res_calendar so that it is
	  unloaded after the tech modules when shutting down gracefully. It
	  is now not possible to unload all the of the calendar modules via
	  "module unload res_calednar.so", but it is still possible to
	  unload them all via "module unload -h res_calendar.so". Review:
	  https://reviewboard.asterisk.org/r/1752/

2012-02-22 21:29 +0000 [r356430-356335]  Paul Belanger <pabelanger@digium.com>

	* apps/app_rpt.c: Add back strsep() function for previous commit

	* apps/app_rpt.c: Missed one strsep() function

	* addons/chan_ooh323.c: Fix -Werror=unused-but-set-variable
	  compiler error (gcc 4.6.2)

2012-02-23 15:37 +0000 [r356475]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix ACK routing for non-2xx responses. When
	  we send an ACK for a 2xx response to an INVITE, we are supposed
	  to use the learned route set. However, when we receive a non-2xx
	  final response to an INVITE, we are supposed to send the ACK to
	  the same place we initially sent the INVITE. We had been doing
	  this up until the changes went in that would build a route set
	  from provisional responses. That introduced a regression where we
	  would use the learned route set under all circumstances. With
	  this change, we now will set the destination of our ACK based on
	  the invitestate. If it is INV_COMPLETED then that means that we
	  have received a non-2xx final response (INV_TERMINATED indicates
	  a 2xx response was received). If it is INV_CANCELLED, then that
	  means the call is being canceled, which means that we should be
	  ACKing a 487 response. The other change introduced here is
	  setting the invitestate to INV_CONFIRMED when we send an ACK
	  *after* the reqprep instead of before. This way, we can tell in
	  reqprep more easily what the invitestate is prior to sending the
	  ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
	  patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
	  (license #5049) (with some slight modifications prior to commit)

2012-02-23 19:49 +0000 [r356521]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c, main/features.c: Fix blind transfer parking
	  issues if the dialed extension is not recognized as a parking
	  extension. Custom parking extensions may not be coded such that
	  the first and only extension priority is the Park application.
	  These custom parking extensions will not be recognized as parking
	  extensions. When a call is blind transferred to an extension that
	  is not recognized as a parking extension, the normal blind
	  transfer code causes the transferred channel to start executing
	  dialplan. Calls that get parked in this manner do not know the
	  original channel name that parked the call so the original parker
	  could never be called back if the parked call is not retrieved
	  before the timeout time. The parking space is also announced to
	  the call being parked as a side effect of not knowing the
	  original parking channel. * Fix handling of BLINDTRANSFER channel
	  variable for call parking. * Fixed SIP blind transfer using the
	  wrong dialplan context variable to check for the parking
	  extension. (closes issue ASTERISK-19322) Reported by: aragon
	  Tested by: rmudgett, jparker Review:
	  https://reviewboard.asterisk.org/r/1730/ JIRA AST-766

2012-02-24 15:07 +0000 [r356650-356604]  Matthew Jordan <mjordan@digium.com>

	* include/asterisk/rtp_engine.h, res/res_srtp.c,
	  channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
	  main/rtp_engine.c: Allow SRTP policies to be reloaded Currently,
	  when using res_srtp, once the SRTP policy has been added to the
	  current session the policy is locked into place. Any attempt to
	  replace an existing policy, which would be needed if the remote
	  endpoint negotiated a new cryptographic key, is instead rejected
	  in res_srtp. This happens in particular in transfer scenarios,
	  where the endpoint that Asterisk is communicating with changes
	  but uses the same RTP session. This patch modifies res_srtp to
	  allow remote and local policies to be reloaded in the underlying
	  SRTP library. From the perspective of users of the SRTP API, the
	  only change is that the adding of remote and local policies are
	  now added in a single method call, whereas they previously were
	  added separately. This was changed to account for the differences
	  in handling remote and local policies in libsrtp. Review:
	  https://reviewboard.asterisk.org/r/1741/ (closes issue
	  ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
	  Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
	  Jordan (license 6283) (with some small modifications for this
	  check-in)

	* res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch for
	  ASTERISK-19253 included properly shutting down the libsrtp
	  library in the case of module unload. Unfortunately, not all
	  distributions have the srtp_shutdown call. As such, this patch
	  removes calling srtp_shutdown.

2012-02-24 18:23 +0000 [r356677]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/tcptls.h, channels/sip/include/sip.h,
	  channels/chan_sip.c: Fix worker thread resource leak in SIP
	  TCP/TLS. The SIP TCP/TLS worker threads were created joinable but
	  noone could join them if they died on their own. * Fix the SIP
	  TCP/TLS worker threads to not be created joinable. *
	  _sip_tcp_helper_thread() only needs one parameter since the pvt
	  parameter is only passed in as NULL and never used. (closes issue
	  ASTERISK-19203) Reported by: Steve Davies Review:
	  https://reviewboard.asterisk.org/r/1714/

2012-02-25 17:21 +0000 [r356797]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Fix crash in app_voicemail during
	  close_mailbox In r354890, a memory leak in app_voicemail was
	  fixed by properly disposing of the allocated heard/deleted
	  pointers. However, there are situations, particularly when no
	  messages are found in a folder, where these pointers are not
	  allocated and not NULL. In that case, an invalid free would be
	  attempted, which could crash app_voicemail. As there are a number
	  of code paths where this could occur, this patch uses the number
	  of messages detected in the folder before it attempts to free the
	  pointers. This resolves the crash detected in the Asterisk Test
	  Suite's check_voicemail_nominal test.

2012-02-27 15:14 +0000 [r356917]  Jonathan Rose <jrose@digium.com>

	* res/res_odbc.c: Remove possible segfaults from res_odbc by adding
	  locks around usage of odbc handle (closes issue ASTERISK-19011)
	  Reported by: Walter Doekes Patches:
	  issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
	  uploaded by Walter Doekes (license 5674) review:
	  https://reviewboard.asterisk.org/r/1719/ review:
	  https://reviewboard.asterisk.org/r/1622/

2012-02-27 16:03 +0000 [r356963]  Terry Wilson <twilson@digium.com>

	* main/features.c: Copy CDR variables when set during a bridge This
	  patch makes sure amaflags, accountcode, and userfield get copied
	  to the bridge CDR when set during a bridge (like via a custom
	  feature). (closes issue ASTERISK-16990) Review:
	  https://reviewboard.asterisk.org/r/1721/

2012-02-27 23:34 +0000 [r357093]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix callerid of Originated calls. Thanks to Matt
	  Riddell for tracking this down. (closes issue ASTERISK-19385)
	  Reported by: ornix

2012-03-06  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.11.0-rc2 Released.

2012-03-05  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.10.0 Released.

2012-03-01  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.10.0-rc4 Released.

	* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
	  a port of 0.

	  In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
	  changed the behavior of ast_find_ourip such that port number was
	  wiped out.  This caused the port in internip (which is used for
	  Contact and Call-ID on NOTIFYs) to be 0.  This change causes
	  ast_find_ourip to be port-preserving again.

2012-02-28  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.10.0-rc3 Released.

	* main/channel.c: Fix callerid of Originated calls.

	  The callerid of originated calls (independent of mechanism) was not
	  being passed to the outbound channel.  This patch fixes that.  Thanks
	  to Matt Riddell for tracking this down.
	  (closes issue ASTERISK-19385)
	  Reported by: ornix

	* channels/chan_sip.c: Fix ACK routing for non-2xx responses.

	  When we send an ACK for a 2xx response to an INVITE, we are supposed
	  to use the learned route set. However, when we receive a non-2xx
	  final response to an INVITE, we are supposed to send the ACK to the
	  same place we initially sent the INVITE.

	  We had been doing this up until the changes went in that would build
	  a route set from provisional responses. That introduced a regression
	  where we would use the learned route set under all circumstances.

	  With this change, we now will set the destination of our ACK based on
	  the invitestate. If it is INV_COMPLETED then that means that we have
	  received a non-2xx final response (INV_TERMINATED indicates a 2xx
	  response was received).  If it is INV_CANCELLED, then that means the
	  call is being canceled, which means that we should be ACKing a 487
	  response.

	  The other change introduced here is setting the invitestate to
	  INV_CONFIRMED when we send an ACK *after* the reqprep instead of
	  before. This way, we can tell in reqprep more easily what the
	  invitestate is prior to sending the ACK.

	  (closes issue ASTERISK-19389)
 	  reported by Karsten Wemheuer
	  patches:
	    ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)

	* channels/chan_sip.c: Fix regressions with regards to route-set
	  creation on early dialogs.

	  This fixes two main issues:
	  1. Asterisk would send a CANCEL to the route created by the provisional
	     response instead of using the same destination it did in the initial
	     INVITE.
	  2. If a new route set arrives in a 200 OK than was in the 1XX response
	     (perfectly possible if our outbound INVITE gets forked), then the
	     route set in the 200 OK needs to overwrite the route set in the 1XX
	     response.
	  (closes issue ASTERISK-19358)
	  Reported by: Karsten Wemheuer
	  Tested by: Karsten Wemheuer
	  patches:
	   ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
	   ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)

 	  Review: https://reviewboard.asterisk.org/r/1749

2012-02-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.10.0-rc2 Released.

	* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
	  codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to
	  account for both lowercase alphatbetic DTMF events, as well as
	  uppercase alphabetic DTMF events.  When this occurred, the comparison
	  of the character buffer containing the event code was changed such
	  that the buffer was first compared against '0' and '9' to determine if
	  it was numeric.  Unfortunately, since the first character in the
	  buffer will typically be '1' in the case of non-numeric event codes
	  (10-16), this caused those codes to be converted to a DTMF event of
	  '1'.  This patch fixes that, and cleans up handling of both
	  application/dtmf-relay and application/dtmf content types.
	  Review: https://reviewboard.asterisk.org/r/1722/
	  (closes issue ASTERISK-19290) Reported by: Ira Emus
	  Tested by: mjordan

	* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
	  uninitialized caller_id storage (closes issue ASTERISK-19311)
	  Reported by: tootai 
	  Tested by: rmudgett

	* channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to
	  r335976. Bad locking order was added to chan_agent to prevent
	  segfaults from having no locking in a patch by irroot. This patch
	  addresses the bad locking order by releasing locks before getting the
	  right locking order to stop deadlocks from occuring when doing
	  multiple interactions with agents. (closes issue ASTERISK-19285)
	  Reported by: Alex Villacis Lasso
	  Review: https://reviewboard.asterisk.org/r/1708/

	* channels/chan_sip.c: Ensure entering T.38 passthrough does not cause
	  an infinite loop. After R340970 Asterisk was still polling the RTCP
	  file descriptor after RTCP is shut down and removed. If the
	  descriptor happened to have data ready when the removal occured then
	  Asterisk would go into an infinite loop trying to read data that it
	  can never actually access. This change disables the audio RTCP file
	  descriptor for the duration of the T.38 transaction. (closes issue
	  ASTERISK-18951) Reported-by: Kristijan Vrban

	* channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: Re-link
	  peers by IP when dnsmgr changes the IP  Asterisk's dnsmgr currently
	  takes a pointer to an ast_sockaddr and updates it anytime an address
	  resolves to something different. There are a couple of issues with
	  this. First, the ast_sockaddr is usually the address of an ast_sockaddr
	  inside a refcounted struct and we never bump the refcount of those
	  structs when using dnsmgr. This makes it possible that a refresh could
	  happen after the destructor for that object is called (despite 
	  ast_dnsmgr_release being called in that destructor). Second, the
	  module using dnsmgr cannot be aware of an address changing without
	  polling for it in the code. If an action needs to be taken on address
	  update (like re-linking a SIP peer in the peers_by_ip table), then
	  polling for this change negates many of the benefits of having dnsmgr
	  in the first place.

2012-02-01 Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.10.0-rc1 Released.

	* Test results:
	  http://bamboo.asterisk.org/browse/TESTING-ASTERISK18100RCS-2

2012-01-30 12:42 +0000 [r353260]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Clarify log WARNING message when port-zero
	  SDP 'm' lines received. Previously, if an m-line in an SDP offer
	  or answer had a port number of zero, that line was skipped, and
	  resulted in an 'Unsupported SDP media type...' warning message.
	  This was misleading, as the media type was not unsupported, but
	  was ignored because the m-line indicated that the media stream
	  had been rejected (in an answer) or was not going to be used (in
	  an offer).

2012-01-29 02:42 +0000 [r353175]  Russell Bryant <russell@russellbryant.com>

	* main/netsock.c: Find even more network interfaces. The previous
	  change made the code look for emN and pciN in addition to what it
	  did originally, which was search for ethN. However, it needed to
	  be looking for pciN#N, so that's what it does now. This also
	  moves the memset() to be before every ioctl().

2012-01-28 14:49 +0000 [r353126]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp_engine.c: Add 'L16-256' MIME subtype alias for slin16.
	  Asterisk has supported the 'L16' MIME subtype for 16kHz signed
	  linear (PCM) audio for quite some time, but some endpoints refer
	  to it as 'L16-256'. This commit adds this as an alias for the
	  existing format.

2012-01-28 04:25 +0000 [r353077]  Russell Bryant <russell@russellbryant.com>

	* main/netsock.c: Update ast_set_default_eid() to find more network
	  interfaces. As of Fedora 15, ethN is not the name of ethernet
	  interfaces. The names are emN or pciN. Update some code that
	  searched for interfaces named ethN to look for the new names, as
	  well. For more information about why this change was made, see
	  this page: http://domsch.com/blog/?p=455

2012-01-27 19:12 +0000 [r352959]  Jonathan Rose <jrose@digium.com>

	* res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
	  with no valid channel not close AMI session. I also went ahead
	  and took a little time to make sure that the manager value
	  AMI_SUCCESS was used instead of just return 0 being thrown around
	  everywhere since that's how we handle this stuff these days.
	  (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
	  res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
	  (license 5766)

2012-01-27 18:22 +0000 [r352955]  Richard Mudgett <rmudgett@digium.com>

	* res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c,
	  channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c,
	  res/res_odbc.c, res/res_srtp.c, main/pbx.c, channels/chan_sip.c,
	  include/asterisk/indications.h: Audit of ao2_iterator_init()
	  usage for v1.8. Fixes numerous reference leaks and missing
	  ao2_iterator_destroy() calls as a result. Review:
	  https://reviewboard.asterisk.org/r/1697/

2012-01-27 00:05 +0000 [r352862]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sip/include/sip.h, channels/chan_sip.c: rfc4235 -
	  Section 4.1: Versions MUST be representable using a non-negative
	  32 bit integer. If a BLF subscription exists for long enough,
	  using %d may print negative version numbers. Unlikely, as 2^32 at
	  1 update per second is ~137 years, or half that before the
	  versions number started going negative. Tested with Asterisk
	  1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested
	  by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/

2012-01-26 20:14 +0000 [r352807]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: Fix outbound DTMF for inband mode (tell
	  asterisk core to generate DTMF sounds). (Closes issue
	  ASTERISK-19233) Reported by: Matt Behrens Patches:
	  chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)

2012-01-26 19:06 +0000 [r352755]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
	  create_addr_from_peer For whatever reason, we don't have a single
	  function for copying data like this from SIP peers to the SIP
	  pvt. This patch adds the copying of amaflags to the sip_pvt, but
	  it would probably be worth discussing this function along with
	  the others that essentially just copy some amount of data from a
	  peer to a private. (Closes issue ASTERISK-19029) Reported by:
	  Matt Lehner

2012-01-26 06:27 +0000 [r352704]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_sip.c: Cleanup dialog-info+xml Notify dialog Make
	  similar to other Notify messages. sample output: <?xml
	  version="1.0"?> <dialog-info
	  xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
	  state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
	  <state>terminated</state> </dialog> </dialog-info> Tested with
	  Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
	  Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1693/

2012-01-25 22:21 +0000 [r352643]  Paul Belanger <pabelanger@digium.com>

	* apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
	  compiler error (gcc 4.6.2)

2012-01-25 21:16 +0000 [r352612]  Kevin P. Fleming <kpfleming@digium.com>

	* main/test.c: Avoid unnecessary rebuilds of main/test.c.
	  main/test.c includes "asterisk/version.h", when it should include
	  "asterisk/ast_version.h" instead (and it should use the
	  ast_get_version() and ast_get_version_num() functions). This
	  commit modifies it to extract the Asterisk version information
	  using the proper APIs, and as a result means that main/test.c no
	  longer needs to be rebuilt when a Subversion checkout is updated
	  or modified.

2012-01-25 17:28 +0000 [r352514-352551]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Remove some extraneous debugging from
	  registry memleak fix

	* channels/chan_sip.c: Clean up some SIP registry-related memory
	  leaks 1) Be sure and free at unload the epa_backend we allocate
	  at startup 2) Do the same sip_registry cleanup at unload we do at
	  reload Review: https://reviewboard.asterisk.org/r/1689/

2012-01-25 16:39 +0000 [r352511]  Jonathan Rose <jrose@digium.com>

	* configs/sip.conf.sample: Redocuments sip types peer, user, friend
	  in sip.conf.sample There was faulty information in the sample
	  config describing user as a synonym for friend so it has been
	  changed to better elaborate on the differences between the three
	  entity types. (closes issue ASTERISK-15537) Reported by: yarique

2012-01-24 22:17 +0000 [r352424]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Don't do a DNS lookup on an outbound
	  REGISTER host if there is an outbound proxy configured. (closes
	  issue ASTERISK-16550) reported by: Olle Johansson

2012-01-24 20:33 +0000 [r352367]  Jonathan Rose <jrose@digium.com>

	* sounds/Makefile: Set core sounds version to 1.4.22. Now that we
	  have the right license for the Russian 1.4.22 sounds as well as
	  the sounds for the Australian English 1.4.22 sounds, we can
	  finally set the sounds to use 1.4.22! (closes issue
	  ASTERISK-18978) Reported by: Cameron Twomey Patches:
	  confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
	  uploaded by Cameron Twomey

2012-01-24 16:59 +0000 [r352291]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_odbc.c: Fix locking issues with channel datastores in
	  func_odbc.c. * Fixed a potential memory leak when an existing
	  datastore is manually destroyed by inline code instead of calling
	  ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
	  Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/

2012-01-24 16:30 +0000 [r352287]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Move RTP timeout check to before bridged
	  channel check so it is actually executed. (issue ASTERISK-19179)
	  Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534)
	  Reported by: kriborgen Patches: chan_sip.patch uploaded by
	  kriborgen (license 6138)

2012-01-23 20:30 +0000 [r352199-352230]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Fix grammar of comment.

	* main/features.c: Fix blind transfers from failing if an 'h'
	  extension is present. This prevents the 'h' extension from being
	  run on the transferee channel when it is transferred via a native
	  transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
	  Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
	  ASTERISK-19173 by Mark Michelson (license 5049) Review:
	  https://reviewboard.asterisk.org/r/1685

2012-01-23 19:12 +0000 [r352144]  Matthew Jordan <mjordan@digium.com>

	* res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27,
	  V29) before starting spandsp layer While the FAXOPT function
	  could be used to set the modem capabilities, the input to that
	  function was not being applied correctly to the spandsp layer.
	  This patch applies the current model capabilities before starting
	  the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
	  Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
	  Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
	  5081) spandsp-modems-10.diff uploaded by mnicholson (license
	  5081)

2012-01-23 17:33 +0000 [r352090]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
	  defined enum values. The invalid value used when notifycid was
	  enabled was benign. As far as the code was concerned -1 and 1 are
	  equivalent. (closes issue ASTERISK-19232) Reported by: Eike
	  Kuiper

2012-01-21 00:20 +0000 [r352029]  Richard Mudgett <rmudgett@digium.com>

	* main/app.c, funcs/func_timeout.c: Fix ast_app_dtget() time unit
	  inconsistency. Note: Noone calls ast_app_dtget() with the timeout
	  parameter of zero so the bad code normally will never get
	  executed. * Fix unnecessary floating point division in
	  func_timeout.c timeout_write() when all other values are
	  integers. (closes issue ASTERISK-16817) Reported by: Dmitry
	  Andrianov

2012-01-21 00:08 +0000 [r352014-352016]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove XXX comment that is not necessary.

	* channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
	  were initiated using a REFER without a prior reINVITE to place
	  the call on hold, AND if Asterisk were sending RTCP reports, then
	  there was a reference for the RTP instance of the transferer.
	  This fixes the issue by merging two similar but slightly
	  conflicting sections of code into a single area. It also adds a
	  stop_media_flows() call in the case that the transferer's UA
	  never sends a BYE to us like it is supposed to. (issue
	  ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/

2012-01-20 19:34 +0000 [r351858-351860]  Kinsey Moore <kmoore@digium.com>

	* codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These
	  changes are in a file that is not compiled by default, and so
	  were missed on earlier checks.

	* codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Allow ilbc
	  code to build under dev mode GCC 4.6.3 found some set/unused
	  variables in the ILBC code.

2012-01-20 16:01 +0000 [r351765]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Accidentally left off a semicolon only in
	  1.8 somehow for previous patch.

2012-01-20 15:48 +0000 [r351760]  Matthew Jordan <mjordan@digium.com>

	* codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from helpfun
	  in ilbc codec gcc version 4.6.2 caught an unused variable in the
	  ilbc codec library. This would prevent compilation with
	  --enable-dev-mode; variable removed.

2012-01-20 15:42 +0000 [r351759]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Adds setting of mwi_from field to
	  check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
	  By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
	  5242)

2012-01-20 12:59 +0000 [r351707]  Stefan Schmidt <sst@sil.at>

	* contrib/asterisk-ng-doxygen: enable doxygen build for files in
	  the channels/sip folder like reqresp_parser.c

2012-01-19 23:17 +0000 [r351618]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
	  fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
	  get_calleridname() parsing and ensure that the output buffer is
	  nul terminated. * Make get_calleridname() truncate the name it
	  parses if the given buffer is too small rather than abandoning
	  the parse and not returning anything for the name. Adjusted
	  get_calleridname_test() unit test to handle the truncation
	  change. * Fix get_in_brackets_test() unit test to check the
	  results of get_in_brackets() correctly. * Fix
	  parse_name_andor_addr() to not return the address of a local
	  buffer. This function is currently not used. * Fix potential NULL
	  pointer dereference in sip_sendtext(). * No need to
	  memset(calleridname) in check_user_full() or tmp_name in
	  get_name_and_number() because get_calleridname() ensures that it
	  is nul terminated. * Reply with an accurate response if
	  get_msg_text() fails in receive_message(). This is academic in
	  v1.8 because get_msg_text() can never fail.

2012-01-19 22:36 +0000 [r351611]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c: Correct output of RTCP jitter statistics
	  in SR and RR reports Change the RTCP RR and SR generation code to
	  convert Asterisk's internal jitter statistics to be represented
	  in RTP timestamp units based on the rate of the codec in use
	  instead of in seconds. (closes issue ASTERISK-14530)

2012-01-19 21:46 +0000 [r351559]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/netsock2.h, channels/chan_sip.c: Eliminates
	  doubling the :port part of SIP Notify Message-Account headers.
	  This patch prevents the domain string from getting mangled during
	  the initreqprep step by moving the initialization to before its
	  immediate use. It also documents this pitfall for the
	  ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
	  by: Yuri Review: https://reviewboard.asterisk.org/r/1678/

2012-01-19 21:11 +0000 [r351504]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Prevent crash when an SDP offer is received
	  with an encrypted video stream when support for video is disabled
	  and res_srtp is loaded. (closes issue ASTERISK-19202) Reported
	  by: Catalin Sanda

2012-01-18 20:54 +0000 [r351450]  Matthew Jordan <mjordan@digium.com>

	* codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c
	  (added), codecs/ilbc/StateConstructW.h (added),
	  codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added),
	  codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added),
	  codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added),
	  codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
	  codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c
	  (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h
	  (added), codecs/ilbc/extract-cfile.awk (added),
	  codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
	  codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c
	  (added), codecs/ilbc/LICENSE_ADDENDUM (added),
	  codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added),
	  codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added),
	  codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added),
	  codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c
	  (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h
	  (added), codecs/ilbc/iLBC_decode.h (added),
	  codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added),
	  codecs/ilbc/hpInput.c (added), codecs/ilbc/gainquant.c (added),
	  codecs/ilbc/iCBSearch.h (added), codecs/ilbc/hpOutput.c (added),
	  codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added),
	  codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
	  codecs/ilbc/hpInput.h (added), codecs/codec_ilbc.c,
	  codecs/ilbc/PATENTS (added), codecs/ilbc/StateSearchW.c (added),
	  codecs/ilbc/hpOutput.h (added),
	  contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LICENSE (added),
	  codecs/ilbc/LPCencode.h (added), codecs/ilbc/StateSearchW.h
	  (added), codecs/ilbc/iCBConstruct.c (added),
	  codecs/ilbc/syntFilter.c (added), codecs/ilbc/iCBConstruct.h
	  (added), codecs/ilbc/iLBC_test.c (added),
	  codecs/ilbc/syntFilter.h (added): Include iLBC source code for
	  distribution with Asterisk This patch includes the iLBC source
	  code for distribution with Asterisk. Clarification regarding the
	  iLBC source code was provided by Google, and the appropriate
	  licenses have been included in the codecs/ilbc folder. Review:
	  https://reviewboard.asterisk.org/r/1675 Review:
	  https://reviewboard.asterisk.org/r/1649 (closes issue:
	  ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan

2012-01-18 14:57 +0000 [r351396]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: The get_pai function in chan_sip.c didn't
	  recognized a proper callerid name and number from a
	  P-Asserted-Identity cause the header parsing logic was wrong.
	  Changing the parsing functions to the sip header parsing APIs in
	  reqresp_parser.h solves this problem. Review:
	  https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
	  Mark Michelson

2012-01-17 17:22 +0000 [r351306]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c: Eliminate odd initialization of probation
	  variable.

2012-01-17 16:55 +0000 [r351287]  Jonathan Rose <jrose@digium.com>

	* CHANGES, res/res_rtp_asterisk.c, configs/rtp.conf.sample: Adds
	  pjmedia probation concepts to res_rtp_asterisk's learning mode.
	  In order to better handle RTP sources with strictrtp enabled
	  (which is now default in 10) using the learning mode to figure
	  out new sources when they change is handled by checking for a
	  number of consecutive (by sequence number) packets received to an
	  rtp struct based on a new configurable value called 'probation'.
	  Also, during learning mode instead of liberally accepting all
	  packets received, we now reject packets until a clear source has
	  been determined. Review: https://reviewboard.asterisk.org/r/1663/

2012-01-17 16:41 +0000 [r351284]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Use built-in parsing functions for Contact
	  and Record-Route headers. If a Contact or a Record-Route header
	  had a quoted string with an item in angle brackets, then we would
	  mis-parse it. For instance, "Bob <1234>" <1234@example.org> would
	  be misparsed as having the URI "1234" The fix for this is to use
	  parsing functions from reqresp_parser.h since they are heavily
	  tested and are awesome. (issue ASTERISK-18990)

2012-01-17 16:06 +0000 [r351233]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Fix udptl issue with initial INVITE
	  introduced by r351027 When an inital INVITE occurs that contains
	  image media, a channel is not yet associated with the SIP dialog.
	  The file descriptor associated with the udptl session needs to be
	  set in initialize_udptl or in sip_new to account for this
	  scenario.

2012-01-17 01:37 +0000 [r351182]  Russell Bryant <russell@russellbryant.com>

	* channels/chan_sip.c: Add some missing locking in chan_sip. This
	  patch adds some missing locking to the function
	  send_provisional_keepalive_full(). This function is called from
	  the scheduler, which is processed in the SIP monitor thread. The
	  associated channel (or pbx) thread will also be using the same
	  sip_pvt and ast_channel so locking must be used. The
	  sip_pvt_lock_full() function is used to ensure proper locking
	  order in a safe manner. In passing, document a suspected
	  reference counting error in this function. The "fix" is left
	  commented out because when the "fix" is present, crashes occur.
	  My theory is that fixing it is exposing a reference counting
	  error elsewhere, but I don't know where. (Or my analysis of this
	  being a problem could have been completely wrong in the first
	  place). Leave the comment in the code for so that someone may
	  investigate it again in the future. Also add a bit of doxygen to
	  transmit_provisional_response(). (closes issue ASTERISK-18979)
	  Review: https://reviewboard.asterisk.org/r/1648

2012-01-16 21:12 +0000 [r351080-351130]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
	  response to INVITE When handling a non-2xx final response on an
	  INVITE transaction, we have to keep the transaction around after
	  we send an ACK in case we receive a retransmission of the
	  response so we can re-transmit the ACK, but also tear down the
	  ast_channel as soon as we transmit the ACK. Before this patch, we
	  could fail at both of these things. Calling
	  sip_alreadygone/needdestroy prevented us from keeping the
	  transaction up and retransmitting the ACK, and queueing
	  CONGESTION was not sufficient to cause the channel to be torn
	  down when originating calls via the CLI, for example. This patch
	  queues a hangup with CONGESTION instead of just queueing
	  CONGESTION for these responses and removes the sip_alreadygone
	  and sip_needdestroy calls from handle_response_invite on non-2xx
	  responses. It relies on the hangup calling sip_scheddestroy. For
	  more information, see section 17.1.1.1 of RFC 3261. (closes issue
	  ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/

	* channels/chan_sip.c: Don't prematurely stop SIP session timer
	  When Asterisk is the UAS (incoming call, endpoint is re-inviting)
	  the SIP session timer expires after half the time the sip
	  endpoint indicates in the Session-expires header in
	  proc_session_timer(). The session timer was being stopped totally
	  and being handled as an error case instead of running again until
	  the second expiry. This patch treats the half-time expiry as a
	  non-error case and continues the timer until the true expiry.
	  (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
	  by: Thomas Arimont Patches: session_timer_fix.diff by Terry
	  Wilson (License #5357) based on session_timer.patch by Thomas
	  Arimont (License #5525)

2012-01-16 19:09 +0000 [r351027]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Create and initialize udptl only when dialog
	  negotiates for image media Prior to this patch, the udptl struct
	  was allocated and initialized when a dialog was associated with a
	  peer that supported T.38, when a new SIP channel was allocated,
	  or what an INVITE request was received. This resulted in any
	  dialog associated with a peer that supported T.38 having udptl
	  support assigned to it, including the UDP ports needed for
	  communication. This occurred even in non-INVITE dialogs that
	  would never send image media. This patch creates and initializes
	  the udptl structure only when the SDP for a dialog specifies that
	  image media is supported, or when Asterisk indicates through the
	  appropriate control frame that a dialog is to support T.38.
	  (closes issue ASTERISK-16698) Reported by: under Tested by:
	  Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
	  (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
	  Broad Tested by: Stefan Schmidt review:
	  https://reviewboard.asterisk.org/r/1668/

2012-01-16 17:04 +0000 [r350975]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c: Add missing code to set direct RTP setup
	  information during dialing.

2012-01-15 20:07 +0000 [r350885-350888]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/asterisk.c: Allow only one thread at a time to do asterisk
	  cleanup/shutdown. Add locking around the really-really-quit part
	  of the core stop/restart part. Previously more than one thread
	  could be called to do cleanup, causing atexit handlers to be run
	  multiple times, in turn causing segfaults. (issue ASTERISK-18883)
	  Reviewed by: Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/1662/ Review:
	  https://reviewboard.asterisk.org/r/1658/

	* utils/extconf.c: Fix -Werror=unused-but-set-variable compile
	  error in utils/extconf.c. Note that I'm not confirming legitimacy
	  of having that file in tree at all. Is anyone using
	  aelparse/conf2ael? (issue ASTERISK-15350)

2012-01-14 16:40 +0000 [r350788-350837]  Kevin P. Fleming <kpfleming@digium.com>

	* autoconf/libcurl.m4, configure, autoconf/ast_gcc_attribute.m4,
	  configure.ac: Ensure that all AC_LANG_PROGRAM calls in the
	  configure script are properly quoted. Recent versions of autoconf
	  (2.68 on my system) won't properly process the configure script
	  unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
	  the script were, but many were not. This patch corrects the
	  unquoted calls.

	* addons/chan_mobile.c, channels/chan_h323.c: Correct some
	  'set-but-not-used' variable warnings.

	* contrib/scripts/install_prereq: Ensure that two prerequisites are
	  properly installed on Debian-style distributions. * Don't specify
	  a specific version of libgmime; newer versions are available now
	  and acceptable. * Install libsrtp so that res_srtp can be built.

2012-01-13 22:05 +0000 [r350736]  Kinsey Moore <kmoore@digium.com>

	* configure, include/asterisk/autoconfig.h.in: Run bootstrap.sh for
	  the for the ASTERISK-18929 fix configure and autoconfig.h.in were
	  not regenerated when the fix was committed.

2012-01-13 21:51 +0000 [r350733]  Richard Mudgett <rmudgett@digium.com>

	* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
	  Correct eventtype names in cel_odbc and cel_pgsql sample files

2012-01-13 21:40 +0000 [r350730]  Kinsey Moore <kmoore@digium.com>

	* bootstrap.sh, main/asterisk.c, configure.ac: Make sure asterisk
	  builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a
	  'struct sockpeercred', not 'struct ucred', which causes
	  compilation of main/asterisk.c to fail in read_credentials().
	  This allows configure to check for sockpeercred and asterisk to
	  deal with it properly. (closes issue ASTERISK-18929) Reported-by:
	  Barry Miller Patch-by: Barry Miller

2012-01-13 20:29 +0000 [r350679]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/config_parser.c: Set port to a default sane value if
	  a bogus one is provided when parsing hostnames.

2012-01-13 17:23 +0000 [r350555-350571]  Richard Mudgett <rmudgett@digium.com>

	* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
	  cel/cel_pgsql.c, cel/cel_odbc.c, cel/cel_manager.c: Use
	  compatible names for event extra data for various CEL backends. *
	  Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change
	  EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190)

	* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
	  main/cel.c, configs/cel_custom.conf.sample, cel/cel_pgsql.c,
	  configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
	  configs/cel.conf.sample, cel/cel_manager.c: Add missing CEL
	  logging fields to various CEL backends. * Add missing eventextra
	  to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
	  EventExtra to cel_manager.c. * Add missing userdeftype support
	  for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
	  (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman

2012-01-13 16:57 +0000 [r350552]  Matthew Jordan <mjordan@digium.com>

	* apps/app_queue.c: Realtime queues failed to load queue
	  information without queue member table Previously, realtime
	  queues could be loaded without defining the queue member table.
	  This allowed for queue members to be dynamic, while the realtime
	  queue definitions could exist in some backing storage. Revision
	  342223 broke this when it changed the return value for
	  realtime_multientry to return NULL when no results are returned.
	  Previously, an empty ast_config object was expected. (closes
	  issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
	  Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
	  Jordan (license 6283)

2012-01-12 15:57 +0000 [r350501]  Jonathan Rose <jrose@digium.com>

	* main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and
	  CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
	  Colledge Patches: features_18.patch uploaded by Nic Colledge
	  (license 6245)

2012-01-11 22:50 +0000 [r350311-350452]  Richard Mudgett <rmudgett@digium.com>

	* main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
	  CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
	  Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
	  #5909) patch uploaded by Corey Farrell

	* CHANGES, apps/app_followme.c, apps/app_dial.c: Make FollowMe
	  optionally update connected line information when the accepting
	  endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
	  with AST_CONTROL_CONNECTED_LINE information so when the parties
	  are initially bridged, the connected line information will be
	  correct. * Added the 'I' option just like the app_dial and
	  app_queue 'I' option. (closes issue ASTERISK-18969) Reported by:
	  rmudgett Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1656/

	* funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
	  function. The time passed by the LOCK function to an internal
	  function was relative time when the function expected absolute
	  time. * Don't use C++ keywords in get_lock(). (closes issue
	  ASTERISK-16868) Reported by: Andrey Solovyev Patches:
	  20101102__issue18207.diff.txt (license #5003) patch uploaded by
	  Andrey Solovyev (modified)

2012-01-09 21:54 +0000 [r350075-350220]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c: Fix joinable thread terminating without
	  joiner memory leak in chan_iax.c. The iax2_process_thread() can
	  exit without anyone waiting to join the thread. If noone is
	  waiting to join the thread then a large memory leak occurs. *
	  Made iax2_process_thread() deatach itself if nobody is waiting to
	  join the thread. (closes issue ASTERISK-17339) Reported by:
	  Tzafrir Cohen Patches:
	  asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
	  (license #5617) patch uploaded by Alex Villacis Lasso (modified)
	  (closes issue ASTERISK-17825) Reported by: wangjin

	* contrib/scripts/live_ast: live_ast: valgrind: run asterisk under
	  valgrind Adds a new sub-command, "valgrind" to live_ast. It runs
	  asterisk under valgrind. The extra command-line parameters are
	  passed to Asterisk as usual, and parameters to valgrind are
	  passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review:
	  https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
	  from http://svn.asterisk.org/svn/asterisk/branches/10

	* contrib/scripts/live_ast, contrib/scripts/valgrind_compare
	  (added): Update contrib script live_ast to invoke Asterisk with
	  valgrind and suppression file. * Added valgrind_compare script to
	  compare two valgrind log files for differences. (issue
	  ASTERISK-17339) Reported by: Tzafrir Cohen Patches:
	  valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
	  live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir
	  Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by
	  Paul Belanger

	* main/asterisk.c: Make Asterisk -x command line parameter imply -r
	  parameter presence. The Asterisk -x command line parameter is
	  documented inconsistently. * Made the -x documentation and
	  behavior consistent. * Since this is also a new year, updated the
	  copyright notices while here. (closes issue ASTERISK-19094)
	  Reported by: Eugene Patches:
	  issueA19094_correct_asterisk_option_x.patch (license #5674) patch
	  uploaded by Walter Doekes (modified) Tested by: Eugene

2012-01-09 15:37 +0000 [r350023]  Kinsey Moore <kmoore@digium.com>

	* apps/app_meetme.c: Prevent SLA settings from getting wiped out on
	  reload If SLA was reloaded without the config file being changed,
	  current settings got wiped out before the SLA reload code decided
	  it wasn't going to reload the file since nothing was changed.
	  Moving the settings reset later in the reload process fixes this.
	  (closes issue AST-744)

2012-01-06 23:17 +0000 [r349968]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't leak CID in From header when
	  presentation=unavailable When someone does
	  Set(CALLERPRES()=unavailable) (or
	  Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
	  header shows "Anonymous" <anonymous@anonymous.invalid>. When
	  sendrpid=yes/pai, the From header will still display the callerid
	  info, even though we supply an rpid header with the anonymous
	  info. It seems like we shouldn't leak that info in any case.
	  Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
	  seems to indicate that one shouldn't send identifying info in the
	  From in this case. This patch anonymizes the From header as well
	  even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
	  https://reviewboard.asterisk.org/r/1649/

2012-01-06 16:46 +0000 [r349819-349872]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_followme.c: Fix memory leaks in app_followme
	  find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
	  Jordan

	* cel/cel_sqlite3_custom.c: Make not assume that the
	  cel_sqlite3_custom SQL table primary key is AcctId. If a table is
	  created by some other application and the primary key is not
	  named "AcctId", cel/cel_sqlite3_custom.c will always try to
	  create the table and fail because it already exists. * Change the
	  SQL table query to not require AcctId as the primary key. (closes
	  issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
	  (license #6337) patch uploaded by socketpair

2012-01-05 22:06 +0000 [r349731]  Kinsey Moore <kmoore@digium.com>

	* main/file.c: Allow playback of formats that don't support seeking
	  ast_streamfile previously did unconditional seeking on files that
	  broke playback of formats that don't support that functionality.
	  This patch avoids the seek that was causing the problem. This
	  regression was introduced in r158062. (closes issue
	  ASTERISK-18994) Patch-by: Timo Teras

2012-01-05 21:46 +0000 [r349672-349728]  Jonathan Rose <jrose@digium.com>

	* main/dsp.c: Fix an issue where dsp.c would interpret multiple
	  dtmf events from a single key press. When receiving calls from a
	  mobile phone into a DISA system on a connection with significant
	  interference, the reporter's Asterisk system would interpret DTMF
	  incorrectly and replicate digits received. This patch resolves
	  that by increasing the number of frames a mismatch has to be
	  detected before assuming the DTMF is over by 1 frame and adjusts
	  dtmf_detect function to reset hits and misses only when an edge
	  is detected. (closes issue ASTERISK-17493) Reported by: Alec
	  Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
	  (license 5546) Review: https://reviewboard.asterisk.org/r/1130/

	* main/asterisk.c: Ensures Asterisk closes when receiving terminal
	  signals in 'no fork' mode. When catching a signal, in no fork
	  mode the console thread is identical to the thread responsible
	  for catching the signal and closing Asterisk, which requires it
	  to first dispense with the console thread. Prior to this patch,
	  if these threads were identical, upon receiving a killing signal,
	  the thread will send an URG signal to itself, which we also catch
	  and then promptly do nothing with. Obviously this isn't useful
	  behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark
	  Patches: quit_on_signals.patch uploaded by Bryon Clark (license
	  6157)

2012-01-04 20:46 +0000 [r349558]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix segfault in chan_dahdi for
	  CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
	  pointer checks in the following chan_dahdi channel callbacks:
	  dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
	  dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
	  Diego Aguirre Tested by: rmudgett

2012-01-04 20:23 +0000 [r349504-349529]  Kinsey Moore <kmoore@digium.com>

	* contrib/init.d/rc.debian.asterisk: Make debian init script
	  conform to the LSB standard Previously, this init script would
	  return 1 if Asterisk was already running. This is incorrect
	  behavior according to the LSB standard and has been fixed by
	  returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
	  johnc

	* contrib/scripts/autosupport, contrib/scripts/autosupport.8:
	  Update autosupport script and man page Added information
	  collection from the output of the utilities: top, free, uptime,
	  ifconfig Added information collection from the output of the
	  Asterisk command 'dahdi show status' Added option / flag '-n,
	  --non-interactive' Updated man page to reflect new option / flag
	  '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
	  issue AST-749)

2012-01-04 19:27 +0000 [r349450-349482]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Adds Subscription-State header to notify
	  with call completion. per RFC3265 (Closes issue ASTERISK-17953)
	  Reported by: George Konopacki Patches: 19400.patch uploaded by
	  mmichelson (license 5049)

	* main/pbx.c: Fix documentation for SayNumber to reflect the fact
	  that language is changed in CHANNEL() (closes issue
	  ASTERISK-18962) reported by: Nir Simionovich

2012-01-27  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.9.0 Released.

	* Test results:
	  http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-6

2012-01-24  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.9.0-rc3 Released.

	* Test results:
	  http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-4

	* main/file.c: Allow playback of formats that don't support
	  seeking.  ast_streamfile previously did unconditional seeking
	  on files that broke playback of formats that don't support that
	  functionality.  This patch avoids the seek that was causing the
	  problem.  (closes issue ASTERISK-18994) Patch-by: Timo Teras

	* channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer
	  is received with an encrypted video stream when support for video
	  is disabled and res_srtp is loaded.  (closes issue ASTERISK-19202)
	  Reported by: Catalin Sanda

	* channels/chan_sip.c: Fix RTP reference leak.  If a blind transfer
	  were initiated using a REFER without a prior reINVITE to place the
	  call on hold, AND if Asterisk were sending RTCP reports, then there
	  was a reference leak for the RTP instance of the transferer.
	  (closes issue ASERISK-19192) Reported by: Tyuta Vitali

	* main/features.c: Fix blind transfers from failing if an 'h' extension
	  is present.  This prevents the 'h' extension from being run on the
	  transferee channel when it is transferred via a native transfer
	  mechanism such as SIP REFER.  (closes issue ASTERISK-19173) Reported
	  by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
	  Mark Michelson (license 5049)

2012-01-13  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.9.0-rc2 Released.

	* Test results:
	  http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-3

	* apps/app_queue.c: Realtime queues failed to load queue
	  information without queue member table.  Revision 342223 
	  broke this when it changed the return value for 
	  realtime_multientry to return NULL when no results are
	  returned.  (closes issue ASTERISK-19170) Reported by: Rene
	  Mendoza Tested by: Rene Mendoza


2011-12-30  Asterisk Development Team <asteriskteam@digium.com>

        * Asterisk 1.8.9.0-rc1 Released.

        * Test results:
          http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-2

2011-12-29 15:13 +0000 [r349339]  Matthew Jordan <mjordan@digium.com>

	* main/rtp_engine.c: Handle AST_CONTROL_UPDATE_RTP_PEER frames in
	  local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER
	  frames in the local bridge loop causes the loop to exit
	  prematurely. This causes a variety of negative side effects,
	  depending on when the loop exits. This patch handles the frame by
	  essentially swallowing the frame in the local loop, as the
	  current channel drivers expect the RTP bridge to handle the
	  frame, and, in the case of the local bridge loop, no additional
	  action is necessary. (issue ASTERISK-19040) (issue
	  ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340)
	  (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/

2011-12-28 21:30 +0000 [r349289]  Sean Bright <sean@malleable.com>

	* main/audiohook.c: Use ast_audiohook_write_list_empty to determine
	  if our lists are empty instead of duplicating that logic.

2011-12-27 20:48 +0000 [r349194]  Matthew Jordan <mjordan@digium.com>

	* res/res_musiconhold.c, res/res_timing_pthread.c,
	  include/asterisk/module.h, res/res_timing_dahdi.c,
	  res/res_timing_timerfd.c: Fix timing source dependency issues
	  with MOH Prior to this patch, res_musiconhold existed at the same
	  module priority level as the timing sources that it depends on.
	  This would cause a problem when music on hold was reloaded, as
	  the timing source could be changed after res_musiconhold was
	  processed. This patch adds a new module priority level,
	  AST_MODPRI_TIMING, that the various timing modules are now loaded
	  at. This now occurs before loading other resource modules, such
	  that the timing source is guaranteed to be set prior to resolving
	  the timing source dependencies. (closes issue ASTERISK-17474)
	  Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
	  Wes Van Tlghem, elguero, Thomas Arimont Patches:
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
	  uploaded by elguero (License #5026)
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
	  uploaded by elguero (License #5026)
	  asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
	  elguero (License #5026) Review:
	  https://reviewboard.asterisk.org/r/1578/

2011-12-27 17:09 +0000 [r349144]  Sean Bright <sean@malleable.com>

	* main/audiohook.c: Once an audiohook is attached to a channel, we
	  continue to transcode all of the frames, even after all of the
	  hooks are detached. This patch short-cicuits us out before we
	  transcode unnecessarily.

2011-12-23 17:25 +0000 [r349044]  Sean Bright <sean@malleable.com>

	* apps/app_chanspy.c: In ChanSpy, don't create audiohooks that will
	  never be used. When ChanSpy is initialized it creates and
	  attaches 3 audiohooks: 1) Read audio off of the channel that we
	  are spying on 2) Write audio to the channel that we are spying on
	  3) Write audio to the channel that is bridged to the channel that
	  we are spying on. The first is always necessary, but the others
	  are used only when specific options are passed to the ChanSpy
	  application (B, d, w, and W to be specific). When those flags are
	  not passed, neither of those audiohooks are ever sent frames, but
	  we still try to process the hooks for each voice frame that we
	  recieve on the channel. So in short - only create and attach
	  audiohooks that we actually need.

2011-12-23 15:24 +0000 [r348992]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c: Fix missing doc tags found while fixing
	  ASTERISK-18689 Add missing <variable></variable> tags in app_dial
	  documentation.

2011-12-23 02:09 +0000 [r348940]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/pbx.h, main/pbx.c, channels/chan_sip.c: Fix
	  extension state callback references in chan_sip. Chan_sip gives a
	  dialog reference to the extension state callback and assumes that
	  when ast_extension_state_del() returns, the callback cannot
	  happen anymore. Chan_sip then reduces the dialog reference count
	  associated with the callback. Recent changes (ASTERISK-17760)
	  have resulted in the potential for the callback to happen after
	  ast_extension_state_del() has returned. For chan_sip, this could
	  be very bad because the dialog pointer could have already been
	  destroyed. * Added ast_extension_state_add_destroy() so chan_sip
	  can account for the sip_pvt reference given to the extension
	  state callback when the extension state callback is deleted. *
	  Fix pbx.c awkward statecbs handling in
	  ast_extension_state_add_destroy() and handle_statechange() now
	  that the struct ast_state_cb has a destructor to call. * Ensure
	  that ast_extension_state_add_destroy() will never return -1 or 0
	  for a successful registration. * Fixed pbx.c statecbs_cmp() to
	  compare the correct information. The passed in value to compare
	  is a change_cb function pointer not an object pointer. * Make
	  pbx.c ast_merge_contexts_and_delete() not perform callbacks with
	  AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
	  deadlocking when those locks are held during the callback. *
	  Removed unused lock declaration for the pbx.c store_hints list.
	  (closes issue ASTERISK-18844) Reported by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1635/

2011-12-22 22:31 +0000 [r348888]  Matthew Jordan <mjordan@digium.com>

	* cel/cel_pgsql.c: Fix for memory leaks / cleanup in cel_pgsql
	  There were a number of issues in cel_pgsql's pgsql_log method: *
	  If either sql or sql2 could not be allocated, the method would
	  return while the pgsql_lock was still locked * If the execution
	  of the log statement succeeded, the sql and sql2 structs were
	  never free'd * Reconnection successes were logged as ERRORs. In
	  general, the severity of several logging statements was reduced
	  (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
	  by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/

2011-12-22 18:38 +0000 [r348833]  Terry Wilson <twilson@digium.com>

	* include/asterisk/frame.h: Allow packetization vaules > 127
	  According to the RTP packetization documentation, and the maximum
	  values listed in AST_FORMAT_LIST, we should support values > that
	  the signed char array that ast_codec_pref makes available to
	  store the value. All places in the code treat the framing field
	  as though it were an int array instaead of a char array anyway,
	  so this just fixes the type of the array. (closes issue
	  ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/

2011-12-20 23:08 +0000 [r348735]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number
	  if it is blank. Some ISDN switches complain or block the call if
	  the RDNIS number is empty. * Made chan_iax2 not save a RDNIS
	  number into the ast_channel if the string is blank. This is what
	  other channel drivers do. (closes issue ASTERISK-17152) Reported
	  by: rmudgett

2011-12-19 21:31 +0000 [r348647]  Richard Mudgett <rmudgett@digium.com>

	* configure, configure.ac: Fix crashes on other platforms caused by
	  interference from Darwin weak symbol support. Support weak
	  symbols on a platform specific basis. The Mac OS X (Darwin)
	  support must be isolated from the other platforms because it has
	  caused other platforms to crash. Several other platforms
	  including Linux have GCC versions that define the weak attribute.
	  However, this attribute is only setup for use in the code by
	  Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
	  Review: https://reviewboard.asterisk.org/r/1617/

2011-12-18 18:27 +0000 [r348516]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/sip.conf.sample, /: Correct two flaws in sip.conf.sample
	  related to AST-2011-013. * The sample file listed *two* values
	  for the 'nat' option as being the default. Only 'force_rport' is
	  the default. * The warning about having differing 'nat' settings
	  confusingly referred to both peers and users. ........ Merged
	  revisions 348515 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2011-12-16 23:51 +0000 [r348310-348464]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, main/features.c: Clean-up on isle five for
	  __ast_request_and_dial() and ast_call_forward(). * Add locking
	  when a channel inherits variables and datastores in
	  __ast_request_and_dial() and ast_call_forward(). Note: The
	  involved channels are not active so there was minimal potential
	  for problems. * Remove calls to ast_set_callerid() in
	  __ast_request_and_dial() and ast_call_forward() because the set
	  information is for the wrong direction. * Don't use C++ keywords
	  for variable names in ast_call_forward(). * Run the redirecting
	  interception macro if defined when forwarding a call in
	  ast_call_forward(). Note: Currently will never execute because
	  the only callers that supply a calling channel supply a hungup or
	  zombie channel. * Make feature_request_and_dial() put the
	  transferee into autoservice when it calls ast_call_forward() in
	  case a redirection interception macro is run. Note: Currently
	  will never happen because the caller channel (Party B) is always
	  hungup at this time. * Make feature_request_and_dial() ignore the
	  AST_CONTROL_PROCEEDING frame to silence a log message.

	* main/channel.c: Fix cut and past error in ast_call_forward().
	  (issue ASTERISK-18836)

	* include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c,
	  res/res_monitor.c, main/channel.c, main/pbx.c,
	  apps/app_authenticate.c, funcs/func_cdr.c, main/features.c: Fix
	  crash during CDR update. The ast_cdr_setcid() and
	  ast_cdr_update() were shown in ASTERISK-18836 to be called by
	  different threads for the same channel. The channel driver thread
	  and the PBX thread running dialplan. * Add lock protection around
	  CDR API calls that access an ast_channel pointer. (closes issue
	  ASTERISK-18836) Reported by: gpluser Review:
	  https://reviewboard.asterisk.org/r/1628/

	* apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
	  CallerID to the announcing channel. ParkAndAnnounce tried to pass
	  the CallerID to the announcing channel but the ID was wiped out
	  by the channel masquerade done when parking the call. * Save the
	  CallerID before parking the channel to pass it to the announcing
	  channel. * Fixed a minor memory leak in ParkAndAnnounce. *
	  Updated some ParkAndAnnounce log messages.

2011-12-14 22:01 +0000 [r348212]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Don't clear LOCALSTATIONID before sending or
	  receiving. The user may set that variable. ASTERISK-18921

2011-12-14 20:34 +0000 [r348154-348157]  Jonathan Rose <jrose@digium.com>

	* configs/features.conf.sample: Fix accidental use of tabs instead
	  of spaces from previous features.conf.sample change

	* configs/features.conf.sample: Document PARKINGSLOT variable in
	  features.conf.sample (issue ASTERISK-16239)

2011-12-13 23:00 +0000 [r348101]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_followme.c, bridges/bridge_builtin_features.c: Fix
	  FollowMe CallerID on outgoing calls. The addition of the
	  Connected Line support changed how CallerID is passed to outgoing
	  calls. The FollowMe application was not updated to pass CallerID
	  to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
	  * Restructured findmeexec() to fix several memory leaks and
	  eliminate some duplicated code. * Made check the return value of
	  create_followme_number(). Putting a NULL into the numbers list is
	  bad if create_followme_number() fails. * Fixed a couple uses of
	  ast_strdupa() inside loops. * The changes to
	  bridge_builtin_features.c fix a similar CallerID issue with the
	  bridging API attended and blind transfers. (Not used at this
	  time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
	  Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1612/

2011-12-13 15:16 +0000 [r348048]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Fix possible misshandling of an incoming SIP
	  response as a peer poke response. Also make sure peer has even
	  qualify enabled when handle a peer poke response. (closes issue
	  ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
	  UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
	  by: David Vossel

2011-12-12 19:22 +0000 [r347995]  Terry Wilson <twilson@digium.com>

	* res/res_srtp.c: Add a separate buffer for SRTCP packets The
	  function ast_srtp_protect used a common buffer for both SRTP and
	  SRTCP packets. Since this function can be called from multiple
	  threads for the same SRTP session (scheduler for SRTCP and
	  channel for SRTP) it was possible for the packets to become
	  corrupted as the buffer was used by both threads simultaneously.
	  This patch adds a separate buffer for SRTCP packets to avoid the
	  problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
	  Collins)

2011-12-09 01:19 +0000 [r347811]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c: Fix some parsing issues in
	  add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
	  potential sign extension issue. * Fix infinite loop in
	  add_exten_to_pattern_tree() handling of character set escape
	  handling. * Added buffer overflow checks in
	  add_exten_to_pattern_tree() character set collection. * Made
	  ignore empty character sets. * Added escape character handling to
	  end-of-range character in character sets. This has a slight
	  change in behavior if the end-of-range character is an escape
	  character. You must now escape it. * Fix potential sign extension
	  issue when expanding character set ranges. * Made remove
	  duplicated characters from character sets. The duplicate
	  characters lower extension matching priority and prevent
	  duplicate extension detection. * Fix escape character handling
	  when the escape character is trying to escape the end-of-string.
	  We could have continued processing characters after the end of
	  the exten string. We could have added the previous character to
	  the pattern matching tree incorrectly. (closes issue
	  ASTERISK-18909) Reported by: Luke-Jr

2011-12-08 21:28 +0000 [r347718]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Fix regression when using tcpenable=no and
	  tlsenable=yes. The tlsenable settings are tucked away in
	  main/tcptls.c, so I missed them when resolving ASTERISK-18837.
	  This should resolve the test suite breakage of the sip tls tests.
	  Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
	  Jordan

2011-12-08 17:50 +0000 [r347595]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Mark channel running the h exten with the
	  soft-hangup flag. When a bridge is broken, ast_bridge_call()
	  might execute the h exten on the calling channel. However, that
	  channel may not have been the channel that broke the bridge by
	  hanging up. The channel executing the h exten must be in a hung
	  up state so things like AGI run in the correct mode. * Make sure
	  ast_bridge_call() marks the channel it is executing the h exten
	  on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
	  to match the pbx.c main dialplan execution loop when it executes
	  the h exten.) (closes issue ASTERISK-18811) Reported by: David
	  Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: David Hajek, rmudgett

2011-12-08 16:19 +0000 [r347531]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Don't crash on INFO automon request with
	  no channel AST-2011-014. When automon was enabled in
	  features.conf, it was possible to crash Asterisk by sending an
	  INFO request if no channel had been created yet. (closes issue
	  ASTERISK-18805) ........ Merged revisions 347530 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2011-12-07 21:36 +0000 [r347438]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c: Update AMI Getvar and Setvar documentation about
	  supplying a channel name. (closes issue ASTERISK-18958) Reported
	  by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621)
	  patch uploaded by rmudgett

2011-12-07 20:23 +0000 [r347369]  Jonathan Rose <jrose@digium.com>

	* apps/app_meetme.c: Fix: Meetme recording variables from realtime
	  DB use null entries over channel variables Meetme would attempt
	  to substitute the realtime values of RECORDING_FILE and
	  RECORDING_FORMAT from the meetme db entry instead of using the
	  channel variable set for those variables in spite of those
	  database entries being NULL or even lacking a column to represent
	  them. (closes issue ASTERISK-18873) Reported by: Byron Clark
	  Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
	  6157)

2011-12-06 23:47 +0000 [r347292]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
	  signals case insensitive. (closes issue ASTERISK-18924) Reported
	  by: Kevin Taylor

2011-12-06 21:44 +0000 [r347239]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c: Documents CHANNEL(musicclass) taking priority over
	  m([x]) in waitExten If waitExten specifies a music class to use
	  with its music on hold option, it will use CHANNEL(musicclass)
	  instead if that channel variable has been set on the initiating
	  channel. This documents that behavior in the waitExten app so
	  that this can be known without checking the documentation of the
	  code in function local_ast_moh_start. (closes issue
	  ASTERISK-18804)

2011-12-06 19:39 +0000 [r347111-347166]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no.
	  When tcpenable=no, sending to transport=tcp hosts was still
	  allowed. Resolving the source address wasn't possible and yielded
	  the string "(null)" in SIP messages. Fixed that and a couple of
	  not-so-correct log messages. (closes issue ASTERISK-18837)
	  Reported by: Andreas Topp Review:
	  https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan

	* apps/app_voicemail.c: Add regression tests for issue
	  ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
	  Reviewed by: Matt Jordan

	* apps/app_voicemail.c: Move setting of voicemail zonetag and
	  locale up a bit. The voicemail [general] zonetag and locale
	  variables weren't loaded until after the mailboxes were
	  initialized. This caused the settings to be unset for those
	  mailboxes until a reload was performed. (closes issue
	  ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
	  Reviewed by: Matt Jordan

2011-12-06 17:05 +0000 [r347058]  Matthew Jordan <mjordan@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: Fixed crash from
	  orphaned MWI subscriptions in chan_sip This patch resolves the
	  issue where MWI subscriptions are orphaned by subsequent SIP
	  SUBSCRIBE messages. When a peer is removed, either by pruning
	  realtime SIP peers or by unloading / loading chan_sip, the MWI
	  subscriptions that were orphaned would still be on the event
	  engine list of valid subscriptions but have a pointer to a peer
	  that no longer was valid. When an MWI event would occur, this
	  would cause a seg fault. (closes issue ASTERISK-18663) Reported
	  by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches:
	  blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
	  Review: https://reviewboard.asterisk.org/r/1610/

2011-12-05 17:39 +0000 [r347006]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Restore call progress code for analog
	  ports. Extracting sig_analog from chan_dahdi lost call progress
	  detection functionality. * Fix analog ports from considering a
	  call answered immediately after dialing has completed if the
	  callprogress option is enabled. (closes issue ASTERISK-18841)
	  Reported by: Richard Miller Patches: chan_dahdi.diff (license
	  #5685) patch uploaded by Richard Miller (Modified by me)
	  sig_analog.c.diff (license #5685) patch uploaded by Richard
	  Miller (Modified by me) sig_analog.h.diff (license #5685) patch
	  uploaded by Richard Miller

2011-12-05 14:56 +0000 [r346954]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c: Resolve duplicate label used in multiple priorities
	  for the same extension. Prior to this patch, if labels with the
	  same name were used for different priorities in the same
	  extension, the new label would be accepted, but it would be
	  unusable since attempts to reach that label would just go to the
	  first one. Now pbx.c detects this, generates a warning in logs,
	  and culls the label before adding it to the dialplan. (closes
	  issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
	  pbx.c.patch uploaded by Kenneth Shumard (License 5077)

2011-12-05 14:45 +0000 [r346951]  Kinsey Moore <kmoore@digium.com>

	* res/res_jabber.exports.in: Fix chan_jingle/gtalk load regression
	  introduced in r346087 Add missing symbol exports for
	  ast_aji_client_destroy and ast_aji_buddy_destroy for usage
	  outside res_jabber. Testing of these changes focused on
	  res_jabber itself, so this problem was missed. Reported-by:
	  Michael Spiceland

2011-12-04 09:57 +0000 [r346899]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
	  domain ACL bypass. The code that allowed admins to create users
	  with domain-only uri's had stopped to work in 1.8 because of the
	  reqresp parser rewrites. This is fixed now: if you have a
	  [mydomain.com] sip user, you can register with useraddr
	  sip:mydomain.com. Note that in that case -- if you're using
	  domain ACLs (a configured domain list) -- mydomain.com must be in
	  the allow list as well. Reviewboard r1606 shows a list of
	  registration combinations and which SIP response codes are
	  returned. Review: https://reviewboard.asterisk.org/r/1533/
	  Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
	  issue ASTERISK-18741)

2011-12-02 16:19 +0000 [r346762]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, channels/chan_h323.c: process null frame
	  pointer returned by ast_rtp_instance_read correctly (closes issue
	  ASTERISK-16697) Reported by: under Patches: segfault.diff
	  (License #5871) patch uploaded by under

2011-12-01 21:11 +0000 [r346700]  Richard Mudgett <rmudgett@digium.com>

	* configs/res_stun_monitor.conf.sample, include/asterisk/stun.h,
	  main/stun.c, res/res_stun_monitor.c: Re-resolve the STUN address
	  if a STUN poll fails for res_stun_monitor. The STUN socket must
	  remain open between polls or the external address seen by the
	  STUN server is likely to change. However, if the STUN request
	  poll fails then the STUN server address needs to be re-resolved
	  and the STUN socket needs to be closed and reopened. * Re-resolve
	  the STUN server address and create a new socket if the STUN
	  request poll fails. * Fix ast_stun_request() return value
	  consistency. * Fix ast_stun_request() to check the received
	  packet for expected message type and transaction ID. * Fix
	  ast_stun_request() to read packets until timeout or an associated
	  response packet is found. The stun_purge_socket() hack is no
	  longer required. * Reduce ast_stun_request() error messages to
	  debug output. * No longer pass in the destination address to
	  ast_stun_request() if the socket is already bound or connected to
	  the destination. (closes issue ASTERISK-18327) Reported by:
	  Wolfram Joost Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1595/

2011-12-01 20:36 +0000 [r346564-346697]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
	  ringing. 183 Ringing isn't even a thing. 183 is actually a
	  session progress message. (closes issue ASTERISK-18925) Reported
	  by: Sebastian Denz Tested by: jrose Patches:
	  asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
	  Denz (License #6139)

	* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
	  r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
	  18 lines Cleaning up chan_sip/tcptls file descriptor closing.
	  This patch attempts to eliminate various possible instances of
	  undefined behavior caused by invoking close/fclose in situations
	  where fclose may have already been issued on a
	  tcptls_session_instance and/or closing file descriptors that
	  don't have a valid index for fd (-1). Thanks for more than a
	  little help from wdoekes. (closes issue ASTERISK-18700) Reported
	  by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
	  Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
	  Review: https://reviewboard.asterisk.org/r/1576/

2011-11-30 19:36 +0000 [r346472]  Leif Madsen <lmadsen@digium.com>

	* configs/queues.conf.sample: Update queues.conf.sample
	  documentation. Update the documentation surrounding the use of
	  MONITOR_EXEC to make it more clear that it can be used for both
	  Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
	  Reported by: David Woolley Patches:
	  issue18817_mixmonitor_queues_doc.diff by Michael L. Young
	  (License #5026)

2011-11-28 14:30 +0000 [r346292]  Stefan Schmidt <sst@sil.at>

	* res/res_rtp_asterisk.c: Fix regression that 'rtp/rtcp set debup
	  ip' only works when also a port was specified. (closes issue
	  ASTERISK-18693) Reported by: Davide Dal Fra Review:
	  https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
	  Doekes

2011-11-23 22:52 +0000 [r346239]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, include/asterisk/acl.h,
	  channels/chan_skinny.c, channels/chan_h323.c, main/acl.c: Fix
	  calls to ast_get_ip() not initializing the address family.

2011-11-23 20:15 +0000 [r346144-346147]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
	  function. In r116240, get_msg_text() got an extra parameter to
	  fix the unwanted addition of trailing newlines to SIP MESSAGE
	  bodies. This caused all linefeeds to be trimmed, which isn't
	  right either. This is a stop-gap; the right fix is to return the
	  original SIP request body. Review:
	  https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan

	* include/asterisk/strings.h: Fix ast_str_truncate signedness
	  warning and documentation. Review:
	  https://reviewboard.asterisk.org/r/1594

2011-11-23 17:12 +0000 [r346086]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_gtalk.c, res/res_jabber.c, channels/chan_jingle.c,
	  include/asterisk/jabber.h: Fix res_jabber resource leaks This
	  should fix almost all resource leaks in res_jabber that involve
	  ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
	  ast_aji_get_client would sometimes bump an object's refcount and
	  sometimes not. Review: https://reviewboard.asterisk.org/r/1553

2011-11-23 16:09 +0000 [r346030]  Terry Wilson <twilson@digium.com>

	* res/res_musiconhold.c: Resume playing existing hold music for
	  cached realtime MOH As a result of the fix for ASTERISK-18039,
	  realtime caching MOH no longer properly resumes playing back a
	  file between different holds in the same call. This is because
	  scanning for new files causes the existing file array to be
	  emptied and we were just comparing that the saved pointer to the
	  filename matched the pointer to the filename in a particular
	  position in the array. An easy fix is to save the filename
	  instead of a pointer to it and then do a strcmp instead of
	  comparing the addresses. (closes issue ASTERISK-18912) Review:
	  https://reviewboard.asterisk.org/r/1596/

2011-11-22 22:55 +0000 [r345976]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/dnsmgr.h, main/dnsmgr.c: Fix dnsmgr entries to
	  ask for the same address family each time. The dnsmgr refresh
	  would always get the first address found regardless of the
	  original address family requested. So if you asked for only IPv4
	  addresses originally, you might get an IPv6 address on refresh. *
	  Saved the original address family requested by
	  ast_dnsmgr_lookup() to be used when the address is refreshed.

2011-11-22 20:29 +0000 [r345923]  Walter Doekes <walter+asterisk@wjd.nu>

	* include/asterisk/logger.h: Clarify why the AST_LOG_* macros exist
	  next to the LOG_* macros. (issue ASTERISK-17973)

2011-11-21 21:03 +0000 [r345828-345829]  Terry Wilson <twilson@digium.com>

	* CHANGES: Change nat=yes to nat=force_rport in CHANGES Fix a small
	  documentation merge issue ASTERISK-18862

	* configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
	  to nat=yes; warn when nat in general and peer differ It is
	  possible to enumerate SIP usernames when the general and
	  user/peer nat settings differ in whether to respond to the port a
	  request is sent from or the port listed for responses in the Via
	  header. In 1.4 and 1.6.2, this would mean if one setting was
	  nat=yes or nat=route and the other was either nat=no or
	  nat=never. In 1.8 and 10, this would mean when one was
	  nat=force_rport and the other was nat=no. In order to address
	  this problem, it was decided to switch the default behavior to
	  nat=yes/force_rport as it is the most commonly used option and to
	  strongly discourage setting nat per-peer/user when at all
	  possible. For more discussion of the issue, please see:
	  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
	  (closes issue ASTERISK-18862) Review:
	  https://reviewboard.asterisk.org/r/1591/ ........ Merged
	  revisions 345776 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
	  revisions 345800 from
	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2

2011-11-19 15:08 +0000 [r345682]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/db.c: Update the documentation to better clarify how the
	  existing commands work. Review:
	  https://reviewboard.asterisk.org/r/1593/

2011-11-17 17:06 +0000 [r345546]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Remove dead code since pri_grab() can never
	  fail. Dead code makes programmers sick. I am sick of looking at
	  it.

2011-11-17 17:04 +0000 [r345545]  Jason Parker <jparker@digium.com>

	* apps/app_confbridge.c: Fix documentation of 's' option. The menu
	  key is #, not *. Reported by p3nguin on #asterisk.

2011-11-16 14:42 +0000 [r345487]  Jonathan Rose <jrose@digium.com>

	* apps/app_voicemail.c: Guarantee messages go into the right
	  folders with multiple recipients Before, using the U flag in
	  Voicemail with multiple recipients would put urgent messages in
	  the INBOX folder for all users past the first thanks to a bug
	  with the message copying function. This would also cause messages
	  to fail to be sent if the INBOX directory hadn't been created for
	  that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
	  Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1589/

2011-11-15 20:09 +0000 [r345219-345431]  Richard Mudgett <rmudgett@digium.com>

	* res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. *
	  Change from using send() to ast_agi_send() so the HANGUP shows up
	  in the AGI debug output. (closes issue ASTERISK-18723) Reported
	  by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch
	  (license #5621) patch uploaded by rmudgett

	* channels/sig_pri.c: Fix typo in sig_pri using wrong structure
	  name. It is fortunate that the typo does not alter generated code
	  since the e->restart.channel and e->ring.channel members are in
	  the same position. (closes issue ASTERISK-18868) Reported by:
	  zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
	  zvision

	* apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused
	  for AMI and realtime. * Add parameter to queue log ADDMEMBER to
	  indicate if the member is paused. (closes issue ASTERISK-18645)
	  Reported by: garlew Patches: paused.diff (License #5337) patch
	  uploaded by garlew Tested by: rmudgett, garlew Review:
	  https://reviewboard.asterisk.org/r/1469/

	* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h,
	  channels/chan_sip.c: Restore SIP DTMF overlap dialing method. The
	  recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
	  working correctly removed a long standing ability to do overlap
	  dialing using DTMF in the early media phase of a call. See
	  ASTERISK-18702 it has a very good description of the issue. I
	  started with Pavel Troller's chan_sip.diff patch on issue
	  ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
	  allowoverlap config option. The new option value causes the
	  Incomplte application to not send anything with chan_sip so the
	  caller can supply more digits via DTMF. * Renames
	  SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
	  since that is what it really means. * Fixed get_destination()
	  inconsistency with the pickup extension matching. * Fixed
	  initialization of PAGE3 of global_flags in reload_config().
	  (closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
	  https://reviewboard.asterisk.org/r/1517/ Review:
	  https://reviewboard.asterisk.org/r/1582/

	* main/pbx.c: Fix Progress spelling error in main/pbx.c. (closes
	  issue ASTERISK-18857) Reported by: David M Patches:
	  mainpbx-trivial.patch (License #6326) patch uploaded by David M

2011-11-14 19:05 +0000 [r345163]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Don't read past end of input when calling write()
	  int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames *
	  sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when
	  new_frames == 1. Otherwise we start reading into adjacent
	  variables declared on the stack. The read end discards what is
	  read, so the values don't matter but it's not a good idea to read
	  past where we want even though new_frames is almost always 1 and
	  should never be large. This patch is basically taken out of
	  kpfleming's eventfd branch, as he mentioned that he remembered
	  fixing it there when I talked to him about this issue. Review:
	  https://reviewboard.asterisk.org/r/1583/

2011-11-14 19:00 +0000 [r345160]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/sip/include/reqresp_parser.h: Update reqresp_parser
	  parse_uri doxygen comments. The issue mentioned in the bug report
	  had been fixed recently by twilson. The reporter included this
	  documentation fix. (closes issue ASTERISK-18572) Reported by:
	  Richard Miller Patch by: Richard Miller (modified)

2011-11-14 15:08 +0000 [r345063]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Ensure that a null vmexten does not cause a
	  segfault When sip_send_mwi_to_peer was modified recently to avoid
	  deadlocks, vmexten was not expected to be null. This change
	  handles that situation to avoid a segfault.

2011-11-14 15:00 +0000 [r345062]  Jonathan Rose <jrose@digium.com>

	* apps/app_voicemail.c: Moves voicemail setup password entry to the
	  end of the setup process. This change was made because
	  forcegreeting and forcename settings in voicemail could be
	  circumvented by hanging up after entering a password, because the
	  only way voicemail currently observes whether a mailbox is new or
	  not is by checking to see if the password is the same as the
	  mailbox number or not. (closes issue ASTERISK-18282) Reported by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/

2011-11-12 16:05 +0000 [r344965]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_misdn.c: mISDN Round Robin break when no channel is
	  available Prevent channels been parsed repetitively.

2011-11-12 00:24 +0000 [r344899]  Terry Wilson <twilson@digium.com>

	* res/res_musiconhold.c: Don't forget to rescan MOH files for
	  cached realtime classes Realtime MOH class caching was
	  implemented because without it, you would build a completely new
	  MOH class and would start the music over at the beginning each
	  time hold was pressed in a conversation. Unfortunately, this
	  broke re-scanning for file changes for realtime MOH classes. This
	  patch corrects that issue. (closes issue ASTERISK-18039) Review:
	  https://reviewboard.asterisk.org/r/1579/

2011-11-11 21:54 +0000 [r344835-344843]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/utils.c, include/asterisk/stringfields.h,
	  include/asterisk/utils.h: Use __alignof__ instead of sizeof for
	  stringfield length storage. Kevin P Fleming suggested that
	  r343157 should use __alignof__ instead of sizeof. For most
	  systems this won't be an issue, but better fix it now while it's
	  still fresh. Review: https://reviewboard.asterisk.org/r/1573

	* channels/sip/reqresp_parser.c: Remove unneeded if(params) checks
	  in reqresp_parser. Nick Lewis added them in
	  https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
	  reason. There is no way that params could become NULL in that
	  piece of code, so I removed these excess checks again.

	* main/manager.c: Fix bad quoting of multiline mxml opaque_data
	  that caused invalid xml. The opaque_data was added and enclosed
	  in single quotes, assuming it would be only a single line. The
	  rest of the lines were appended after the closing quote. (closes
	  issue ASTERISK-18852) Reported by: peep_ on IRC Review:
	  https://reviewboard.asterisk.org/r/1577

2011-11-11 20:42 +0000 [r344823]  Matthew Jordan <mjordan@digium.com>

	* main/file.c: Video format was treated as audio when removed from
	  the file playback scheduler This patch fixes the format type
	  check in ast_closestream and filestream_destructor. Previously a
	  comparison operator was used, but since audio formats are no
	  longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
	  that have a value greater than the video formats), a bitwise AND
	  operation is used instead. Duplicated code was also moved to
	  filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
	  Bedrij Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1580/

2011-11-11 20:10 +0000 [r344769]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Fix regression introduced by SDP fixups If
	  capability is adjusted when switching to UDPTL during fax
	  transmission, fax teardown fails. Make sure capability is only
	  touched if RTP is active. This regression was introduced in
	  R344385.

2011-11-11 18:35 +0000 [r344661-344715]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Check sip.conf maxforwards parameter for
	  range 1 <= x <= 255. JIRA AST-710

	* main/cli.c: Make CLI "core show channel" not hold the channel
	  lock during console output. Holding the channel lock while the
	  CLI "core show channel" command is executing can slow down the
	  system. It could block the system if the console output is halted
	  or paused. * Made capture the CLI "core show channel" output into
	  a buffer to be output after the channel is unlocked. * Removed
	  use of C++ keyword as a variable name. out renamed to obuf. *
	  Checked allocation of obuf for failure so will not crash. (closes
	  issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
	  rmudgett

2011-11-11 15:21 +0000 [r344608]  Jonathan Rose <jrose@digium.com>

	* main/pbx.c: Fix a segmentation fault when using an extension with
	  CID matching and no CID. Attempting to call an extension which
	  used Caller ID matching with a channel that has an empty caller
	  id string would result in a segmentation fault. (closes issue
	  ASTERISK-18392 Reported By: Ales Zelenik

2011-11-10 22:59 +0000 [r344536-344539]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Fix potential deadlock calling ast_call() with
	  channel locks held. Fixed app_queue.c:ring_entry() calling
	  ast_call() with the channel locks held. Chan_local attempts to do
	  deadlock avoidance in its ast_call() callback and could deadlock
	  if a channel lock is already held.

	* apps/app_queue.c: Make AMI event AgentCalled get
	  CallerID/ConnectedLine info from the incoming channel. It was
	  strange that the AgentCalled AMI event would get most of its
	  information from the incoming channel but then get the CallerID
	  information from the outgoing channel. Before connected line
	  support was added, this information was always the same at this
	  point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
	  Tested by: rmudgett

2011-11-10 21:14 +0000 [r344385-344439]  Kinsey Moore <kmoore@digium.com>

	* apps/app_meetme.c: Fix another incorrect case with meetme's PIN
	  logic and add documentation This fixes an issue where a user of a
	  dynamic conference was asked for a PIN twice. This also adds
	  documentation to assist in future modifications to the piece of
	  code responsible for PIN checking. (closes issue AST-670)

	* channels/sip/include/sip.h, channels/chan_sip.c: Fix several bugs
	  with SDP parsing and well-formedness of responses Fix bug
	  ASTERISK-16558 which dealt with the order of responses to
	  incoming streams defined by SDP. Fix unreported bug where
	  offering multiple same-type streams would cause Asterisk to reply
	  with an incorrect SDP response missing one or more streams
	  without a proper declination. Fix bugs related to a single
	  non-audio stream being offered with responses requesting codecs
	  that were not offered in the initial invite along with an
	  additional audio stream that was not in the initial invite.
	  Review: https://reviewboard.asterisk.org/r/1516/

2011-11-10 16:18 +0000 [r344330]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_rtp_asterisk.c: only attempt to do stun handling on ipv4
	  or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified)
	  ASTERISK-18490

2011-11-09 20:37 +0000 [r344268]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix deadlock during dialplan reload. Another
	  deadlock between the conlock/hints and channels/channel locking
	  orders. * Don't hold the channel and private lock in sip_new()
	  when calling ast_exists_extension(). (closes issue
	  ASTERISK-18740) Reported by: Byron Clark Patches:
	  sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
	  Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
	  uploaded by Byron Clark Tested by: Byron Clark

2011-11-09 19:57 +0000 [r344215]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
	  channels/sip/reqresp_parser.c: Don't treat a host:port string as
	  a domain The domain matching code prior to 1.8 used to manually
	  remove the port from the host:port string when determining if an
	  incoming request matched the list of domains. When switching to
	  the new parsing functions, the documentation implied that the
	  "domain" was being returned by these functions, when instead it
	  was returning the "hostport" as defined by RFC 3261. This led to
	  confusion and resulted in 1.8+ rejecting an incoming request from
	  x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch
	  renames the "domain" variables in the parsing functions to
	  "hostport" to more accurately describe what it is that they are
	  returning and also properly truncates the resulting hostport
	  strings when dealing with domain matching. Review:
	  https://reviewboard.asterisk.org/r/1574/

2011-11-09 18:42 +0000 [r344158]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c,
	  addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h: (closes
	  issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
	  ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
	  Tested by: Fabrizio Lazzaretti

2011-11-09 18:38 +0000 [r344157]  Terry Wilson <twilson@digium.com>

	* tests/test_netsock2.c: Add a unit test for
	  ast_sockaddr_split_hostport Review:
	  https://reviewboard.asterisk.org/r/1575/

2011-11-09 17:13 +0000 [r344102]  Kinsey Moore <kmoore@digium.com>

	* apps/app_meetme.c: Fix pin parameter behavior regression in
	  MeetMe The last time this code was touched (by me), a subtlety
	  was missed based on the difference between needing to check a
	  pin's validity and the need to prompt for a pin. (closes issue
	  ASTERISK-18488)

2011-11-09 15:25 +0000 [r344048]  Matthew Nicholson <mnicholson@digium.com>

	* formats/format_wav.c: don't call ltohl() twice on the same value
	  ASTERISK-18739 Patch by: pawel (modified)

2011-11-08 19:25 +0000 [r343936]  Walter Doekes <walter+asterisk@wjd.nu>

	* pbx/pbx_config.c: Fix crash when dialplan remove include is
	  called with too few arguments. "dialplan remove include x from y"
	  crashed when the amount of arguments was less than 6. (closes
	  issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
	  Andrey Solovyev

2011-11-08 17:58 +0000 [r343851]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
	  variable if unknown host configured crash. * Fixed a LOG_ERROR
	  message referencing the config variable list v that had
	  previously been processed and became NULL. * Added error return
	  value set that was missing in an ast_append_ha() error return
	  path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
	  issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
	  (license #5674) patch uploaded by Walter Doekes Tested by:
	  Michele

2011-11-08 13:26 +0000 [r343791]  Leif Madsen <lmadsen@digium.com>

	* build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A
	  hardcoded a branch number was in the prep_tarball which could not
	  work. Changed it to the variable.

2011-11-07 21:40 +0000 [r343690]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: respect case changes in peer names on sip
	  reload ASTERISK-18669

2011-11-07 21:13 +0000 [r343637]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
	  changing dialogs hash key callid. Changing an object value used
	  as a container key requires removing the object from the
	  container and reinserting it. * Created change_callid_pvt() to
	  call instead of build_callid_pvt(). The change_callid_pvt() will
	  correctly change the dialog callid so the ao2 conainter can
	  explicitly unlink it.

2011-11-07 20:27 +0000 [r343621]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Prevent BLF subscriptions from causing
	  deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
	  was causing deadlocks. This function now requires that both the
	  peer and associated pvt be unlocked before it is called for cases
	  where peer and peer->mwipvt form a circular reference. (closes
	  issue ASTERISK-18663) Review:
	  https://reviewboard.asterisk.org/r/1563/

2011-11-07 19:36 +0000 [r343577]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix deadlock if peer is destroyed while
	  sending MWI notice. A dialog cannot be destroyed by the
	  ao2_callback dialog_needdestroy because of a deadlock between the
	  dialogs container lock and the RWLOCK of the events subscription
	  list. * Create dialogs_to_destroy container to hold dialogs that
	  will be destroyed. * Ensure that the event subscription callback
	  will never happen with an invalid peer pointer by making the
	  event callback removal the first thing in the peer destructor
	  callback. (closes issue ASTERISK-18747) Reported by: Gregory
	  Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/

2011-11-03 20:26 +0000 [r343375]  Walter Doekes <walter+asterisk@wjd.nu>

	* res/res_config_sqlite.c: Fix sqlite config driver segfault and
	  broken queries The sqlite realtime handler assumed you had a
	  static config configured as well. The realtime multientry handler
	  assumed that you weren't using dynamic realtime. (closes issue
	  ASTERISK-18354) (closes issue ASTERISK-18355) Review:
	  https://reviewboard.asterisk.org/r/1561

2011-11-03 19:56 +0000 [r343336]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_dialgroup.c: Remove invalid flag given to iterator in
	  func_dialgroup.c

2011-11-03 16:15 +0000 [r343281]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/dlist.c, addons/ooh323c/src/dlist.h: Final fix
	  memleaks in GkClient codes, same for Timer codes. (these memleaks
	  stop development of gk codes, now i can continue) Fix
	  printHandler 'Unbalanced Structure' issues with locking
	  printHandler data for single thread.

2011-11-03 15:33 +0000 [r343220-343276]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/sip.h: Make room for the fax detect flags
	  The original REGISTERTRYING flag, in addition to being impossible
	  to check, also encroached on the space for the flag above it.
	  This patch moves the flags that were below REGISTERTRYING back to
	  where they were as though we had just removed the REGISTERTRYING
	  option.

	* channels/sip/include/sip.h, contrib/realtime/mysql/sippeers.sql,
	  channels/chan_sip.c: Remove registertrying option in chan_sip
	  This option is not only useless, but has been broken since
	  inception since the flag was never copied from the peer where it
	  is set to the pvt where it was checked. RFC 3261 specificially
	  states that you should not send a provisional response to a
	  non-INVITE request, and if we did fix the code so that it worked,
	  it would cause the same kind of user enumeration vulnerability
	  that we've discussed with the nat= setting. This patch removes
	  registertrying option and any code that would have sent a 100
	  response to a register. Review:
	  https://reviewboard.asterisk.org/r/1562/

2011-11-02 22:21 +0000 [r343157-343181]  Walter Doekes <walter+asterisk@wjd.nu>

	* channels/chan_sip.c: Fix improper warning introduced by r342927
	  and more tweaks Changeset r342927 introduced a warning which was
	  only supposed to be emitted when a found realtime peer had an
	  empty (or no) name. It turned out that there were some
	  inconsistencies left. Now found peers with an empty name are
	  explicitly ignored like before r342927 but better. Reviewed by:
	  Stefan Schmidts, Terry Wilson Review:
	  https://reviewboard.asterisk.org/r/1560

	* main/utils.c, include/asterisk/stringfields.h,
	  include/asterisk/utils.h: Ensure that string field lengths are
	  properly aligned Integers should always be aligned. For some
	  platforms (ARM, SPARC) this is more important than for others.
	  This changeset ensures that the string field string lengths are
	  aligned on *all* platforms, not just on the SPARC for which there
	  was a workaround. It also fixes that the length integer can be
	  resized to 32 bits without problems if needed. (closes issue
	  ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
	  Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
	  https://reviewboard.asterisk.org/r/1549

2011-11-02 19:32 +0000 [r343047-343102]  Leif Madsen <lmadsen@digium.com>

	* apps/app_authenticate.c: Add note about how Authenticate()
	  application with option 'd' works. (closes issue ASTERISK-17422)
	  Reported by: Leif Madsen

	* configs/queues.conf.sample: Update documentation for leastrecent
	  strategy. In queues.conf.sample the leastrecent strategy was
	  incorrectly described. Now updated to reflect how the strategy
	  actually checks peers. (closes issue ASTERISK-17854) Reported by:
	  Sebastian Denz Patches: queues.conf-doc_issue.patch (License
	  #6139)

2011-11-02 13:44 +0000 [r342990]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_meetme.c: Modify comments in MeetMe application
	  documentation about DAHDI. The MeetMe application documentation
	  has some comments about usage of DAHDI, and they were a bit
	  outdated relative to modern DAHDI releases. This patch changes
	  the comment to just tell the user that a functional DAHDI timing
	  source is required, and no longer mention 'dahdi_dummy', since
	  that module does not exist in current DAHDI releases.

2011-11-01 20:53 +0000 [r342869-342927]  Walter Doekes <walter+asterisk@wjd.nu>

	* main/config.c, channels/chan_sip.c,
	  configs/extconfig.conf.sample, include/asterisk/config.h: Several
	  fixes to the chan_sip dynamic realtime peer/user lookup There
	  were several problems with the dynamic realtime peer/user lookup
	  code. The lookup logic had become rather hard to read due to lots
	  of incremental changes to the realtime_peer function. And, during
	  the addition of the sipregs functionality, several possibilities
	  for memory leaks had been introduced. The insecure=port matching
	  has always been broken for anyone using the sipregs family. And,
	  related, the broken implementation forced those using sipregs to
	  *still* have an ipaddr column on their sippeers table. Thanks
	  Terry Wilson for comprehensive testing and finding and fixing
	  unexpected behaviour from the multientry realtime call which
	  caused the realtime_peer to have a completely unused code path.
	  This changeset fixes the leaks, the lookup inconsistenties and
	  that you won't need an ipaddr column on your sippeers table
	  anymore (when you're using sipregs). Beware that when you're
	  using sipregs, peers with insecure=port will now start matching!
	  (closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
	  Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
	  Wilson Review: https://reviewboard.asterisk.org/r/1395

	* UPGRADE.txt, configs/res_ldap.conf.sample, res/res_realtime.c,
	  configs/dbsep.conf.sample, main/config.c,
	  contrib/realtime/mysql/sipfriends.sql (removed),
	  contrib/realtime/mysql/sippeers.sql (added),
	  configs/res_config_mysql.conf.sample,
	  configs/extconfig.conf.sample: Cleanup references to sipusers and
	  sipfriends dynamic realtime families Somewhere between 1.4 and
	  1.8 the sipusers family has become completely unused. Before
	  that, the sipfriends family had been obsoleted in favor of
	  separate sipusers and sippeers families. Apparently, they have
	  been merged back again into a single family which is now called
	  "sippeers". Reviewed by: irroot, oej, pabelanger Review:
	  https://reviewboard.asterisk.org/r/1523

2011-10-31 15:58 +0000 [r342769]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_iax2.c, main/pbx.c: Fixed invalid memory access
	  when adding extension to pattern match tree When an extension is
	  removed from a context, its entry in the pattern match tree is
	  not deleted. Instead, the extension is marked as deleted. When an
	  extension is removed and re-added, if that extension is also a
	  prefix of another extension, several log messages would report an
	  error and did not check whether or not the extension was deleted
	  before accessing the memory. Additionally, if the extension was
	  already in the tree but previously deleted, and the pattern was
	  at the end of a match, the findonly flag was not honored and the
	  extension would be erroneously undeleted. Additionaly, it was
	  discovered that an IAX2 peer could be unregistered via the CLI,
	  while at the same time it could be scheduled for unregistration
	  by Asterisk. The unregistration method now checks to see if the
	  peer was already unregistered before continuing with an
	  unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
	  Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
	  Review: https://reviewboard.asterisk.org/r/1526

2011-10-29 04:19 +0000 [r342661]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_linkedlists.c, include/asterisk/linkedlists.h: Fix
	  AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
	  AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
	  iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
	  the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
	  list if AST_LIST_INSERT_BEFORE_CURRENT() or
	  AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
	  cut and paste error using the wrong variable in
	  AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
	  for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
	  AST_LIST_INSERT_LIST_AFTER().

2011-10-27 19:34 +0000 [r342545-342602]  Jonathan Rose <jrose@digium.com>

	* res/res_rtp_multicast.c: Fix sequence number overflow over 16
	  bits causing codec change in RTP packets. Sequence number was
	  handled as an unsigned integer (usually 32 bits I think, more
	  depending on the architecture) and was put into the rtp packet
	  which is basically just a bunch of bits using an or operation.
	  Sequence number only has 16 bits allocated to it in an RTP packet
	  anyway, so it would add to the next field which just happened to
	  be the codec. This makes sure the sequence number is set to be a
	  16 bit integer regardless of architecture (hopefully) and also
	  makes it so the incrementing of the sequence number does bitwise
	  or at the peak of a 16 bit number so that the value will be set
	  back to 0 when going beyond 65535 anyway. (closes issue
	  ASTERISK-18291) Reported by: Will Schick Review:
	  https://reviewboard.asterisk.org/r/1542/

	* res/res_jabber.c: Cleanup reference leaks in res_jabber
	  res_jabber.c had a number of places where astobjs would be
	  referenced and have their reference counts bumped without having
	  a dereference made before the object lost scope. This patch adds
	  a number of ASTOBJ_UNREFs to resolve that. Review:
	  https://reviewboard.asterisk.org/r/1478/

2011-10-25 22:04 +0000 [r342484-342487]  Richard Mudgett <rmudgett@digium.com>

	* main/astobj2.c: Check fopen return value for ao2 reference debug
	  output. Reported by: wdoekes Patched by: wdoekes Review:
	  https://reviewboard.asterisk.org/r/1539/

	* channels/sig_pri.c: Change D-channel warning to be less confusing
	  on non-NFAS setups. The "No D-channels available! Using Primary
	  channel as D-channel anyway!" WARNING message has been confusing
	  on non-NFAS setups. The message refers to things that are NFAS
	  specific. * Changed the warning to several different warnings to
	  be more accurate for the situation and less confusing as a
	  result: "No D-channels up! Switching selected D-channel from X to
	  Y.", "No D-channels up!", and "D-channel is down!".

2011-10-25 21:08 +0000 [r342380-342435]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c: Use int for storing ao2_container_count instad
	  of size_t AST-676

	* apps/app_queue.c: Simplify queue membercount code Despite an
	  ominous sounding comment stating that membercount was for "logged
	  in" members only and thus we couldn't use ao2_container_count(),
	  I could not find a single place in the code where that seemed to
	  be accurate. The only time we decremented membercount was when we
	  were marking something dead or actually removing it. The only
	  places we incremented it were either after ao2_link(), or trying
	  to correct for having set it to 0 during a reload. In every case
	  where we were correcting the value, it seemed that we were trying
	  to make the count actually match what ao2_container_count() would
	  return. The only place I could find where we made a determination
	  about something being "logged in" or not, we didn't trust the
	  membercount, but instead looked at devicestate, paused, etc. This
	  patch removes membercount, replaces its use with
	  ao2_container_count, and manually adds the results of
	  ao2_container_count to a "membercount" field for ast_data queue
	  query results. This patch also would fix AST-676, but as it is
	  slightly riskier than the previously committed fix, the two
	  commits have been made separately. Reivew:
	  https://reviewboard.asterisk.org/r/1541/

	* apps/app_queue.c: Properly update membercount for reloaded
	  members Since q->membercount is set to 0 before reloading, it is
	  important to increment it again for reloaded members as well as
	  added. (closes issue AST-676) Review:
	  https://reviewboard.asterisk.org/r/1541/

2011-10-25 19:08 +0000 [r342276-342328]  Kinsey Moore <kmoore@digium.com>

	* pbx/pbx_spool.c: Fix compilation on Snow Leopard/FreeBSD for
	  pbx_spool.c One of the changes in the recent spool handling of
	  hardlinks patch was just outside a HAVE_INOTIFY block and caused
	  compilation to fail in some build environments. This has been
	  corrected.

	* pbx/pbx_spool.c: Fix spool handling to allow call files to be
	  hardlinked into place This fixes the inotify code to handle call
	  files being hardlinked into the spool directory. The smsq utility
	  does this, instead of rename(), to ensure that it cannot
	  accidentally overwrite an existing spool file. A rename() might
	  do that, but link() will definitely not. The inotify code had
	  broken this, because it would wait for an IN_CLOSE_WRITE event on
	  the file... which was never forthcoming, since it was never
	  opened. Now we look for IN_OPEN events following the IN_CREATE
	  event, and only wait for an IN_CLOSE_WRITE if the file was
	  actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331)
	  Review: https://reviewboard.asterisk.org/r/1391/

2011-10-25 01:23 +0000 [r342223]  Terry Wilson <twilson@digium.com>

	* main/config.c, include/asterisk/config.h: Return NULL when no
	  results returned for realtime_multientry It was not documented
	  what the return value should be when no entries were returned
	  with the multientry realtime callback. This change forces
	  consistent behavior even if the backends return an empty
	  ast_config. Review: https://reviewboard.asterisk.org/r/1521/

2011-10-24 19:49 +0000 [r342061]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Outbound SIP OPTIONS messages will now
	  include fromuser of related peer. This behavior matches up more
	  closely with the way invite/register/etc are handled. This patch
	  also modifies some adjacent code for code style compliance.
	  Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
	  Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
	  by Jeremy Kister (license #6232)

2011-10-23 11:36 +0000 [r341906-341921]  Gregory Nietsky <gregory@distrotech.co.za>

	* apps/app_queue.c: Revert Janitor patch 341906 For now

	* apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
	  patch is related to work on RB1538

2011-10-21 16:41 +0000 [r341806-341809]  Matthew Nicholson <mnicholson@digium.com>

	* pbx/pbx_lua.c: only process args that exist ASTERISK-18395

	* pbx/pbx_lua.c: don't limit the length of app and function
	  arguments ASTERISK-18395

2011-10-20 21:54 +0000 [r341717]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/features.h, main/features.c, res/res_agi.c: Fix
	  AGI exec Park to honor the Park application parameters. The fix
	  for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
	  application because the channel needed to be masqueraded to
	  prevent a crash. Since the Park application now always
	  masquerades the channel into the parking lot, the special check
	  is no longer needed. The fix also resulted in AGI exec Park
	  attempting to double park the call and not honor the Park
	  application parameters. * Removed no longer necessary call to
	  ast_masq_park_call() by AGI exec for the Park application.
	  (Reverts -r146923) * Fix Park application to only return 0 or -1.
	  The AGI exec Park was causing broken pipe error messages because
	  the Park application returned 1 on successful park. (closes issue
	  ASTERISK-18737)

2011-10-20 21:26 +0000 [r341664-341704]  Paul Belanger <pabelanger@digium.com>

	* funcs/func_callerid.c: Fixed typo from previous commit

	* funcs/func_callerid.c: Updated documentation for the optional CID
	  parameter with CALLERID

2011-10-20 15:11 +0000 [r341529]  Terry Wilson <twilson@digium.com>

	* include/asterisk/strings.h: Clean up ast_check_digits The code
	  was originally copied from the is_int() function in the AEL code.
	  wdoekes pointed out that the function should take a const char*
	  and that their was an unneeded variable. This is now fixed.

2011-10-19 18:59 +0000 [r341435]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_gtalk.c: Outgoing calls with Google Voice Google
	  has recently make some changes (again) to their protocol. Rather
	  then patching asterisk to flip between the two different methods,
	  we now allow both. Lets hope this keeps Google Voice happy for a
	  while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
	  Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
	  6311)

2011-10-19 07:38 +0000 [r341379]  Terry Wilson <twilson@digium.com>

	* include/asterisk/strings.h, channels/chan_sip.c: Don't use
	  is_int() since it doesn't link well on all platforms Just create
	  an normal API function in strings.h that does the same thing just
	  to be safe. ASTERISK-17146

2011-10-19 07:15 +0000 [r341366]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
	  when Asterisk has not yet received a Contact URI from a UAS

2011-10-18 23:37 +0000 [r341314]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't resolve numeric hosts or contact
	  unresolved hosts If a SIP dial string contains a numeric hostname
	  that is not a peer name, don't try to resolve it as it is
	  unlikely that someone really means Dial(SIP/0.0.4.26) when
	  Dial(SIP/1050) is called. Also, make sure that create_addr
	  returns -1 if an address isn't resolved so that we don't attempt
	  to send SIP requests to an address that doesn't resolve. (closes
	  issue ASTERISK-17146, ASTERISK-17716) Review:
	  https://reviewboard.asterisk.org/r/1532/

2011-10-18 23:20 +0000 [r341312]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: fix issue on channel numbering (calls could
	  have same channel number on heavy loaded system)

2011-10-18 21:03 +0000 [r341254]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, channels/sip/include/sip.h,
	  channels/chan_mgcp.c, include/asterisk/features.h,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_sip.c, main/features.c: More parking issues. * Fix
	  potential deadlocks in SIP and IAX blind transfer to parking. *
	  Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect
	  the parkext_exclusive option with transfers
	  (Park(,,,,,exclusive_lot) parameter). Created
	  ast_park_call_exten() and ast_masq_park_call_exten() to maintian
	  API compatibility. * Made masq_park_call() handle a failed
	  ast_channel_masquerade() setup. * Reduced excessive struct
	  parkeduser.peername[] size.

2011-10-17 17:35 +0000 [r341189]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Initialize variables before calling
	  parse_uri If parse_uri was called with an empty URI, some
	  pointers would be modified and an invalid read could result. This
	  patch avoids calling parse_uri with an empty contact uri when
	  parsing REGISTER requests. AST-2011-012 (closes issue
	  ASTERISK-18668)

2011-10-17 16:23 +0000 [r341108-341112]  Paul Belanger <pabelanger@digium.com>

	* apps/app_voicemail.c: Fix previous commit

	* apps/app_voicemail.c: Voicemail compiler flags are 'core' support

2011-10-17 15:35 +0000 [r341088]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't try to remove peers without IPs from
	  peers_by_ip (closes issue ASTERISK-18696)

2011-10-17 15:08 +0000 [r341074]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
	  include of asterisk/md5.h in pbx_realtime.c . A commit needed to
	  test the commit message.

2011-10-14 21:36 +0000 [r341022]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/embed_modules.xml, Makefile.moddir_rules: Change the
	  internal name of the menuselect options that are used to control
	  whether modules are embedded or not; using just the bare category
	  name led to accidentally enabling these options when users used
	  the wrong "--enable" operation on the menuselect command line.
	  Now the internal option names are prefixed with "EMBED_", so they
	  won't be the same as the name of the category containing the
	  modules they control the embedding of.

2011-10-14 20:49 +0000 [r340970]  Kinsey Moore <kmoore@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_sip.c: Quiet RTCP Receiver
	  Reports during fax transmission RTCP is now disabled for
	  "inactive" RTP audio streams during SIP T.38 sessions. The
	  ability to disable RTCP streams in res_rtp_asterisk was missing,
	  so this code was added to support the bug fix. (closes issue
	  ASTERISK-18400)

2011-10-14 16:33 +0000 [r340878]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Avoid unnecessary WARNING message Add
	  AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
	  displaying a WARNING message. (closes issue ASTERISK-18610) Patch
	  by: Kristijan_Vrban

2011-10-14 15:58 +0000 [r340863]  Jonathan Rose <jrose@digium.com>

	* codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c,
	  funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml,
	  res/res_fax.c, apps/app_celgenuserevent.c: Fixes some support
	  level info so that it can be read by menuselect. (issue
	  ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/

2011-10-13 22:48 +0000 [r340809]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix DTMF blind transfer continuing to execute
	  dialplan after transfer. Party A calls Party B. Party A DTMF
	  blind transfers Party B to Party C. Party A channel continues to
	  execute dialplan. * Fixed the return value of
	  builtin_blindtransfer() to return the correct value after a
	  transfer so the dialplan will not keep executing. * Removed
	  unnecessary connected line update that did not really do
	  anything. * Made access to GOTO_ON_BLINDXFR thread safe in
	  check_goto_on_transfer(). * Fixed leak of xferchan for failure
	  cases in check_goto_on_transfer(). * Updated debug messages in
	  builtin_blindtransfer() and check_goto_on_transfer(). (closes
	  issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett

2011-10-13 06:58 +0000 [r340717]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: storing the route-set also on a 181 response
	  not only on 180,182 or 183.

2011-10-13 06:52 +0000 [r340662-340715]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Initialize ast_sockaddr before calling
	  ast_sockaddr_resolve Avoid possible jump based on unitialized
	  value

	* res/res_config_sqlite.c: Don't skip the query field on a realtime
	  multi query There is no documented reason to not add the query
	  field to the varlist returned by a realtime multi query, despite
	  the config category being set to its value. Of course, there is
	  no documentation that the category should be set to the value
	  either. There is lots of no documentation when it comes to
	  realtime. But, other engines do not skip this field so I am
	  forcing this backend to follow the convention, because not doing
	  so is very silly.

2011-10-12 20:30 +0000 [r340576]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: Store route-set from provisional SIP
	  responses so early-dialog requests can be routed properly

2011-10-12 20:19 +0000 [r340534]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Update SIP realtime fullcontact regardless
	  of caching We should update the fullcontact field in the realtime
	  table whether or not rtcachefriends is set. There is no reason to
	  treat a non-cached realtime entity differently than a cached in
	  this regard. (closes issue ASTERISK-18446) Reported by: wdoekes

2011-10-12 20:07 +0000 [r340470-340522]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Initialize the PRI channel alarms properly
	  on startup. The PRI channel alarms were initialized with an
	  inverted sense. (closes issue ASTERISK-18710) Reported by:
	  Tzafrir Cohen

	* apps/app_meetme.c: Update MeetMe p and X option documentation
	  when interacting with the s option. ASTERISK-12175 changed the p
	  and X options to not interfere with the s option when they are
	  used together. It makes more sense for the s option to have
	  priority for the DTMF '*' key since it cannot change its
	  activation code. Otherwise, you could not use option s with the p
	  or X options. JIRA AST-671

2011-10-12 16:27 +0000 [r340418]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_sip.c: Fix verbose messages when IPv6 logic was
	  added (closes issue ASTERISK-18612) Reported by: Tim Osman

2011-10-11 21:03 +0000 [r340279-340365]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
	  Add protection for SS7 channel allocation and better glare
	  handling. * Added a CLI "ss7 show channels" command that might
	  prove useful for future debugging. * Made the incoming SS7
	  channel event check and gripe message uniform. * Made sure that
	  the DNID string for an incoming call is always initialized.
	  (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
	  Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
	  patch uploaded by rmudgett

	* channels/sip/include/dialog.h, channels/chan_sip.c: Fix some
	  potential deadlocks pointed out by helgrind. * Fixed deadlock
	  potential calling dialog_unlink_all() in __sip_autodestruct().
	  Found by helgrind. * Fixed deadlock potential in
	  handle_request_invite() after calling sip_new(). Found by
	  helgrind. * The sip_new() function now returns with the created
	  channel already locked. * Removed the dead code that starts a PBX
	  in in sip_new(). No sip_new() callers caused that code to be
	  executed and it was a bad thing to do anyway. * Removed unused
	  parameters and return value from dialog_unlink_all(). * Made
	  dialog_unlink_all() and __sip_autodestruct() safely obtain the
	  owner and private channel locks without a deadlock avoidance
	  loop.

	* include/asterisk/manager.h, main/manager.c: Convert registered
	  AMI actions to ao2 objects. * Fixed race between calling an AMI
	  action callback and unregistering that action. Refixes
	  ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
	  memory leak if an AMI action failed to get registered because is
	  already was registered. Part of the ao2 conversion. * Fixed AMI
	  ListCommands action not walking the actions list with a lock
	  held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
	  stack usage. * Fix AMI Originate action Variable header requiring
	  a space after the header colon. Reported by Yaroslav Panych on
	  the asterisk-dev list. * Increased the number of listed variables
	  allowed per AMI Originate action Variable header to 64. * Fixed
	  AMI GetConfigJSON action output format. * Fixed usage of res
	  contents outside of scope in append_channel_vars(). * Fixed
	  inconsistency of config file channelvars option. The values no
	  longer accumulate with every channelvars option in the config
	  file. Only the last value is kept to be consistent with the CLI
	  "manager show settings" command. (closes issue ASTERISK-18479)
	  Reported by: Jaco Kroon

2011-10-11 00:43 +0000 [r340263]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* include/asterisk/sha1.h, main/channel.c, main/sha1.c: Update SHA1
	  code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the
	  code was originally taken. It has a slightly better code, and a
	  better phrased license (simple 3-clause BSD). * main/sha1.c is
	  sha1.c from RFC 6234 with formatting changes only. *
	  include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
	  6234. * Removed unused include of asterisk/sha1.h from
	  main/channels.c Review: https://reviewboard.asterisk.org/r/1503/

2011-10-10 20:23 +0000 [r340164]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_sip.c: Updated chan_sip to place calls on hold if
	  SDP address in INVITE is ANY This patch fixes the case where an
	  INVITE is received with c=0.0.0.0 or ::. In this case, the call
	  should be placed on hold. Previously, we checked for the address
	  being null; this patch keeps that behavior but also checks for
	  the ANY IP addresses. Review:
	  https://reviewboard.asterisk.org/r/1504/ (closes issue
	  ASTERISK-18086) Reported by: James Bottomley Tested by: Matt
	  Jordan

2011-10-10 14:14 +0000 [r340108]  Matthew Nicholson <mnicholson@digium.com>

	* doc/appdocsxml.dtd, main/loader.c, main/xmldoc.c, main/pbx.c,
	  main/manager.c, res/res_fax.c, apps/app_fax.c,
	  include/asterisk/module.h, res/res_agi.c,
	  include/asterisk/xmldoc.h: Load the proper XML documentation when
	  multiple modules document the same application. This patch adds
	  an optional "module" attribute to the XML documentation spec that
	  allows the documentation processor to match apps with identical
	  names from different modules to their documentation. This patch
	  also fixes a number of bugs with the documentation processor and
	  should make it a little more efficient. Support for multiple
	  languages has also been properly implemented. ASTERISK-18130
	  Review: https://reviewboard.asterisk.org/r/1485/

2011-10-09 01:16 +0000 [r339830-339938]  Igor Goncharovskiy <igor.goncharovsky@gmail.com>

	* channels/chan_unistim.c: Fix compilation issue, caused by missed
	  session structure (closes issue ASTERISK-18694) Reported by:
	  alex70

	* channels/chan_unistim.c: Fix segfault in Unistim channel (closes
	  issue ASTERISK-18638) Reported by: jonnt

	* channels/chan_unistim.c: Fix char array cast as short array in
	  send_client() function (for ARM platform) (closes issue
	  ASTERISK-17314) Reported by: jjoshua

2011-10-07 19:34 +0000 [r339625-339776]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_url.c: Initialize option flags for SendURL application.
	  (closes issue ASTERISK-18574) Reported by: marcelloceschia

	* autoconf/ast_ext_lib.m4, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Fix regression in
	  configure script for libpri capability checks. JIRA AST-598 added
	  the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2
	  persistence issues with some telcos. ASTERISK-18535 attempted to
	  fix the unexpected requirement that libpri *must* have that
	  feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT
	  lines made the PRI optional features required. Unfortunately, I
	  thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for
	  libpri and deleted those lines for libpri. The result was the
	  HAVE_PRI_xxx defines that control the ability to use optional
	  libpri features were also deleted. * Created
	  AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
	  features in a library that the source code could take advantage
	  of if the code supports the feature. (closes issue
	  ASTERISK-18687) Reported by: Norbert Tested by: rmudgett

	* main/udptl.c, channels/chan_sip.c: Fix debugging messages
	  generated by 'udptl debug'. * Makes chan_sip set the tag to the
	  channel name. * Fixes received debug message sequence number. *
	  Removed tx/rx debug message type since it was hard coded to 0. *
	  Made udptl.c logged message header consistent if possible: "UDPTL
	  (%s): ". * Removed unused rx_expected_seq_no from struct
	  ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P.
	  Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: Matthew Nicholson

2011-10-05 21:30 +0000 [r339566]  Leif Madsen <lmadsen@digium.com>

	* build_tools/prep_tarball: Update prep_tarball script to download
	  pre-exported documentation. I've updated the prep_tarball script
	  to now download the pre-exported documentation from the Asterisk
	  wiki. This will give us more control over what is being included
	  in the tarball releases, and will make both the PDF and HTML
	  exported documentation look much better (especially when viewing
	  from a console). (Closes issue ASTERISK-18677)

2011-12-15  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0 Released.

2011-12-09  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0-rc5 Released.

	* Fixed crash from orphaned MWI subscriptions in chan_sip

	  This patch resolves the issue where MWI subscriptions are orphaned
	  by subsequent SIP SUBSCRIBE messages.  When a peer is removed, either
	  by pruning realtime SIP peers or by unloading / loading chan_sip, the
	  MWI subscriptions that were orphaned would still be on the event engine
	  list of valid subscriptions but have a pointer to a peer that no longer
	  was valid.  When an MWI event would occur, this would cause a seg fault.

	  (closes issue ASTERISK-18663)
	  Review: https://reviewboard.asterisk.org/r/1610/

	* Don't crash on INFO automon request with no channel

	  AST-2011-014. When automon was enabled in features.conf, it was possible
	  to crash Asterisk by sending an INFO request if no channel had been
	  created yet.

	  (closes issue ASTERISK-18805)

	* Default to nat=yes; warn when nat in general and peer differ

	  AST-2011-013.  It is possible to enumerate SIP usernames when the general and
	  user/peer nat settings differ in whether to respond to the port a request is
	  sent from or the port listed for responses in the Via header. In 1.4 and
	  1.6.2, this would mean if one setting was nat=yes or nat=route and the other
	  was either nat=no or nat=never. In 1.8 and 10, this would mean when one
	  was nat=force_rport and the other was nat=no.

	  In order to address this problem, it was decided to switch the default
	  behavior to nat=yes/force_rport as it is the most commonly used option
	  and to strongly discourage setting nat per-peer/user when at all
	  possible.

	  For more discussion of the issue, please see:
	    http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html

	    (closes issue ASTERISK-18862)
	    Review: https://reviewboard.asterisk.org/r/1591/

2011-11-16  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0-rc4 Released.

	* Ensure that a null vmexten does not cause a segfault.

	  When sip_send_mwi_to_peer was modified recently to avoid deadlocks,
	  vmexten was not expected to be null.  This change handles that
	  situation to avoid a segfault.

	  (closes issue ASTERISK-18663)

2011-11-09  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0-rc3 Released.

	* Prevent BLF subscriptions from causing deadlocks

	  Fix a locking inversion in sip_send_mwi_to_peer that was causing
	  deadlocks.
	  This function now requires that both the peer and associated pvt be
	  unlocked
	  before it is called for cases where peer and peer->mwipvt form a
	  circular
	  reference.

	  (closes issue ASTERISK-18663)
	  Review: https://reviewboard.asterisk.org/r/1563/

	* Fix deadlock if peer is destroyed while sending MWI notice.

	  A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
	  because of a deadlock between the dialogs container lock and the
	  RWLOCK of the events subscription list.

	  * Create dialogs_to_destroy container to hold dialogs that will be
	  destroyed.

	  * Ensure that the event subscription callback will never happen with
	  an invalid peer pointer by making the event callback removal the first
	  thing in the peer destructor callback.

	  (closes issue ASTERISK-18747)
	  Reported by: Gregory Hinton Nietsky

	  Review: https://reviewboard.asterisk.org/r/1564/

	* Fix issue with setting defaultenabled on categories that are already
	  enabled by default.

	  (closes issue ASTERISK-18738)
	  Reported by: Paul Belanger

2011-10-18  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0-rc2 Released.

	* AST-2011-012

	* menuselect/menuselect.c: Fix --enable/--enable-category.

	------------------------------------------------------------------------
	r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
	Fix regression in configure script for libpri capability checks.

	JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
	2 persistence issues with some telcos.  ASTERISK-18535 attempted to fix
	the unexpected requirement that libpri *must* have that feature to work
	with Asterisk.  The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
	optional features required.  Unfortunately, I thought
	AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
	deleted those lines for libpri.  The result was the HAVE_PRI_xxx defines
	that control the ability to use optional libpri features were also
	deleted.

	* Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
	features in a library that the source code could take advantage of if the
	code supports the feature.
	
	(closes issue ASTERISK-18687)
	Reported by: Norbert
	Tested by: rmudgett
	------------------------------------------------------------------------
	r340878 | twilson | 2011-10-14 11:33:28 -0500 (Fri, 14 Oct 2011) | 8 lines

	Avoid unnecessary WARNING message

	Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
	displaying a WARNING message.

	(closes issue ASTERISK-18610)
	 Patch by: Kristijan_Vrban
	------------------------------------------------------------------------
	r341088 | twilson | 2011-10-17 10:35:05 -0500 (Mon, 17 Oct 2011) | 4 lines

	Don't try to remove peers without IPs from peers_by_ip

	(closes issue ASTERISK-18696)
	------------------------------------------------------------------------

2011-10-05  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.8.0-rc1 Released.

2011-10-05 21:30 +0000 [r339566]  Leif Madsen <lmadsen@digium.com>

	* build_tools/prep_tarball: Update prep_tarball script to download
	  pre-exported documentation. I've updated the prep_tarball script
	  to now download the pre-exported documentation from the Asterisk
	  wiki. This will give us more control over what is being included
	  in the tarball releases, and will make both the PDF and HTML
	  exported documentation look much better (especially when viewing
	  from a console). (Closes issue ASTERISK-18677)

2011-10-05 17:01 +0000 [r339506-339511]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Fix Dial F option notes formatting.

	* main/manager.c: Fix XML error in AMI action Challenge.

2011-10-05 16:31 +0000 [r339505]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: The app name in the documentation must match what
	  we register the application as.

2011-10-05 16:26 +0000 [r339406-339504]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c: Add missing documentation of required AMI action
	  Challenge AuthType header. (closes issue ASTERISK-18554) Reported
	  by: Vlad Povorozniuc Patches:
	  __20110919-manager-challenge-docs.patch.txt (license #4999) patch
	  uploaded by Leif Madsen

	* Makefile: Make always create the MOH directory
	  (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
	  by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
	  #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
	  Keuter

2011-10-04 19:33 +0000 [r339297-339352]  Jonathan Rose <jrose@digium.com>

	* main/say.c: Removes improper use of sound 'and' in German
	  language mode from application saynumber Asterisk would say 'Five
	  hundert und sechs und zwanzig' instead of 'Five hundert sechs und
	  zwanzig'... which is both weird sounding and wrong. This patch
	  makes sure Asterisk will only say the 'and' word between the
	  single digit and double digit places. (closes issue
	  ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
	  upstream_germand_no_and.diff (License #5402) uploaded by Lionel
	  Elie Mamane

	* res/res_jabber.c: Reverting revision 333265 due to component
	  connection problems it introduces. I'm going to attempt some
	  generic res_jabber cleanup and come up with a new fix for this
	  problem, but first it seems prudent to remove this rather broad
	  attempt to fix it and instead approach this problem either from
	  the same angle but looking only at canceling (or possibly
	  rescheduling) the send when we absolutely know it will cause a
	  segfault or, if that can't be easily accomplished, strictly from
	  the devstate side of things. Also, I'm pretty sure a lot of the
	  code in res_jabber isn't thread safe. (issue ASTERISK-18626)
	  (issue ASTERISK-18078)

2011-10-04 11:44 +0000 [r339244]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/memheap.c: fix forget declaration in previous
	  change

2011-10-03 20:12 +0000 [r339144-339147]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_sip.c: Remove duplicated Maxforwards line in AMI
	  output. (Closes issue ASTERISK-18637) Reported by: Jacek
	  Konieczny Patches: asterisk-sipshowpeer.patch (License #6298)
	  uploaded by Jacek Konieczny

	* apps/app_dial.c: Make documentation for Dial() options 'F' and
	  'F()' more clear. (Closes issue ASTERISK-18646) Reported by:
	  Physis Heckman Tested by: Richard Mudgett

2011-10-03 18:42 +0000 [r339087]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/memheap.c: destroy memheap mutex properly
	  before memheap deleted (fix memory leak occured after r304950
	  changes with DEBUG_THREAD compile option)

2011-10-03 18:40 +0000 [r339086]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c, main/file.c: Properly ignore
	  AST_CONTROL_UPDATE_RTP_PEER in more places After the change in
	  r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a
	  re-invite happens. If we receive a re-invite from a device the
	  waitstream_core was not aware of the new control frame and would
	  drop the call. (closes issue ASTERISK-18610) Reported by:
	  Kristijan_Vrban

2011-09-30 22:05 +0000 [r338800]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix segfault in analog_ss_thread() not
	  checking ast_read() for NULL. NOTE: The problem was reported
	  against v1.6.2. It is unlikely to ever happen on v1.8 and above
	  since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
	  version in sig_analog.c has largely replaced it. (closes issue
	  ASTERISK-18648) Reported by: Stephan Bosch Patches:
	  jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Stephan Bosch

2011-09-30 18:54 +0000 [r338718]  Jonathan Rose <jrose@digium.com>

	* configs/queues.conf.sample: Adds documentation for
	  QueueMemberStatus event generation

2011-09-30 16:27 +0000 [r338663]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix formatting of AMI header for SIP show
	  peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
	  issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
	  asterisk-sipshowpeer_response_end.patch (license #6298) patch
	  uploaded by Jacek Konieczny

2011-09-30 09:31 +0000 [r338609]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c, configure.ac: Remove r338137 and r338138.

2011-09-29 21:12 +0000 [r338555]  Paul Belanger <pabelanger@digium.com>

	* tests/test_linkedlists.c, tests/test_amihooks.c,
	  tests/test_security_events.c, tests/test_locale.c,
	  tests/test_logger.c, tests/test_dlinklists.c: Test modules should
	  depend on the TEST_FRAMEWORK flag

2011-09-29 20:54 +0000 [r338551]  Jason Parker <jparker@digium.com>

	* tests/test_db.c, tests/test_netsock2.c: Test modules have a
	  support level of core.

2011-09-29 18:31 +0000 [r338492]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_sip.c: Update documentation for SIP_HEADER. The
	  SIP_HEADER function only works on the the initial SIP INVITE. The
	  documentation was updated in trunk, but not in 1.8 or 10, so I'm
	  making them match. (Closes issue ASTERISK-18640)

2011-09-29 12:13 +0000 [r338416]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/sip/include/sip.h, channels/chan_sip.c: The rtptimeout
	  setting is ignored on a per peer basis. Not only is the
	  rtptimeout ignored in some cases but rtpkeepalive and
	  rtpholdtimeout is affected. this commit also removes
	  rtptimeout/rtpholdtimeout on text rtp. (closes issue
	  ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452

2011-09-28 22:35 +0000 [r338235-338322]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Make duplicate call ptr warning message more
	  helpful. * Adds the value of the call ptr to the duplicate call
	  ptr message to help trace why there is a duplicate call ptr.

	* include/asterisk/logger.h: Fix inconsistency in
	  LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue
	  ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch
	  (license #6278) patch uploaded by Luke H

2011-09-28 20:52 +0000 [r338227]  Jason Parker <jparker@digium.com>

	* tests/test_db.c, tests/test_netsock2.c, build_tools/cflags.xml,
	  channels/chan_usbradio.c, build_tools/cflags-devmode.xml,
	  agi/agi.xml, utils/utils.xml, build_tools/embed_modules.xml: Add
	  support levels to non-module sections of menuselect (cflags,
	  utils, etc).

2011-09-28 20:24 +0000 [r338224]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix chan_dahd compiling with gcc 4.6 when
	  PRI and SS7 not present. (closes issue ASTERISK-18357) Reported
	  by: Matthew Nicholson

2011-09-28 07:28 +0000 [r338137-338138]  TransNexus OSP Development <support@transnexus.com>

	* configure.ac: Updated for checking OSP Toolkit version 4.0.0.

	* apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.

2011-09-27 20:10 +0000 [r338084]  Paul Belanger <pabelanger@digium.com>

	* apps/app_macro.c: Upgrade app_macro to core

2011-09-26 19:30 +0000 [r337973]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, main/cel.c, main/manager.c,
	  funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
	  main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_manager.c,
	  cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c,
	  main/pbx.c, cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c,
	  tests/test_gosub.c, include/asterisk/cel.h: Fix deadlock when
	  using dummy channels. Dummy channels created by
	  ast_dummy_channel_alloc() should be destoyed by
	  ast_channel_unref(). Using ast_channel_release() needlessly grabs
	  the channel container lock and can cause a deadlock as a result.
	  * Analyzed use of ast_dummy_channel_alloc() and made use
	  ast_channel_unref() when done with the dummy channel. (Primary
	  reason for the reported deadlock.) * Made
	  app_dial.c:dial_exec_full() not call ast_call() holding any
	  channel locks. Chan_local could not perform deadlock avoidance
	  correctly. (Potential deadlock exposed by this issue. Secondary
	  reason for the reported deadlock since the held lock was part of
	  the deadlock chain.) * Fixed some uses of
	  ast_dummy_channel_alloc() not checking the returned channel
	  pointer for failure. * Fixed some potential chan=NULL pointer
	  usage in func_odbc.c. Protected by testing the bogus_chan value.
	  * Fixed needlessly clearing a 1024 char auto array when setting
	  the first char to zero is enough in manager.c:action_getvar().
	  (closes issue ASTERISK-18613) Reported by: Thomas Arimont
	  Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
	  uploaded by rmudgett Tested by: Thomas Arimont

2011-09-23 19:14 +0000 [r337839-337898]  Gregory Nietsky <gregory@distrotech.co.za>

	* contrib/init.d/rc.archlinux.asterisk: Spelling fix

	* apps/app_queue.c: Make sure a CDR is on the stack for call in the
	  Queue. Only let update_cdr act on the last CDR in the stack. In
	  some circumstances [Attended transfer to queue] a CDR record is
	  not inserted for this call where it should. (closes issue
	  ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266

2011-09-23 00:44 +0000 [r337774]  Russell Bryant <russell@digium.com>

	* configs/res_pktccops.conf.sample: Comment out entries in sample
	  res_pktccops.conf. With these options enabled, they can cause
	  Asterisk to freak out by SYN flooding a network and eating the
	  CPU. Obviously it would be good to fix the code so that this
	  can't happen, but we can at least change the default
	  configuration so it doesn't happen. This was reported downstream
	  to the Fedora issue tracker:
	  https://bugzilla.redhat.com/show_bug.cgi?id=658431

2011-09-22 21:29 +0000 [r337720]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Made ISDN not add numbering plan prefix
	  strings to empty numbers. When the Caller-ID is restricted, the
	  expected behavior is for the Caller-ID to be blank. In
	  chan_dahdi, the national prefix is placed onto the Caller-ID
	  number even if it is restricted (empty) causing the Caller-ID to
	  be the national prefix rather than blank. This behavior was lost
	  when sig_pri was extracted from chan_dahdi. * Made not add prefix
	  strings to empty connected line, calling, and ANI number strings.
	  (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches:
	  jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Kris Shaw

2011-09-22 11:39 +0000 [r337430-337541]  Gregory Nietsky <gregory@distrotech.co.za>

	* res/res_srtp.c: Add warned to ast_srtp to prevent errors on each
	  frame from libsrtp The first 9 frames are not reported as some
	  devices dont use srtp from first frame these are suppresed. the
	  warning is then output only once every 100 frames.

	* channels/chan_h323.c: If IP address is used in chan_h323 host
	  parameter of peer configuration. module tries to resolve IP
	  address to IP address and fails. Simple fix to set family of
	  socket this is a hangover from ipv6 changes. (closes issue
	  ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500)

	* main/channel.c: Its possible to loose audio on ast_write when the
	  channel is not transcoded correctly. in the case of DAHDI the
	  channel is hungup. This patch tries to "fix" the problem and make
	  the channel compatiable and warn the user of this problem. Please
	  note there is a underlying problem with codec negotion this does
	  not fix the problem it does try to rectify it and prevent loss of
	  service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
	  issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
	  ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
	  (issue ASTERISK-18422)

2011-09-21 21:18 +0000 [r337325-337353]  Tilghman Lesher <tilghman@meg.abyt.es>

	* apps/app_voicemail.c: More silly spacing changes

	* apps/app_voicemail.c: Dumb little spacing fix.

	* funcs/func_curl.c: Escape commas in keys and values, when keys
	  and values are enumerated by commas. Review:
	  https://reviewboard.asterisk.org/r/1433

2011-09-20 22:38 +0000 [r337118]  Matthew Jordan <mjordan@digium.com>

	* main/app.c, apps/app_followme.c, apps/app_voicemail.c,
	  apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
	  apps/app_minivm.c: Fix for incorrect voicemail duration in
	  external notifications This patch fixes an issue where the
	  voicemail duration was being reported with a duration
	  significantly less than the actual sound file duration.
	  Voicemails that contained mostly silence were reporting the
	  duration of only the sound in the file, as opposed to the
	  duration of the file with the silence. This patch fixes this by
	  having two durations reported in the __ast_play_and_record family
	  of functions - the sound_duration and the actual duration of the
	  file. The sound_duration, which is optional, now reports the
	  duration of the sound in the file, while the actual full duration
	  of the file is reported in the duration parameter. This allows
	  the voicemail applications to use the sound_duration for minimum
	  duration checking, while reporting the full duration to external
	  parties if the voicemail is kept. (issue ASTERISK-2234) (closes
	  issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
	  House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
	  https://reviewboard.asterisk.org/r/1443

2011-09-20 22:18 +0000 [r337115]  Leif Madsen <lmadsen@digium.com>

	* contrib/init.d/rc.redhat.asterisk: Update RedHat Init script to
	  work with Heartbeat. The current RedHat init script was not LSB
	  compatible. This change will make it LSB compatible so that it
	  can work correctly with Heartbeat. (Closes issue ASTERISK-18253)
	  Reported by: c0rnoTa

2011-09-20 21:04 +0000 [r337061]  Kinsey Moore <kmoore@digium.com>

	* tests/test_pbx.c, main/pbx.c: Make CANMATCH with the new pattern
	  match engine behave more like the old one When checking an
	  extension for E_CANMATCH using the new extension matching
	  algorithm, an exact match was not returned as a possible match
	  resulting in the queue failing to allow a caller to exit on DTMF.
	  This removes the requirement that an extension be longer than
	  acquired digits for an E_CANMATCH operation to succeed. (closes
	  issue ASTERISK-18044) Review:
	  https://reviewboard.asterisk.org/r/1367/

2011-09-20 19:10 +0000 [r336977-337007]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_ss7.c: Check if a channel was created before using
	  the pointer in sig_ss7_new_ast_channel(). Fixes the crash in
	  ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
	  libss7 access lock protection. * Prevent cancelling the
	  ss7_linkset() thread at inoportune times just like the
	  pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
	  Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
	  patch uploaded by rmudgett (attached to related ASTERISK-17966)

	* channels/sig_ss7.c: Fix deadlock from not releasing SS7 linkset
	  lock. sig_ss7_hangup() failed to release the SS7 linkset lock if
	  the call had the alreadyhungup flag set. * Made unlock the SS7
	  linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
	  set. * Made ss7_start_call() not hold any locks while creating
	  the channel for an incoming call to prevent deadlock. * Made
	  ss7_grab() a void function, since it could never fail, to
	  simplify calling code. * Made obtain the channel lock to do
	  softhangup in some places. Patches: jira_ast_668_v1.8.patch
	  (license #5621) patch uploaded by rmudgett JIRA AST-668

2011-09-20 00:56 +0000 [r336877]  Russell Bryant <russell@digium.com>

	* res/res_rtp_asterisk.c: Fix crashes in ast_rtcp_write(). This
	  patch addresses crashes related to RTCP handling. The backtraces
	  just show a crash in ast_rtcp_write() where it appears that the
	  RTP instance is no longer valid. There is a race condition with
	  scheduled RTCP transmissions and the destruction of the RTP
	  instance. This patch utilizes the fact that ast_rtp_instance is a
	  reference counted object and ensures that it will not get
	  destroyed while a reference is still around due to scheduled RTCP
	  transmissions. RTCP transmissions are scheduled and executed from
	  the chan_sip scheduler context. This scheduler context is
	  processed in the SIP monitor thread. The destruction of an RTP
	  instance occurs when the associated sip_pvt gets destroyed (which
	  happens when the sip_pvt reference count reaches 0). However, the
	  SIP monitor thread is not the only thread that can cause a
	  sip_pvt to get destroyed. The sip_hangup function, executed from
	  a channel thread, also decrements the reference count on a
	  sip_pvt and could cause it to get destroyed. While this is being
	  changed anyway, the patch also removes calling ast_sched_del()
	  from within the RTCP scheduler callback. It's not helpful. Simply
	  returning 0 prevents the callback from being rescheduled. (closes
	  issue ASTERISK-18570) Related issues that look like they are the
	  same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
	  (issue ASTERISK-15257) (issue ASTERISK-13334) (issue
	  ASTERISK-9977) (issue ASTERISK-9716) Review:
	  https://reviewboard.asterisk.org/r/1444/

2011-09-19 22:07 +0000 [r336791]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't interfere with T.38 reinvites This is
	  an update to the fix for ASTERISK-18340 and ASTERISK-17725

2011-09-19 20:27 +0000 [r336733]  Tilghman Lesher <tilghman@meg.abyt.es>

	* Makefile.rules, include/asterisk/optional_api.h, Makefile,
	  configure, include/asterisk/autoconfig.h.in, main/Makefile,
	  codecs/gsm/Makefile, configure.ac: Various changes to allow 1.8
	  to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6
	  extended to work on 10.7 and later. * Now uses the 'weak' symbol
	  for Lion systems, which no longer support 'weak_import' Closes
	  ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej.

2011-09-19 20:07 +0000 [r336716]  Jonathan Rose <jrose@digium.com>

	* res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c,
	  apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
	  apps/app_morsecode.c: Document applications that play audio and
	  do not answer unanswered calls. This patch is part of an effort
	  to document early media and its usage. If you are interested in
	  contributing to this documentation effort, there are probably
	  other applications worth documenting as well as an Asterisk wiki
	  article at
	  https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application

2011-09-19 18:46 +0000 [r336658]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt, apps/app_dial.c: Made Dial d and H options no longer
	  immediately auto-answer the calling leg. The Dial d and H options
	  break DTMF attended transfer atxferdropcall option. 1) Party A
	  calls party B. 2) Party B does a DTMF attended transfer to Party
	  C. If the dialplan uses the Dial d or H options to call Party C
	  then the Dial application answers the call immediately before
	  initiating the call leg to Party C. The premature answer causes
	  the transfer code to not invoke the atxferdropcall=no behavior
	  for a blonde transfer since Party C has "answered". The transfer
	  code thinks that Party B has "consulted" with Party C when Party
	  B hangs up and completes the transfer to Party A. Party A now
	  hears ringback until Party C actually answers. ASTERISK-13294
	  Dial d option. ASTERISK-11067 Dial H option to disconnect before
	  answer. The referenced issues made Dial answer with the d and H
	  options because many SIP and ISDN phones cannot send DTMF before
	  the call is connected. * Made require the dialplan to control
	  when or if the call needs to be answered to use the Dial
	  application d and H options. (The call is no longer surprise
	  answered when using the Dial d or H options.) Review:
	  https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
	  AST-666

2011-09-19 16:21 +0000 [r336591]  Jason Parker <jparker@digium.com>

	* contrib/realtime/postgresql/realtime.sql,
	  configs/cel_odbc.conf.sample, sounds/Makefile,
	  contrib/realtime/mysql/sipfriends.sql,
	  contrib/realtime/mysql/voicemail.sql, cel/cel_odbc.c, /,
	  contrib/realtime/mysql/iaxfriends.sql,
	  contrib/realtime/mysql/meetme.sql: Remove weird mergeinfo props
	  that make merges annoying sometimes.

2011-09-19 15:41 +0000 [r336572]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/get_ilbc_source.sh: Update get_ilbc_source.sh
	  script to work again. Recently iLBC support in Asterisk has
	  changed after the acquisition of GIPS by Google. More information
	  about how this may affect you is available in a blog post at:
	  http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/

2011-09-19 15:25 +0000 [r336569]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Rework sig_pri_hangup() to be simpler and
	  clearer. JIRA AST-675

2011-09-19 13:33 +0000 [r336501]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add diversion header to a 302 redirect
	  response if we have diversion data (closes issue ASTERISK-18143)
	  patch by oej

2011-09-19 13:27 +0000 [r336499]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_h323.c: A long time ago in a galaxy far far away a
	  IPv6 update was made, chan_h323 was not updated causeing all to
	  flee to chan_ooh323. the brave Jedi [asterisk developers]
	  pondered this miscarrige of justice and restored order to the
	  force for the sake of closing out 2 old issues. (closes issue
	  ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread,
	  sybasesql Tested by: irroot Reviewed by: IRC (russellb,
	  kpfleming)

2011-09-19 12:06 +0000 [r336378-336440]  Olle Johansson <oej@edvina.net>

	* main/manager.c: Make sure manager_debug option is reset at reload

	* Makefile: Revert accidental change that fixes OS/X Lion support

	* Makefile, channels/chan_sip.c: Add missing unlock at MWI message
	  sending time (closes issue ASTERISK-18573) Patches:
	  sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
	  Thanks to irrot for the reminder, to Gregory for the patch!

2011-09-16 22:10 +0000 [r336312-336314]  Terry Wilson <twilson@digium.com>

	* funcs/func_frame_trace.c: Whitespace fix

	* funcs/func_frame_trace.c: Add missing frame types to
	  func_frame_trace Also casts control frames to the proper enum so
	  that the compile will catch new additions.

2011-09-16 19:53 +0000 [r336294]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/frame.h, main/channel.c, main/rtp_engine.c,
	  channels/chan_sip.c: Fix bad RTP media bridges in directmedia
	  calls on peers separated by multiple Asterisk nodes. In a
	  situation involving devices on separate Asterisk trunks, the
	  remote RTP bridge would break when starting a call with
	  directmedia. This patch queues a new type of control frame so
	  that our RTP bridge loop can properly detect when these
	  situations occur and check to see if peers need to be updated in
	  order to send their media to the proper location. (Closes issue
	  ASTERISK-18340) Reported by: Thomas Arimont (Closes issue
	  ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose

2011-09-16 19:06 +0000 [r336234]  Sean Bright <sean@malleable.com>

	* UPGRADE.txt: Make a note that inotify won't work with an NFS
	  mounted spooler directory.

2011-09-16 10:09 +0000 [r335978-336166]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_misdn.c: The round robin routing routine in
	  chan_misdn.c is broken. it rotates between ports but never checks
	  the channels in the ports. i have extensivly tested it and
	  verified it works on 1 upto 4 ports. before the patch only 1 out
	  of each port was used now all are used as expected. (closes issue
	  ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed
	  by: irroot Review: https://reviewboard.asterisk.org/r/1410/

	* apps/app_queue.c: Locking order in app_queue.c causes deadlocks.
	  a channel lock must never be held with the queues container lock
	  held. the deadlock occured on masquerade. the queues container
	  lock is a relic of the past the old queue module lock. with ao2
	  there is no need to hold this lock when dealing with members this
	  patch removes unneeded locks. (closes issue ASTERISK-18101)
	  (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason
	  Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by:
	  Matthew Nicholson Review:
	  https://reviewboard.asterisk.org/r/1402/

	* channels/chan_agent.c: lock the channel before calling
	  ast_bridged_channel() to prevent a seg fault. AMI agents list
	  called on shutdown causes a segfault, introducing proper locking
	  will prevent this. (closes issue ASTERISK-18092) Reported by:
	  agustina Patches: chan_agent.patch (License #5041) patch uploaded
	  by irroot

2011-09-14 18:21 +0000 [r335851-335911]  Richard Mudgett <rmudgett@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Remove
	  unnecessary libpri dependency checks in the configure script.
	  Using the --with-pri option with the configure script generated
	  an error about not having PRI_L2_PERSISTENCE if you did not have
	  the absolute latest libpri SVN checkout installed. The
	  AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script
	  seems to be for libraries that are dependent upon other libraries
	  and not necessarily for optional/added features within a library.
	  (closes issue ASTERISK-18535) Reported by: Michael Keuter

	* channels/chan_dahdi.c: Fixed cut-n-paste regression using the
	  wrong variable. Fixes the missing DAHDI channels when using the
	  newer chan_dahdi.conf sections for channel configuration. (closes
	  issue ASTERISK-18496) Reported by: Sean Darcy Patches:
	  jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: Sean Darcy, rmudgett

2011-09-14 13:28 +0000 [r335790]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c: The tech and data members of
	  fast_originate_helper are not string fields. ASTERISK-17709

2011-09-13 22:10 +0000 [r335720]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_directed_pickup.c: Remove obsolete todo comment about
	  PICKUPRESULT.

2011-09-13 21:33 +0000 [r335716]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
	  parse the option "defaultlanguage" from the [options] section of
	  asterisk.conf, as in the sample config file. Otherwise the
	  build-time default language (normally "en") is always the default
	  one. Review: https://reviewboard.asterisk.org/r/1342/
	  Signed-off-by: Tzafrir Cohen (License #5035)
	  <tzafrir.cohen@xorcom.com>

2011-09-13 21:30 +0000 [r335714]  Paul Belanger <pabelanger@digium.com>

	* apps/app_meetme.c: Meetme should have 'core' support level
	  (closes issue ASTERISK-18542)

2011-09-13 18:52 +0000 [r335655]  Tilghman Lesher <tilghman@meg.abyt.es>

	* configure, configure.ac: Move mandatory checks closer to the
	  beginning of the file. If these are going to fail, they should
	  fail as quickly as possible.

2011-09-13 18:20 +0000 [r335618]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, main/manager.c: Don't limit the size of appdata for
	  manager originate actions. ASTERISK-17709 Patch by: tilghman
	  (with modifications)

2011-09-13 07:11 +0000 [r335497]  Russell Bryant <russell@digium.com>

	* main/event.c, include/asterisk/event.h, res/ais/evt.c: Fix a
	  crash in res_ais. This patch resolves a crash observed in a load
	  testing environment that involved the use of the res_ais module.
	  I observed some crashes where the event delivery callback would
	  get called, but the length parameter incidcating how much data
	  there was to read was 0. The code assumed (with good reason I
	  would think) that if this callback got called, there was an event
	  available to read. However, if the rare case that there's nothing
	  there, catch it and return instead of blowing up. More
	  specifically, the change always ensure that the size of the
	  received event in the cluster is always big enough to be a real
	  ast_event. Review: https://reviewboard.asterisk.org/r/1423/

2011-09-12 15:54 +0000 [r335431-335433]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Properly set caller_warning and callee_warning
	  before we try to use them. ASTERISK-18199 Patch by: elguero
	  Testing by: rtang

	* bridges/bridge_multiplexed.c: Prevent a race condition when the
	  bridge technology changes. This change was ported from asterisk
	  10. ASTERISK-18155

2011-09-12 14:21 +0000 [r335320-335341]  Kinsey Moore <kmoore@digium.com>

	* apps/app_dial.c: Ensure frames are not written to dialed channel
	  if ringback is requested When a single channel was dialed and
	  there was media to be forwarded to the calling channel, the media
	  was written without regard for ringback causing silence to be
	  heard in some circumstances. This regression was introduced when
	  the meaning of "single" changed to mean only the number of
	  channels dialed. (closes issue ASTERISK-18083)

	* channels/chan_iax2.c: Prevent IAX2 from getting IPv6 addresses
	  via DNS IAX2 does not support IPv6 and getting such addresses
	  from DNS can cause error messages on the remote end involving bad
	  IPv4 address casts in the presence of IPv6/IPv4 tunnels. This
	  patch ensures that IAX2 will not encounter IPv6 addresses via DNS
	  queries. (closes issue ASTERISK-18090)

2011-09-12 13:25 +0000 [r335319]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Lock the peer->mvipvt to avoid crashes with
	  SIP history enabled After the launch of 1.6 event-based MWI we
	  have two threads handling the peer->mwipvt, which cause issues
	  with SIP history additions in combination with the max limit for
	  number of history entries. Review:
	  https://reviewboard.asterisk.org/r/1373/ (closes issue
	  ASTERISK-18288) Thanks to irrot for peer review. Work with this
	  bug funded by IPvision AS

2011-09-12 11:09 +0000 [r335259]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: build_peer doesnt unlink a peer object from
	  peers_by_ip container which leads to a wrong refcounter value.
	  adding an ao2_unlink from the peers_by_ip container fix it.
	  Review: https://reviewboard.asterisk.org/r/1428/

2011-09-09 16:09 +0000 [r335064]  Matthew Jordan <mjordan@digium.com>

	* channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
	  main/channel.c, channels/chan_usbradio.c, main/dial.c,
	  channels/chan_dahdi.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, funcs/func_frame_trace.c,
	  main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
	  include/asterisk/frame.h, channels/sig_ss7.c,
	  channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
	  main/pbx.c, addons/chan_ooh323.c, channels/chan_sip.c: Updated
	  SIP 484 handling; added Incomplete control frame When a SIP phone
	  uses the dial application and receives a 484 Address Incomplete
	  response, if overlapped dialing is enabled for SIP, then the 484
	  Address Incomplete is forwarded back to the SIP phone and the
	  HANGUPCAUSE channel variable is set to 28. Previously, the
	  Incomplete application dialplan logic was automatically
	  triggered; now, explicit dialplan usage of the application is
	  required. Additionally, this patch adds a new AST_CONTOL_FRAME
	  type called AST_CONTROL_INCOMPLETE. If a channel driver receives
	  this control frame, it is an indication that the dialplan expects
	  more digits back from the device. If the device supports overlap
	  dialing it should attempt to notify the device that the dialplan
	  is waiting for more digits; otherwise, it can handle the frame in
	  a manner appropriate to the channel driver. (closes issue
	  ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew
	  Jordan Review: https://reviewboard.asterisk.org/r/1416/

2011-09-08 22:27 +0000 [r334953]  Richard Mudgett <rmudgett@digium.com>

	* main/logger.c: Fix crash with res_fax when MALLOC_DEBUG and "core
	  stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
	  enabled when res_fax tries to unregister its logger level. * Make
	  ast_logger_unregister_level() use ast_free() instead of free().
	  When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
	  a call to free(). Therefore, if you allocated memory with a form
	  of ast_malloc you must free it with ast_free.

2011-09-07 19:35 +0000 [r334843]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_iax2.c: Cleanup chan_iax2.c log messages Review:
	  https://code.asterisk.org/code/cru/CR-AST-11

2011-09-07 19:31 +0000 [r334840]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix AMI action Park crash. * Made AMI action
	  Park not say anything to the parker channel (AMI header Channel2)
	  since the AMI action is a third party parking the call. (This is
	  a change in behavior that cannot be preserved without a lot of
	  effort.) * Made not play pbx-parkingfailed if the Park 's' option
	  is used. JIRA AST-660

2011-09-07 13:26 +0000 [r334682]  Stefan Schmidt <sst@sil.at>

	* main/features.c: Adding the Feature to sent a Reason Header in a
	  SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
	  before doing a masquerade in the pickup function.

2011-09-07 08:12 +0000 [r334616-334620]  Alec L Davis <sivad.a@paradise.net.nz>

	* CHANGES, apps/app_queue.c: peroid typo

	* main/pbx.c: Prevent segfault if call arrives before Asterisk is
	  fully booted. Prevent ast_pbx_start and ast_run_start from
	  starting a new thread unless asterisk is fully booted. alecdavis
	  (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1407/

2011-09-06 13:48 +0000 [r334453]  Gregory Nietsky <gregory@distrotech.co.za>

	* apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
	  LIMIT is not portable. Regression from r312212 (closes issue
	  ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
	  Review: https://reviewboard.asterisk.org/r/1415/

2011-09-23  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.7.0 Released.

2011-09-19  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.7.0-rc2 Released.

	* r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) |
	11 lines

	Fixed cut-n-paste regression using the wrong variable.

	Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
	sections for channel configuration.

	(closes issue ASTERISK-18496)

	* r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) |
	13 lines

	Remove unnecessary libpri dependency checks in the configure script.

	Using the --with-pri option with the configure script generated an
	error
	about not having PRI_L2_PERSISTENCE if you did not have the absolute
	latest libpri SVN checkout installed.

	The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems
	to
	be for libraries that are dependent upon other libraries and not
	necessarily for optional/added features within a library.

	(closes issue ASTERISK-18535)

	* r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7
	lines

	Update get_ilbc_source.sh script to work again.

	Recently iLBC support in Asterisk has changed after the acquisition of
	GIPS
	by Google. More information about how this may affect you is available
	in a
	blog post at:

	  http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/

	* r335714 | pabelanger | 2011-09-13 16:30:18 -0500 (Tue, 13 Sep 2011)
	| 4 lines

	Meetme should have 'core' support level

	(closes issue ASTERISK-18542)

2011-09-07  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.7.0-rc1 Released.

2011-09-06 13:48 +0000 [r334453]  Gregory Nietsky <gregory@distrotech.co.za>

	* apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
	  LIMIT is not portable. Regression from r312212 (closes issue
	  ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
	  Review: https://reviewboard.asterisk.org/r/1415/

2011-09-02 20:59 +0000 [r334296-334355]  Richard Mudgett <rmudgett@digium.com>

	* res/res_musiconhold.c: MusicOnHold has extra unref which may lead
	  to memory corruption and crash. The problem happens when a call
	  is disconnected and you had started a MOH class that does not use
	  the files mode. If you define REF_DEBUG and recreate the problem,
	  it will announce itself with the following warning: Attempt to
	  unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
	  class is still in a container! * Fixed moh_alloc() and
	  moh_release() functions not handling the state->class reference
	  consistently. (closes issue ASTERISK-18346) Reported by: Mark
	  Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
	  Review: https://reviewboard.asterisk.org/r/1404/

	* main/config.c, include/asterisk/config.h: Fix potential memory
	  allocation failure crashes in config.c. * Added required checks
	  to the returned memory allocation pointers to prevent crashes. *
	  Made ast_include_rename() create a replacement ast_variable list
	  node if the new filename is longer than the available space.
	  Fixes potential crash and memory leak. * Factored out
	  ast_variable_move() from ast_variable_update() so
	  ast_include_rename() can also use it when creating a replacement
	  ast_variable list node. * Made the filename stuffed at the end of
	  the struct a minimum allocated size in ast_variable_new() in case
	  ast_include_rename() changes the stored filename. * Constify
	  struct char pointers pointing to strings stuffed at the end of
	  the struct for: ast_variable, cache_file_mtime, and
	  ast_config_map. * Factored out cfmtime_new() to remove inlined
	  code and allow some struct pointers to become const. * Removed
	  the list lock from struct cache_file_mtime that was never used. *
	  Added doxygen comments to several structure elements and better
	  documented what strings are stuffed at the struct end char array.
	  * Reworked ast_config_text_file_save() and set_fn() to handle
	  allocation failure of the include file scratch pad object
	  tracking blank lines. * Made ast_config_text_file_save() fn[]
	  declared with PATH_MAX to ensure it is long enough for any
	  filename with path. Also reduced the number of container fileset
	  buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
	  https://reviewboard.asterisk.org/r/1378/

2011-09-01 17:38 +0000 [r334229-334234]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/pbx.c: Remove 1.6 compatibility documentation from 1.8, as
	  it no longer applies.

	* res/res_config_odbc.c: Create a local alias for
	  ast_odbc_clear_cache. As a function pointer, the reference has to
	  be resolved at load time irrespective of the RTLD_LAZY flag.
	  Creating a local alias solves this problem, because the structure
	  is initialized with that local function pointer, while the actual
	  function can remain lazily linked until runtime. The reason why
	  this is important is because we lazily load function references
	  during the module loading process, in order to obtain priority
	  values for each module, ensuring that modules are loaded in the
	  correct order. Previous to this change, when this module was
	  initially loaded, the module loader would emit a symbol
	  resolution error, because of the above requirement. Closes
	  ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
	  Walter Doekes, patch by me)

2011-08-31 18:50 +0000 [r334156]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Disable T.38 when we get a invite with image
	  media port set to 0 ASTERISK-17678

2011-08-31 15:57 +0000 [r334009-334012]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: No DAHDI channel available for conference,
	  user introduction disabled. The following error will consistently
	  occur when trying to dial into a MeetMe conference when the
	  server does not have DAHDI hardware installed: app_meetme.c: No
	  DAHDI channel available for conference, user introduction
	  disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
	  correctly during compilation and install of Asterisk/Dahdi,
	  including associated modules, etc., a chan_dahdi.conf
	  configuration file in /etc/asterisk is not created by FreePBX if
	  hardware does not exist, causing MeetMe to be unable to open a
	  DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
	  channel when there is no chan_dahdi.conf file to load. (closes
	  issue ASTERISK-17398) Reported by: Preston Edwards Patches:
	  jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett

	* main/channel.c, channels/chan_agent.c: Call pickup race leaves
	  orphaned channels or crashes. Multiple users attempting to pickup
	  a call that has been forked to multiple extensions either crashes
	  or fails a masquerade with a "bad things may happen" message.
	  This is the scenario that is causing all the grief: 1) Pickup
	  target is selected 2) target is marked as being picked up in
	  ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app
	  dial or queue gets a chance to hang up losing calls and calls
	  ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP
	  YET BY ast_do_pickup() with ast_channel_masquerade(),
	  ast_hangup() completes successfully and the channel is no longer
	  in the channels container. 6) ast_do_pickup() then calls
	  ast_channel_masquerade() to schedule the masquerade on the dead
	  channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the
	  dead channel 8) bad things happen while doing the masquerade and
	  in the process ast_do_masquerade() puts the dead channel back
	  into the channels container 9) The "orphaned" channel is visible
	  in the channels list if a crash does not happen. This patch does
	  the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a
	  successfully hung-up channel and not release the channel lock
	  until that has happened. * Made __ast_channel_masquerade() not
	  setup a masquerade if either channel has AST_FLAG_ZOMBIE set. *
	  Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer
	  work. (closes issue ASTERISK-18222) Reported by: Alec Davis
	  Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes
	  issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by:
	  rmudgett, Alec Davis, irroot, Karsten Wemheuer Review:
	  https://reviewboard.asterisk.org/r/1400/

2011-08-31 15:18 +0000 [r334006]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Correct an AMI protocol violation with
	  SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n".
	  Since other commands are ended by using \r\n this confuses any
	  interfacing script. (closes issue ASTERISK-17486)

2011-08-30 21:16 +0000 [r333947]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c,
	  addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooCalls.h:
	  cleanups in ACF/ARJ GK replies processing fixed long (24 sec)
	  pause if acf/arj proccessed before ast_cond_wait called to wait
	  this

2011-08-29 21:38 +0000 [r333836]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Refresh peer address if DNS unavailable at
	  peer creation If Asterisk starts and no DNS is available,
	  outbound registrations will fail indefinitely. This patch copies
	  the address from the sip_registry struct, which will be updated,
	  to the peer->addr when necessary. If dnsmgr is enabled, the
	  registration fails without the patch because even though the
	  address on the registry is updated via dnsmgr, the address is
	  just copied on the first try. Since we use ast_sockaddr_copy,
	  dnsmgr can't update the address that is copied to the sip_pvt or
	  peers. Closes issue ASTERISK-18000 Review:
	  https://reviewboard.asterisk.org/r/1335/

2011-08-29 21:06 +0000 [r333784-333785]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h: Add some do not hold locks notes to
	  channel.h

	* addons/chan_mobile.c: Fix deadlock potential of
	  chan_mobile.c:mbl_ast_hangup().

2011-08-29 17:11 +0000 [r333630]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Fixed improperly formatted TestEvent AMI
	  message in app_voicemail

2011-08-29 15:55 +0000 [r333569]  Jonathan Rose <jrose@digium.com>

	* res/res_jabber.c: Accidental use of variable client->status
	  instead of client->state in from ASTERISK-18078 (issue
	  ASTERISK-18078)

2011-08-28 09:49 +0000 [r333507]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
	  GCC 4.6 detects variables that get assined to, but never used
	  later. Also removes some remmed-out lines that become invalid.
	  (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
	  (License #5035) <tzafrir.cohen@xorcom.com>,

2011-08-26 16:19 +0000 [r333378]  Jonathan Rose <jrose@digium.com>

	* res/res_jabber.c: [patch] Buddies are always auto-registered when
	  processing the roster Reporter said autoregister flag was ignored
	  for registering 'buddies' which had a subscription to us.
	  Verified that this was the case and observed how the patch
	  addressed this and made sure it didn't break anything. (closes
	  issue ASTERISK-14233) Reported by: Simon Arlott Patches:
	  asterisk-0015229.patch (license #5756) patch uploaded by Simon
	  Arlott Tested by: Jonathan Rose

2011-08-26 14:36 +0000 [r333339-333354]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Fixed incorrect pointer copy to structure
	  copy in revision 333339

	* apps/app_voicemail.c: Bug fixes for voicemail user emailsubject /
	  emailbody. This code change fixes a few issues with the voicemail
	  user override of emailbody and emailsubject, including escaping
	  the strings, potential memory leaks, and not overriding the
	  voicemail defaults. Revision 325877 fixed this for
	  ASTERISK-16795, but did not fix it for ASTERISK-16781. A
	  subsequent check-in prevented 325877 from being applied to 10.
	  This check-in resolves both issues, and applies the changes to
	  1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by:
	  Sebastien Couture Tested by: mjordan (closes issue
	  ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review:
	  https://reviewboard.asterisk.org/r/1374

2011-08-25 19:00 +0000 [r333267]  Jason Parker <jparker@digium.com>

	* Makefile: Fix for DESTDIR spaces patch.

2011-08-25 18:47 +0000 [r333265]  Jonathan Rose <jrose@digium.com>

	* res/res_jabber.c: Segfault when publishing device states via XMPP
	  and not connected When using publishing device state with
	  res_jabber, Asterisk will attempt to send a device state using
	  the unconnected client using iks_send_raw and crash. This patch
	  checks the validity of the connection before attempting to send
	  the device state. (closes issue ASTERISK-18078) Reported by:
	  Michael L. Young Patches:
	  res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
	  patch uploaded by Michael L. Young Tested by: Jonathan Rose

2011-08-25 15:27 +0000 [r333201]  Jason Parker <jparker@digium.com>

	* makeopts.in, sounds/Makefile, Makefile, build_tools/mkpkgconfig,
	  configure, configure.ac: Fix installation into directories
	  containing spaces. This also vastly simplifies the logic in
	  sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul
	  Belanger Review: https://reviewboard.asterisk.org/r/1379/

2011-08-23 18:14 +0000 [r333010]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Memory Leak in app_queue The patch that was
	  committed in the 1.6.x versions of Asterisk for ASTERISK-15862
	  actually fixed two issues. One was not applicable to 1.8 but the
	  other is. queue_leak.patch fixes the portion applicable to 1.8.
	  (closes issue ASTERISK-18265) Reported by: Fred Schroeder
	  Patches: queue_leak.patch (license #5049) patch uploaded by
	  mmichelson Tested by: Thomas Arimont

2011-08-23 18:11 +0000 [r333009]  Matthew Nicholson <mnicholson@digium.com>

	* UPGRADE.txt, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: default 'sipstorecause' to no We've
	  decided to disable this feature by default in future 1.8
	  versions. This would be an unexpected behavior change for anyone
	  depending on that SIP_CAUSE update in their dialplan. Please
	  refer to the asterisk-dev mailing list more information:
	  http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
	  (issue AST-580)

2011-08-22 22:11 +0000 [r332939-332945]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, main/config.c, include/asterisk/config.h:
	  Revert previous commit. Not ready yet.

	* apps/app_queue.c, main/config.c, include/asterisk/config.h:
	  Signed

	* main/config.c: Minor code optimizations. * Simplify
	  ast_category_browse() logic for easier understanding. * Remove
	  dead code in ast_variable_delete() and simplify some of its
	  logic.

2011-08-22 19:41 +0000 [r332876]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_gtalk.c: Revert previous commit It seems google is
	  still making changes to the protocol. (issue ASTERISK-18301)

2011-08-22 19:32 +0000 [r332874]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Reference leaks in app_queue. * Fixed
	  load_realtime_queue() leaking a queue reference when it
	  overwrites q when processing a realtime queue. (issue
	  ASTERISK-18265) * Make join_queue() unreference the queue
	  returned by load_realtime_queue() when it is done with the
	  pointer. The load_realtime_queue() returns a reference to the
	  just loaded realtime queue. * Fixed queues container reference
	  leak in queues_data_provider_get(). * queue_unref() should not
	  return q that was just unreferenced. * Made logic in
	  __queues_show() and queues_data_provider_get() when calling
	  load_realtime_queue() easier to understand.

2011-08-22 18:15 +0000 [r332817]  Matthew Jordan <mjordan@digium.com>

	* main/app.c, configs/manager.conf.sample,
	  include/asterisk/manager.h, apps/app_voicemail.c,
	  include/asterisk/test.h, main/manager.c, main/file.c,
	  main/test.c: Review: https://reviewboard.asterisk.org/r/1364/
	  This update adds a new AMI event, TestEvent, which is enabled
	  when the TEST_FRAMEWORK compiler flag is defined. It also adds
	  initial usage of this event to app_voicemail. The TestEvent AMI
	  event is used extensively by the voicemail tests in the Asterisk
	  Test Suite.

2011-08-22 18:14 +0000 [r332759-332816]  Richard Mudgett <rmudgett@digium.com>

	* res/res_config_pgsql.c, res/res_config_odbc.c: Memory leaks in
	  realtime_multi_xxx() when database access returns error. * Fix
	  realtime_multi_pgsql() configuration memory leak when the
	  database access returns an error. * Fix realtime_multi_odbc()
	  configuration category use after free when the database access
	  returns an error.

	* main/config.c: Memory leak reading realtime database variable
	  list. Calling ast_load_realtime() can leak the last list node if
	  the read list only contains empty variable value items. * Fixed
	  list filter loop in ast_load_realtime() to delete the list node
	  immediately instead of the next time through the loop. The next
	  time through the loop may not happen if the node to delete is the
	  last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
	  Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
	  patch uploaded by rmudgett

2011-08-21 14:31 +0000 [r332699]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_gtalk.c: Fix outgoing calls in chan_gtalk (closes
	  issue ASTERISK-18301) Reported by: az1324

2011-08-18 21:26 +0000 [r332559]  Terry Wilson <twilson@digium.com>

	* main/netsock2.c: Fix possible error on stringification of
	  IPv4-mapped addrs The FreeBSD netsock2 test has been failing for
	  a while. We were pasing sa->len to getnameinfo instead of
	  sa_tmp->len. ASTERISK-18289

2011-08-18 19:28 +0000 [r332503]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_dahdi.c: CRC4 in "dahdi show status" gives wrong
	  impression to T1 users Change CRC4 to CRC in the output of "dahdi
	  show status" so that it can apply in more situations without
	  confusing users, especially since T1 lines use CRC6 instead of
	  CRC4. (closes issue AST-471)

2011-08-18 14:46 +0000 [r332355-332446]  Tilghman Lesher <tilghman@meg.abyt.es>

	* build_tools/cflags.xml, build_tools/cflags-devmode.xml: Move
	  BETTER_BACKTRACES out of development mode, as it's useful when
	  DEBUG_THREADS is enabled.

	* makeopts.in, sounds/Makefile, Makefile, agi/Makefile,
	  utils/Makefile, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, Makefile.moddir_rules: Re-add support for spaces in
	  pathnames, including now spaces in DESTDIR. This was initially
	  added to 1.8 prior to release, primarily to support the standard
	  paths on Mac OS X, but was partially reverted recently in
	  Subversion, due to the lack of support for spaces in DESTDIR.
	  This commit restores support for the standard paths on Mac OS X,
	  and also includes support for spaces in DESTDIR. (closes issue
	  ASTERISK-18290) Reported by: pabelanger Review:
	  https://reviewboard.asterisk.org/r/1326/

2011-08-17 17:35 +0000 [r332320]  Terry Wilson <twilson@digium.com>

	* res/res_timing_timerfd.c: Don't read from a disarmed or invalid
	  timerfd Numerous isues have been reported for deadlocks that are
	  caused by a blocking read in res_timing_timerfd on a file
	  descriptor that will never be written to. This patch adds some
	  checks to make sure that the timerfd is both valid and armed
	  before calling read(). Should fix: ASTERISK-1842, ASTERISK-18197,
	  ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others.

2011-08-17 15:51 +0000 [r332264]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Outgoing BRI
	  calls fail when using Asterisk 1.8 with HA8, HB8, and B410P
	  cards. France Telecom brings layer 2 and layer 1 down on BRI
	  lines when the line is idle. When layer 1 goes down Asterisk
	  cannot make outgoing calls and the HA8 and HB8 cards also get IRQ
	  misses. The inability to make outgoing calls is because the line
	  is in red alarm and Asterisk will not make calls over a line it
	  considers unavailable. The IRQ misses for the HA8 and HB8 card
	  are because the hardware is switching clock sources from the line
	  which just brought layer 1 down to internal timing. There is a
	  DAHDI option for the B410P card to not tell Asterisk that layer 1
	  went down so Asterisk will allow outgoing calls: "modprobe
	  wcb4xxp teignored=1". There is a similar DAHDI option for the HA8
	  and HB8 cards: "modprobe wctdm24xxp bri_teignored=1".
	  Unfortunately that will not clear up the IRQ misses when the
	  telco brings layer 1 down. * Add layer 2 persistence option to
	  customize the layer 2 behavior on BRI PTMP lines. The new option
	  has three settings: 1) Use libpri default layer 2 setting. 2)
	  Keep layer 2 up. Bring layer 2 back up when the peer brings it
	  down. 3) Leave layer 2 down when the peer brings it down. Layer 2
	  will be brought up as needed for outgoing calls. JIRA AST-598

2011-08-17 14:31 +0000 [r332234]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: print a warning instructing the user to
	  disable storesipcause if we process 100 or more scheduler entries
	  at a time AST-580

2011-08-16 20:10 +0000 [r332176]  Paul Belanger <pabelanger@digium.com>

	* tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c,
	  tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
	  tests/test_func_file.c, tests/test_security_events.c,
	  tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
	  tests/test_locale.c, tests/test_acl.c, tests/test_devicestate.c,
	  tests/test_utils.c, tests/test_aoc.c, tests/test_astobj2.c,
	  tests/test_poll.c, tests/test_amihooks.c,
	  tests/test_substitution.c, tests/test_heap.c,
	  tests/test_ast_format_str_reduce.c, tests/test_expr.c,
	  tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
	  tests/test_dlinklists.c, tests/test_event.c: Flag test modules as
	  'core' Review: https://reviewboard.asterisk.org/r/1369/

2011-08-16 17:38 +0000 [r332118]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: ASTERISK-18067 ASTERISK-15479 - White Space
	  affects mailbox value, multiple MWI subs Before, having multiple
	  subscriptions to mailboxes on a sip peer set via the mailbox
	  setting in sip.conf would only result in updates being sent on
	  whichever mailbox triggered the mwi event. Now all of them get
	  counted regardless. Also fixes a bug involving parsing of the
	  mailbox option in sip.conf so that trailing and leading spaces
	  before/after commas are trimmed. (closes issue ASTERISK-18067)
	  Reported by: aragon (closes issue ASTERISK-15479) Reported by:
	  Ben Winslow Patches:
	  chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
	  patch uploaded by Ben Winslow

2011-08-16 16:31 +0000 [r332100]  Richard Mudgett <rmudgett@digium.com>

	* CHANGES, configs/features.conf.sample, main/asterisk.c,
	  main/features.c: Fix multiple parking issues. JIRA ASTERISK-17183
	  Multi-parkinglot directs calls to wrong parkinglot. JIRA
	  ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
	  ParkedCall() with no extension should pickup first available call
	  and does not. JIRA AST-576 Issues with parking lots * Removed
	  searching for parking lots by extension. Parking lots can only be
	  found by the parking lot name since parking lot access extensions
	  and spaces are not guaranteed to be unique. * Added
	  parking_lot_name option to the Park and ParkedCall applications.
	  Updated documentation for Park and ParkedCall applications. * Add
	  parkext_exclusive configuration option to make parking entry
	  extensions specify which parking lot they access. (closes issue
	  ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
	  David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
	  Quezada (closes issue ASTERISK-17430) Reported by: Philippe
	  Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
	  AST-624 'next' setting for findslot does nothing * Reimplemented
	  since findslot feature option broken by -r114655. (closes issue
	  ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
	  JIRA ASTERISK-15792 Dialplan continues execution after transfer
	  to park. This happens for DTMF attended transfer, DTMF blind
	  transfer, and DTMF one-touch-parking if the party initiating
	  these features also initiated the call. * Fixed the return code
	  from the affected builtin features when parking a call. (closes
	  issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
	  rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
	  the expected call when picking up a parked call. This is mostly a
	  documentation problem. However, the option is not reset to the
	  default when features.conf is reloaded. * Updated
	  features.conf.sample documentation for courtesytone and
	  parkedplay options. * Reset the parkedplay option to default when
	  features.conf is reloaded. JIRA AST-615 AMI Park action followed
	  by features reload results in orphaned channels in parking lot. *
	  Reloading features.conf will not touch parking lots that have
	  calls still parked in them. Reload again at a later time. Misc
	  additional fixes: * Added unit test for parking lot dialplan
	  usage checking. * Made update connected line when a parked call
	  is retrieved from a parking lot. * Made retrieved parked call
	  stop ringing or MOH depending upon how the call was waiting in
	  the parking lot. * Made CLI "features show" indicate if the
	  parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
	  variable to allow dynamic parking lots to specify the parking lot
	  access extension. * Made AMI ParkedCalls action ParkedCall events
	  have a Parkinglot header. * Made AMI ParkedCalls action
	  ParkedCallsComplete event have a Total header. * Fixed potential
	  deadlock from AMI Park action holding channel locks while calling
	  masq_park_call(). * Fixed several places where ast_strdupa() were
	  used inside of loops. (Mostly fixed by refactoring the loop body
	  into its own function.) * Fixed copy_parkinglot() copying too
	  much from the source parking lot. Extracted the parking lot
	  configuration settings into struct parkinglot_cfg. * Refactored
	  courtesytone playing code to put the channel not playing the tone
	  in autoservice. * Fix when pbx-parkingfailed is played that the
	  other channel is put in autoservice if it exists. * Fixed
	  parkinglot reference leak in parked_call_exec() error paths. *
	  Fixed parkinglot_unref() use of parkinglot after it was unreffed.
	  * Made destroy the struct ast_parkinglot parkings lock when done.
	  * Refactored the features.conf parking lot configuration code to
	  eliminate redundancy. * Fixed feature reload to better protect
	  parking lots. * Fixed parking lot container reference leak in
	  handle_parkedcalls(). * Fixed the total count in
	  handle_parkedcalls(). Review:
	  https://reviewboard.asterisk.org/r/1358/

2011-08-16 15:06 +0000 [r332021-332026]  Matthew Nicholson <mnicholson@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: use
	  DEFAULT_STORE_SIP_CAUSE to set the default value for the
	  'storesipcause' option AST-580

	* configs/sip.conf.sample, CHANGES, channels/chan_sip.c: Added the
	  'storesipcause' option to sip.conf to allow the user to disable
	  the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having
	  chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
	  significant performance penalty because of the usage of the
	  MASTER_CHANNEL() dialplan function. AST-580

2011-08-15 17:24 +0000 [r331955]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix some minor chan_dahdi config load
	  issues. * Address chan_dahdi.conf dahdichan option todo item
	  about needing line number. * Make ignore_failed_channels option
	  also apply to dahdichan option. * Don't attempt to create a
	  default pseudo channel if the chan_dahdi.conf channel/channels
	  option is not allowed. * Add a similar check for dahdichan in
	  normal chan_dahdi.conf sections as is done in users.conf.

2011-08-15 15:21 +0000 [r331886]  Paul Belanger <pabelanger@digium.com>

	* main/rtp_engine.c: Fix noisy message when briding channels
	  (closes issue ASTERISK-18270) Reported by: Federico Alves

2011-08-15 15:12 +0000 [r331867]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes locking inversion issues present in
	  the handling of the sip REFER method. (closes issue
	  ASTERISK-18082) Reported by: James Van Vleet

2011-08-12 19:01 +0000 [r331774]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c: Unlock the channel before calling update_queue.
	  Holding the channel lock when calling update_queue which attempts
	  to lock the queue lock can cause a deadlock. This deadlock
	  involves the following chain: 1. hold chan lock -> wait queue
	  lock 2. hold queue lock -> wait agent list lock 3. hold agent
	  list lock -> wait chan list lock 4. hold chan list lock -> wait
	  chan lock

2011-08-12 18:58 +0000 [r331714-331771]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Suppress warning message when using
	  DAHDITransfer or DAHDIHangup. * The fake event should only be
	  processed by the channel that currently owns the private and not
	  the associated call waiting or 3-way channel. JIRA AST-620 JIRA
	  SWP-3616

	* channels/chan_dahdi.c: AMI actions DAHDIHangup and DAHDITransfer
	  have no effect. The AMI actions DAHDIHangup and DAHDITransfer
	  have no effect on a DAHDI channel. These two AMI actions are
	  highly specialized to analog channels and appear to make the
	  channel behave like a jack port for headsets. * Made the faked
	  DAHDI event get processed before a normal media stream read in
	  dahdi_read() instead of trying to trigger an exception read by
	  setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
	  long ago that changed how AST_FLAG_EXCEPTION is processed in the
	  core. Unfortunately, the faked DAHDI events no longer worked when
	  that happened. * Updated the DAHDI AMI action documentation for
	  the following actions: DAHDITransfer, DAHDIHangup,
	  DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
	  DAHDIRestart. * Made use sscanf() instead of atoi() for better
	  error checking of the DAHDIChannel header string. JIRA AST-620
	  JIRA SWP-3616

2011-08-12 16:30 +0000 [r331658]  Terry Wilson <twilson@digium.com>

	* tests/test_netsock2.c: Fix netsock2 multiple zero-expansion test
	  Remove erroneous single bracket.

2011-08-12 16:20 +0000 [r331649]  Kinsey Moore <kmoore@digium.com>

	* main/logger.c: Logger does not warn of failure to open logging
	  channels Currently, logger only prints an error message to stderr
	  when it fails to open a logger channel where many users will not
	  see it because the logger lock is held. The alternative provided
	  by this patch is to log the error to all attached consoles in the
	  hopes that it will be easier to see. Additionally, this patch
	  prevents the failed logger channel from being added to the list
	  where it would silently fail on each call to the Asterisk logger.
	  (closes issue ASTERISK-16231) Review:
	  https://reviewboard.asterisk.org/r/1338

2011-08-12 15:49 +0000 [r331635]  Jonathan Rose <jrose@digium.com>

	* apps/app_dial.c, apps/app_meetme.c: Fixes 32bit compilation
	  warnings brought on by 331634 in app_dial and app_meetme

2011-08-11 21:46 +0000 [r331578]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, apps/app_meetme.c: Use proper values for 64-bit
	  option flags. Also, reusing bits es no bueno, so change the value
	  of a duplicate. (issue ASTERISK-18239)

2011-08-11 21:39 +0000 [r331575]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_shell.c: Segfault in shell_helper in func_shell.c. The
	  return value of popen() was not checked for failure to open.
	  (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael
	  Myles Tested by: rmudgett

2011-08-10 22:23 +0000 [r331517]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: SIP Notify via AMI or CLI leaks SIP PVTs Any
	  SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2.
	  Removing the additional ref just before the invite and adding an
	  unref following it corrects the issue as seen via REF_DEBUG. The
	  unref existed in a distant revision and it appears as though the
	  wrong ref operation was removed. (closes issue ASTERISK-18091)
	  Review: https://reviewboard.asterisk.org/r/1332/

2011-08-10 20:29 +0000 [r331461]  Richard Mudgett <rmudgett@digium.com>

	* main/logger.c: Output of queue log not started until logger
	  reloaded. ASTERISK-15863 caused a regression with queue logging.
	  The output of the queue log is not started until the logger
	  configuration is reloaded. * Queue log initialization is
	  completely delayed until the first message is posted to the queue
	  log system. Including the initial opening of the queue log file.
	  * Fixed rotate_file() ROTATE strategy to give the file just
	  rotated out to the configured exec function after rotate. Just
	  like the other strategies. * Fixed logger reload to always post
	  the queue reload entry instead of just if there is a queue log
	  file. * Refactored some code to eliminate some redundancy and to
	  reduce stack utilization. (closes issue ASTERISK-17036) JIRA
	  SWP-2952 Reported by: Juan Carlos Valero Patches:
	  jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by
	  rmudgett Tested by: rmudgett (closes issue ASTERISK-18208)
	  Reported by: Christian Pinedo Review:
	  https://reviewboard.asterisk.org/r/1333/

2011-08-31  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.6.0 Released.

2011-08-25  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.6.0-rc3 Released.

	------------------------------------------------------------------------
	r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines

	Fix installation into directories containing spaces.

	This also vastly simplifies the logic in sounds/Makefile

	(Closes issue ASTERISK-18290)
	Reported by: Paul Belanger
	Review: https://reviewboard.asterisk.org/r/1379/
	------------------------------------------------------------------------

2011-08-22  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.6.0-rc2 Released.

	------------------------------------------------------------------------
	r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines

	Segfault in shell_helper in func_shell.c.

	The return value of popen() was not checked for failure to open.

	(closes issue ASTERISK-18109)
	JIRA SWP-3633
	Reported by: Michael Myles
	Tested by: rmudgett
	------------------------------------------------------------------------
	r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 13 lines

	Re-add support for spaces in pathnames, including now spaces in DESTDIR.

	This was initially added to 1.8 prior to release, primarily to support the
	standard paths on Mac OS X, but was partially reverted recently in Subversion,
	due to the lack of support for spaces in DESTDIR.  This commit restores support
	for the standard paths on Mac OS X, and also includes support for spaces in
	DESTDIR.

	(closes issue ASTERISK-18290)
	Reported by: pabelanger

	Review: https://reviewboard.asterisk.org/r/1326/
	------------------------------------------------------------------------
	r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 7 lines

	Fix possible error on stringification of IPv4-mapped addrs

	The FreeBSD netsock2 test has been failing for a while. We were
	pasing sa->len to getnameinfo instead of sa_tmp->len.

	ASTERISK-18289
	------------------------------------------------------------------------

2011-08-10  Asterisk Development Team <asteriskteam@digium.com>

	* Asterisk 1.8.6.0-rc1 Released.

2011-08-10 13:47 +0000 [r331315]  Kinsey Moore <kmoore@digium.com>

	* main/manager.c: AMI action ModuleReload returns Error if Module:
	  missing or empty An empty string was not being checked for
	  properly causing identification of the module to be reloaded to
	  fail and return an Error with message "No such module." (closes
	  issue AST-616)

2011-08-09 22:12 +0000 [r331248]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, apps/app_parkandannounce.c, main/pbx.c,
	  channels/chan_sip.c, main/features.c: Misc minor items found in
	  code. * Add some reentrancy protection in pbx.c when creating the
	  contexts_table hash table. * Fix inverted test in chan_sip.c
	  conditional code. * Fix uninitialized variable and use of the
	  wrong variable in chan_iax2.c. * Fix test of return value in
	  app_parkandannounce.c. Explicitly testing for -1 is bad if the
	  function does not actually return that value when it fails. *
	  Fixup some comments and add some curly braces in features.c.

2011-08-09 16:13 +0000 [r331146]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooGkClient.c,
	  addons/chan_ooh323.c: move ast_cond_signal for admitted call
	  after all data filled/freed clear all log channels by pointed
	  number not only first free allocated callToken in ooh323_answer

2011-08-09 15:58 +0000 [r331142]  Jason Parker <jparker@digium.com>

	* doc/asterisk.8: Regenerate asterisk man page from sgml.

2011-08-08 20:52 +0000 [r331038]  Kinsey Moore <kmoore@digium.com>

	* res/res_musiconhold.c: In-queue MOH stops after a periodic
	  announcement If the seek value is past the end of file when
	  resuming G.722 MOH, MOH will cease to function for the duration
	  of the MOH session through all starts and stops until saved state
	  is cleared. Adjusting the code to guarantee a single valid read
	  (which is already assumed) fixes the bug. (closes issue
	  ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
	  Tested-by: Jonathan Rose <jrose@digium.com>

2011-08-04 20:29 +0000 [r330843]  Terry Wilson <twilson@digium.com>

	* configure, configure.ac: Make libsrtp instructions more explicit
	  when linking fails (closes issue ASTERISK-18139)

2011-08-04 19:37 +0000 [r330827]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooCmdChannel.c,
	  addons/ooh323c/src/ooGkClient.c: change gk client behaivour on
	  rrq/grq failures to setup timers and next tries after timeout
	  instead of complete failure in the ooh323 stack

2011-08-03 15:14 +0000 [r330705-330762]  Kinsey Moore <kmoore@digium.com>

	* main/Makefile: editing files in main/editline does not ensure
	  rebuild of libedit.a When editing a source file in main/editline,
	  the build system does not rebuild libedit.a and uses the already
	  existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this
	  problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes

	* channels/chan_dahdi.c, channels/sig_analog.c: Call pickup broken
	  for DAHDI channels when beginning with # The call pickup feature
	  did not work on DAHDI devices for anything other than feature
	  codes beginning with * since all feature codes in chan_dahdi were
	  originally hard-coded to begin with *. This patch is also applied
	  to chan_dahdi.c to fix this bug with radio modes. (closes issue
	  AST-621) Review: https://reviewboard.asterisk.org/r/1336/

2011-08-02 20:51 +0000 [r330648]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_jabber.c: Convert an error message to actually be
	  helpful.

2011-08-02 16:15 +0000 [r330575-330581]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes crash in chan_iax2. Fixes crash in
	  chan_iax2 resulting from an edge case in the way control frames
	  are queued during calltoken negotiation is complete. (closes
	  issue ASTERISK-17610) Reported by: mgrobecker

	* channels/chan_sip.c: Optimization to buffer initialization fix.

	* channels/chan_sip.c: Fixes uninitialized string buffer in log
	  message. (closes issue ASTERISK-17200) Reported by: lmadsen

2011-08-01 15:22 +0000 [r330433]  Kinsey Moore <kmoore@digium.com>

	* main/say.c: Incorrect playback for Spanish in some circumstances
	  When you say the time in spanish and it is 01:00 - 01:59 or 13:00
	  - 13:59 you must use female pronunciation "1F". The function
	  "say_date_with_format_es" does not take this in account. (closes
	  ASTERISK-15016) Patch-by: Luis Jimenez

2011-07-30 23:56 +0000 [r330368]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Remove some redundant locking code in
	  ast_do_masquerade(). Also updated some comments.

2011-07-30 15:25 +0000 [r330311]  Gregory Nietsky <gregory@distrotech.co.za>

	* main/channel.c: prevent double masqurading channels when one is
	  been hung up and deadlock avoidance is used. There is a race
	  condition in ast_do_masquerade / ast_hangup (at least) Reported
	  by me signed off by schmidts with input from David Vossel Review:
	  https://reviewboard.asterisk.org/r/1323/

2011-07-29 17:18 +0000 [r330203-330213]  Sean Bright <sean@malleable.com>

	* formats/format_wav.c: Correct the check for O_RDONLY.

	* formats/format_wav.c: Only write to wav files that were opened to
	  be written to.

2011-07-28 21:42 +0000 [r330107]  Terry Wilson <twilson@digium.com>

	* main/term.c: Make console colors work for TERM=xterm-256color

2011-07-28 17:04 +0000 [r330050]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Merged revisions 330033 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
	  28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
	  outgoing call legs of a data call are using different formats:
	  a-law, u-law. When the call is bridged, the media stream is run
	  through translation to convert the media formats. The translation
	  is bad for data calls. * Make incoming call that does not
	  explicitly specify u-law or a-law use the DAHDI channel's default
	  law. The outgoing call always uses the default law from the DAHDI
	  channel. (closes issue ABE-2800) Patches:
	  jira_abe_2800_companding.patch (license #5621) patch uploaded by
	  rmudgett ..........

2011-07-28 15:45 +0000 [r329994]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Fix a SIP transfer deadlock. The locking in
	  this function is very scary. There are like 6 structs involved.
	  (closes issue AST-470)

2011-07-28 15:26 +0000 [r329991]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: check for CONFIG_STATUS_FILE_INVALID when loading
	  the res_fax config file Patch by: tzafrir Reported by: tzafrir
	  (closes issue ASTERISK-18161)

2011-07-28 11:34 +0000 [r329895]  Sean Bright <sean@malleable.com>

	* channels/chan_sip.c: Make the output of Externhost in 'sip show
	  settings' more consistent.

2011-07-27 19:27 +0000 [r329782]  Leif Madsen <lmadsen@digium.com>

	* apps/app_confbridge.c: Change support for ConfBridge() in 1.8 to
	  Extended.

2011-07-27 19:17 +0000 [r329767]  Sean Bright <sean@malleable.com>

	* Makefile.moddir_rules: Explicitly sort the module list so that
	  the menuselect lists are sorted. (closes issue ASTERISK-18141)
	  Reported by: Richard Miller Patches: sort-order.diff uploaded by
	  seanbright (License #5060) Tested by: leifmadsen

2011-07-27 18:10 +0000 [r329709]  Jonathan Rose <jrose@digium.com>

	* configs/indications.conf.sample: Fix New Zealand indications
	  profile based on http://www.telepermit.co.nz/TNA102.pdf (closes
	  issue ASTERISK-16263) Reported by: richardf Patches:
	  nz-indications.patch uploaded by richardf (License #6015)

2011-07-27 04:23 +0000 [r329613]  Tilghman Lesher <tilghman@meg.abyt.es>

	* cdr/cdr_odbc.c: Duration and billsec are swapped in high
	  resolution time. Closes ASTERISK-18024 Patches:
	  20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)

2011-07-26 14:04 +0000 [r329527-329529]  Jonathan Rose <jrose@digium.com>

	* apps/app_voicemail.c: Changes sound file for prepend
	  "then-press-pound" to "vm-then-pound" which is the same prompt,
	  only it turned out "then-press-pound" was part of extra sounds.
	  Also, vm is more appropriate anyway.

	* main/app.c, apps/app_voicemail.c, include/asterisk/app.h,
	  configs/voicemail.conf.sample: Fixes some voicemail forwarding
	  behavior based around prepend mode. Formerly, prepend forwarding
	  would have the user record a message with no useful prompt and an
	  expectation for the user to push a button on the phone when
	  finished recording. If a length of silence was detected instead,
	  the recording would be canceled and the user would re-enter the
	  voicemail forwarding menu. Subsequent time-outs in prepend
	  recording would also bug out in the sense that they would write
	  over the original message and get sent to the recipient
	  regardless of whether they timed out or were accepted. This patch
	  fixes this issue and adds a prompt which will be played after a
	  timeout informing the user that they needed to press a button.
	  Currently, the sound files that we have are somewhat inadquate
	  for this, so after the call we simply have Allison say "Please
	  try again. Then press pound." which actually relies on two
	  separate sound files. Just one would be more appropriate.
	  reporter: Vlad Povorozniuc Review:
	  https://reviewboard.asterisk.org/r/1327/

2011-07-25 19:49 +0000 [r329471]  Paul Belanger <pabelanger@digium.com>

	* main/enum.c: Decrease verbose messages to debug, to help clean up
	  CLI.

2011-07-22 21:10 +0000 [r329144-329333]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c: Fix memory leak in an allocation error path of
	  handle_statechange(). * Make use buffer accessor function in
	  handle_statechange() rather than directly accessing the struct
	  member. * Make use less redundant loop construct for iterating
	  over hints.

	* main/pbx.c: Deadlocks dealing with dialplan hints during reload.
	  There are two remaining different deadlocks reported dealing with
	  dialplan hints. The deadlock in ASTERISK-17666 is caused by
	  invalid locking order in ast_remove_hint(). The hints container
	  must be locked before the hint object. The deadlock in
	  ASTERISK-17760 is caused by a catch-22 situation in
	  handle_statechange(). The deadlock is caused by not having the
	  conlock before calling the watcher callbacks. Unfortunately,
	  having that lock causes a different deadlock as reported in
	  ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
	  handle_statechange() no longer call the watcher callbacks holding
	  any locks that matter. * Made hint ao2 destructor do the watcher
	  callbacks for extension deactivation to guarantee that they get
	  called. * Fixed hint reference leak in ast_add_hint() if the
	  callback container constructor failed. * Fixed hint reference
	  leak in complete_core_show_hint() for every hint it found for CLI
	  tab completion. * Adjusted locking in
	  ast_merge_contexts_and_delete() for safety. * Added
	  context_merge_lock to prevent ast_merge_contexts_and_delete() and
	  handle_statechange() from interfering with each other. * Fixed
	  ast_change_hint() not taking into account that the extension is
	  used for the hash key. (closes issue ASTERISK-17666) Reported by:
	  irroot Tested by: irroot JIRA SWP-3318 (closes issue
	  ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
	  SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Document
	  parkinglot in chan_dahdi.conf.sample. * Document existing feature
	  in chan_dahdi.conf.sample. * Remove some dead code related to the
	  parkinglot option.

	* apps/app_directed_pickup.c: Update PickupChan documentation. The
	  PickupChan uses the ampersand as the argument separator. Was
	  documented as: PickupChan(channel[,channel2[,...][,options]])
	  Fixed documentation to:
	  PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
	  This is a continuation of ASTERISK-17494 for v1.8 and later.
	  (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
	  pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
	  by Erik Smith Tested by: Erik Smith

	* main/features.c: Dialplan bridge() app mutex 'current_dest_chan'
	  freed more times than we've locked! This appears to be a leftover
	  from when ast_channel was converted to ao2 objects. Simply
	  removed the extraneous unlock. (closes issue ASTERISK-17772)

2011-07-20 21:20 +0000 [r329027]  Paul Belanger <pabelanger@digium.com>

	* UPGRADE.txt: Asterisk now requires libpri 1.4.11+ for PRI
	  support.

2011-07-20 20:52 +0000 [r329012]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  Backport useful CLI "pri show channels" command to v1.8. The "pri
	  show channels" command is useful for debuging to see if there are
	  any stuck B channels. .......... r307964 | rmudgett | 2011-02-15
	  15:42:55 -0600 (Tue, 15 Feb 2011) | 9 lines Add CLI "pri show
	  channels" command. List the current mapping of DAHDI B channels
	  to Asterisk channel names and which calls are on hold or
	  call-waiting. Calls on hold or call-waiting are not associated
	  with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 ..........
	  r308205 | rmudgett | 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011)
	  | 1 line Add more verbage to CLI command 'pri show channels'
	  usage. .......... r312579 | rmudgett | 2011-04-04 11:17:58 -0500
	  (Mon, 04 Apr 2011) | 59 lines Change also updates 'pri show
	  channels' command with the "chan idle" column to report if a
	  channel is available for use.

2011-07-20 20:16 +0000 [r328987]  Terry Wilson <twilson@digium.com>

	* tests/test_netsock2.c: We can't guarantee an eth0 is present
	  FreeBSD test fails on this case presumably because there is no
	  eth0 on the test machine. Better to just remove this test for
	  now.

2011-07-20 19:00 +0000 [r328935]  Kinsey Moore <kmoore@digium.com>

	* channels/chan_sip.c: Inband DTMF regression The functionality of
	  inband DTMF in chan_sip relied upon
	  ast_rtp_instance_dtmf_mode_get/set not working properly to avoid
	  calling ast_rtp_instance_dtmf_begin/end on RTP streams with
	  inband DTMF. According to documentation,
	  ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
	  never inband. This fixes the regression introduced in revision
	  328823.

2011-07-19 21:29 +0000 [r328878]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile, Makefile, Makefile.moddir_rules: Revert partial
	  attempt at handling pathnames with spaces. Revision 299794
	  attempted to improve the build system to be able to handle
	  pathnames (primarily DESTDIR) with spaces in them, since this is
	  common on some platforms (including Mac OSX). Unfortunately, the
	  changes were incomplete and did not actually provide the desired
	  behavior, and as a side effect the functionality that ensured
	  that stale headers in the Asterisk 'include' directory were
	  removed got broken. In addition, the check for stale (and
	  possibly incompatible) modules in the Asterisk 'modules'
	  directory also got broken, and would never report any stale
	  modules. Users upgrading to this version or later versions would
	  then see unexpected module load errors. Since there are few users
	  who actually want to install Asterisk into paths that contain
	  spaces, and a proper fix for the build system would take many
	  hours, the best solution for now is to just revert the partial
	  solution.

2011-07-19 17:57 +0000 [r328770-328823]  Kinsey Moore <kmoore@digium.com>

	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
	  main/rtp_engine.c, channels/chan_sip.c: RTP bridge away with
	  inband DTMF and feature detection When deciding whether Asterisk
	  was allowed to bridge the call away from the core, chan_sip did
	  not take into account the usage of features on dialed channels
	  that require monitoring of DTMF on channels utilizing inband
	  DTMF. This would cause Asterisk to allow the call to be locally
	  or remotely bridged, preventing access to the data required to
	  detect activations of such features. (closes 17237) Review:
	  https://reviewboard.asterisk.org/r/1302/

	* apps/app_meetme.c: MeetMe requests a PIN twice in some
	  circumstances If a call to MeetMe includes both the dynamic(D)
	  and always request PIN(P) options, MeetMe will ask for the PIN
	  two times: once for creating the conference and once for entering
	  the conference. This behavior was introduced in rev 311616 when
	  adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
	  controlling PIN entry for joining a conference. (closes AST-601)
	  Review: https://reviewboard.asterisk.org/r/1305/

2011-07-19 01:35 +0000 [r328716]  Terry Wilson <twilson@digium.com>

	* tests/test_linkedlists.c (added), include/asterisk/linkedlists.h:
	  Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the
	  element passed in if it isn't found. This commit also adds linked
	  list unit tests. Review: https://reviewboard.asterisk.org/r/1321/

2011-07-18 20:47 +0000 [r328593-328663]  Mark Murawki <markm@intellasoft.net>

	* apps/app_dial.c: app_dial may double free a channel datastore
	  When starting a call with originate, and having the callee
	  channel run Bridge() on pickup, we will double free the
	  dialed_interface_info datastore, causing a crash. Make sure to
	  check if the datastore still exists before trying to free it.
	  (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested
	  by: Mark Murawski

	* channels/chan_sip.c: If the sip private structure is null,
	  sip_setoption() will defref the null pointer and crash. Ideally,
	  sip_setoption shouldn't be called if there is a lack of a sip
	  private structure. But this will fix a crash. (closes issue
	  ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
	  Murawski

	* main/asterisk.c: Fixed invalid read and null pointer deref on
	  asterisk shutdown. In some cases when starting asterisk with -c
	  and hitting control-c to shutdown, there will be an invalid read
	  and null pointer deref causing a crash. (closes issue
	  ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark
	  Murawski, Kinsey Moore, Tilghman Lesher

2011-07-18 07:10 +0000 [r328540]  Tilghman Lesher <tilghman@meg.abyt.es>

	* funcs/func_odbc.c: Typo

2011-07-15 20:41 +0000 [r328446]  Leif Madsen <lmadsen@digium.com>

	* apps/app_macro.c, channels/chan_jingle.c, apps/app_dahdibarge.c,
	  apps/app_readfile.c, apps/app_setcallerid.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, cdr/cdr_sqlite.c,
	  channels/chan_h323.c: Revert changes to defaultenabled state for
	  modules in Asterisk 1.8

2011-07-15 19:22 +0000 [r328427]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c: small gk processing fixes: -
	  decrease for 1 second registration ttl for very low expirations
	  (some providers expire few earlier than TTL) - delete rrq and
	  registration expire timers on URQ received as we make new
	  registration.

2011-07-14 23:12 +0000 [r328302]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Missing SIP pvt and channel unlock in
	  sip_set_rtp_peer(). Regression introduced by -r326144. Add
	  missing SIP pvt and channel unlock in sip_set_rtp_peer().

2011-07-14 20:13 +0000 [r328209]  Leif Madsen <lmadsen@digium.com>

	* apps/app_image.c, res/res_http_post.c, formats/format_wav_gsm.c,
	  utils/stereorize.c, pbx/pbx_loopback.c, funcs/func_shell.c,
	  main/features.c, channels/chan_alsa.c, apps/app_externalivr.c,
	  formats/format_jpeg.c, res/res_speech.c, formats/format_gsm.c,
	  apps/app_milliwatt.c, formats/format_g719.c,
	  apps/app_saycounted.c, apps/app_fax.c, apps/app_echo.c,
	  funcs/func_math.c, channels/chan_agent.c, apps/app_dahdiras.c,
	  utils/astman.c, res/res_ael_share.c, apps/app_transfer.c,
	  apps/app_playback.c, res/res_config_curl.c, funcs/func_curl.c,
	  apps/app_waitforring.c, channels/chan_misdn.c, tests/test_skel.c,
	  addons/cdr_mysql.c, codecs/codec_ilbc.c, apps/app_zapateller.c,
	  apps/app_chanspy.c, apps/app_cdr.c, tests/test_substitution.c,
	  funcs/func_md5.c, utils/muted.c, tests/test_gosub.c,
	  funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
	  cdr/cdr_radius.c, formats/format_siren7.c,
	  apps/app_controlplayback.c, funcs/func_config.c, main/manager.c,
	  bridges/bridge_builtin_features.c, funcs/func_volume.c,
	  cdr/cdr_sqlite.c, funcs/func_aes.c, funcs/func_frame_trace.c,
	  tests/test_devicestate.c, res/res_agi.c, tests/test_astobj2.c,
	  apps/app_confbridge.c, apps/app_ivrdemo.c,
	  res/res_clioriginate.c, res/res_calendar_icalendar.c,
	  funcs/func_dialplan.c, funcs/func_db.c,
	  tests/test_ast_format_str_reduce.c, res/res_fax.c,
	  res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
	  apps/app_waituntil.c, channels/chan_console.c,
	  apps/app_getcpeid.c, apps/app_queue.c, funcs/func_global.c,
	  funcs/func_extstate.c, channels/chan_usbradio.c,
	  apps/app_flash.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
	  formats/format_g729.c, funcs/func_dialgroup.c, funcs/func_env.c,
	  res/res_timing_dahdi.c, funcs/func_strings.c,
	  res/res_calendar_caldav.c, apps/app_chanisavail.c,
	  formats/format_sln16.c, apps/app_ices.c, apps/app_exec.c,
	  bridges/bridge_multiplexed.c, cel/cel_odbc.c,
	  formats/format_pcm.c, pbx/pbx_ael.c, formats/format_h263.c,
	  cdr/cdr_manager.c, res/res_clialiases.c, funcs/func_sprintf.c,
	  tests/test_app.c, apps/app_softhangup.c, codecs/codec_g726.c,
	  apps/app_morsecode.c, utils/smsq.c, bridges/bridge_simple.c,
	  tests/test_sched.c, apps/app_talkdetect.c, apps/app_db.c,
	  res/res_calendar_ews.c, funcs/func_callcompletion.c,
	  tests/test_acl.c, funcs/func_cdr.c, utils/ael_main.c,
	  utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c,
	  channels/chan_vpb.cc, res/res_snmp.c, apps/app_dictate.c,
	  apps/app_authenticate.c, res/res_phoneprov.c, funcs/func_logic.c,
	  res/res_jabber.c, funcs/func_uri.c,
	  funcs/func_audiohookinherit.c, res/res_config_odbc.c,
	  funcs/func_odbc.c, res/res_realtime.c, codecs/codec_resample.c,
	  formats/format_h264.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  tests/test_amihooks.c, codecs/codec_lpc10.c, channels/chan_sip.c,
	  cdr/cdr_syslog.c, funcs/func_lock.c, res/res_adsi.c,
	  utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
	  formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
	  formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
	  addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c,
	  funcs/func_module.c, apps/app_sayunixtime.c,
	  cdr/cdr_adaptive_odbc.c, res/res_smdi.c, tests/test_time.c,
	  apps/app_skel.c, funcs/func_srv.c, apps/app_amd.c,
	  pbx/pbx_realtime.c, apps/app_url.c, apps/app_dial.c,
	  apps/app_page.c, channels/chan_bridge.c, apps/app_privacy.c,
	  codecs/codec_speex.c, apps/app_disa.c, res/res_mutestream.c,
	  res/res_monitor.c, apps/app_macro.c, res/res_timing_kqueue.c,
	  res/res_fax_spandsp.c, channels/chan_unistim.c,
	  funcs/func_base64.c, addons/app_mysql.c,
	  channels/chan_multicast_rtp.c, apps/app_meetme.c,
	  utils/hashtest.c, res/res_musiconhold.c, apps/app_followme.c,
	  res/res_config_sqlite.c, cdr/cdr_csv.c,
	  tests/test_security_events.c, formats/format_ilbc.c,
	  funcs/func_enum.c, channels/chan_phone.c,
	  tests/test_stringfields.c, funcs/func_groupcount.c,
	  tests/test_locale.c, addons/chan_mobile.c, cdr/cdr_custom.c,
	  res/res_security_log.c, apps/app_parkandannounce.c,
	  apps/app_while.c, apps/app_jack.c, res/res_rtp_asterisk.c,
	  apps/app_nbscat.c, codecs/codec_a_mu.c, tests/test_dlinklists.c,
	  res/res_convert.c, pbx/pbx_lua.c, utils/astcanary.c,
	  channels/chan_oss.c, tests/test_strings.c, res/res_srtp.c,
	  cdr/cdr_tds.c, res/res_timing_pthread.c,
	  apps/app_directed_pickup.c, channels/chan_h323.c,
	  cel/cel_sqlite3_custom.c, apps/app_senddtmf.c,
	  funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c,
	  funcs/func_speex.c, apps/app_dahdibarge.c, channels/chan_local.c,
	  tests/test_logger.c, apps/app_record.c, apps/app_playtones.c,
	  bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
	  channels/chan_iax2.c, res/res_pktccops.c,
	  res/res_rtp_multicast.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
	  funcs/func_pitchshift.c, channels/chan_skinny.c,
	  apps/app_dumpchan.c, main/http.c, cdr/cdr_odbc.c,
	  utils/refcounter.c, res/res_calendar_exchange.c, res/res_ais.c,
	  codecs/codec_g722.c, tests/test_expr.c, funcs/func_timeout.c,
	  cel/cel_tds.c, formats/format_wav.c, formats/format_ogg_vorbis.c,
	  funcs/func_cut.c, apps/app_speech_utils.c, apps/app_sendtext.c,
	  funcs/func_channel.c, utils/hashtest2.c, pbx/pbx_config.c,
	  funcs/func_iconv.c, apps/app_mixmonitor.c, formats/format_g726.c,
	  res/res_odbc.c, apps/app_voicemail.c, tests/test_heap.c,
	  addons/format_mp3.c, formats/format_sln.c, apps/app_readexten.c,
	  apps/app_userevent.c, codecs/codec_gsm.c, channels/chan_gtalk.c,
	  cdr/cdr_pgsql.c, tests/test_func_file.c, apps/app_setcallerid.c,
	  apps/app_osplookup.c, cel/cel_manager.c, cel/cel_custom.c,
	  tests/test_utils.c, apps/app_minivm.c, apps/app_mp3.c,
	  res/res_timing_timerfd.c, apps/app_directory.c,
	  res/res_config_ldap.c, formats/format_siren14.c,
	  apps/app_adsiprog.c, res/res_config_pgsql.c, apps/app_read.c,
	  funcs/func_version.c, codecs/codec_alaw.c, agi/eagi-test.c,
	  res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
	  apps/app_forkcdr.c, funcs/func_blacklist.c, pbx/pbx_dundi.c,
	  apps/app_sms.c, apps/app_stack.c, funcs/func_devstate.c,
	  apps/app_verbose.c, addons/res_config_mysql.c,
	  utils/check_expr.c, funcs/func_rand.c, apps/app_readfile.c,
	  codecs/codec_adpcm.c, apps/app_test.c, tests/test_event.c:
	  Introduce <support_level> tags in MODULEINFO. This change
	  introduces MODULEINFO into many modules in Asterisk in order to
	  show the community support level for those modules. This is used
	  by changes committed to menuselect by Russell Bryant recently
	  (r917 in menuselect). More information about the support level
	  types and what they mean is available on the wiki at
	  https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States

2011-07-14 19:21 +0000 [r328205]  Jonathan Rose <jrose@digium.com>

	* res/res_monitor.c: Monitor application arguments requirements
	  fixed. Monitor was requiring options in spite of no individual
	  option on Monitor being required. Review:
	  https://reviewboard.asterisk.org/r/1320/

2011-07-13 18:46 +0000 [r328014]  Richard Mudgett <rmudgett@digium.com>

	* configs/features.conf.sample: Add ATXFER_NULL_TECH note in
	  features.conf.sample.

2011-07-12 22:53 +0000 [r327950]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: Correct double-free situation in manager output
	  processing. The process_output() function calls ast_str_append()
	  and xml_translate() on its 'out' parameter, which is a pointer to
	  an ast_str buffer. If either of these functions need to
	  reallocate the ast_str so it will have more space, they will free
	  the existing buffer and allocate a new one, returning the address
	  of the new one. However, because process_output only receives a
	  pointer to the ast_str, not a pointer to its caller's variable
	  holding the pointer, if the original ast_str is freed, the caller
	  will not know, and will continue to use it (and later attempt to
	  free it). (reported by jkroon on #asterisk-dev)

2011-07-12 20:07 +0000 [r327890]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_directory.c: search in the current context for 'a' and
	  'o' instead of 'default'

2011-07-12 19:38 +0000 [r327888]  Jason Parker <jparker@digium.com>

	* Makefile: Fix uninstall target, so that modules dir gets cleared
	  again.

2011-07-12 19:10 +0000 [r327852]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Added additional checks for mailbox /
	  password beginning with '*' character A bug existed such that if
	  a user entered a password with '*', and the extension 'a' did not
	  exist, an invalid mailbox would be created and the user
	  authenticated. The code was changed to prevent this from
	  occurring, and to prevent users from having mailboxes or
	  passwords defined that begin with the '*' character. (closes
	  issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
	  Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/

2011-07-12 15:35 +0000 [r327793]  Tilghman Lesher <tilghman@meg.abyt.es>

	* tests/test_substitution.c: Use 'printf' (POSIX issue 4) instead
	  of 'echo -n', for portability. The problem with using 'echo -n'
	  is that it is not portable. While BSD systems required that the
	  '-n' option be removed and interpreted, System V required that
	  all strings should be echoed with no interpretation of options.
	  This fundamental difference of behavior means that it is never
	  possible to use the '-n' flag to echo in tests which are meant to
	  be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
	  builtin 'echo' uses the System V semantics of the command, and
	  thus the SHELL test failed on that platform.
	  http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16

2011-07-11 19:41 +0000 [r327682]  Terry Wilson <twilson@digium.com>

	* include/asterisk/jingle.h, channels/chan_gtalk.c: Update
	  chan_gtalk to work with changed GMail-based calls The messages
	  sent by the GMail client have changed, but include the old-style
	  messages as well. This patch checks for this case and uses the
	  old-style offer. (closes issue ASTERISK-18084) Review:
	  https://reviewboard.asterisk.org/r/1312/

2011-07-11 13:53 +0000 [r327512]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, tests/test_substitution.c: reset our buffer each
	  iteration when doing variable substitution

2011-07-11 10:56 +0000 [r327411-327412]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/Makefile: Properly building the Debian armhf (HardFloat)
	  port. Remove the line that should have been removed in r327411.

	* main/Makefile: fix building the Debian armhf (HardFloat) port
	  Fixes
	  http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
	  (Missing pthreads)

2011-07-08 22:27 +0000 [r327258]  Jason Parker <jparker@digium.com>

	* main/db1-ast/mpool, addons, cdr, formats, codecs/gsm/src, funcs,
	  addons/ooh323c/src, bridges, codecs/lpc10, main/db1-ast/btree,
	  codecs/g722, main, main/db1-ast/recno, channels/sip, res, pbx,
	  res/ael, channels, main/stdtime, addons/ooh323c/src/h323, codecs,
	  utils, main/db1-ast/hash, cel, apps, main/db1-ast/db: Add .o
	  files to svn:ignore property, since it's only ignored if locally
	  configured to do so.

2011-07-08 21:41 +0000 [r327211]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: INVITE 403 Forbidden response always
	  retransmits the maximum times. Asterisk sends a 403 Forbidden
	  response if authentication fails for an INVITE as required.
	  However, it ignores the ACK and keeps retransmitting the
	  response. * Made not delete the to-tag in the dialog so the
	  expected ACK can be matched with the dialog and stop the
	  retransmissions.

2011-07-08 19:52 +0000 [r327106]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, tests/test_substitution.c: Reset our ast_str before
	  passing it on to dialplan function backends. It is possible for a
	  dialplan backend to not modify the given buffer or ast_str and
	  still return success. This causes any previous value stored in
	  the buffer to be used as if the new function call provided it.
	  Some functions also append to the given buffer assuming it is
	  empty. The test_substitution unit test has also been modified to
	  detect this problem. (closes issue ASTERISK-17878)

2011-07-08 16:00 +0000 [r327044-327046]  Russell Bryant <russell@digium.com>

	* tests/test_netsock2.c: Fix an error and add more log message info
	  to help see why this fails on FreeBSD.

	* channels/chan_dahdi.c: Resolve some set-but-unused-variable
	  warnings.

2011-07-08 01:08 +0000 [r326985]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c: Some code cleanup in pbx.c * Mostly comment and
	  format changes. * ast_context_remove_extension_callerid() and
	  ast_add_extension_nolock() will write lock the found specific
	  context. * ast_context_find() will now tolerate a NULL name. *
	  Eliminated some inlined versions of find_context() and
	  find_context_locked().

2011-07-07 19:17 +0000 [r326830]  Tilghman Lesher <tilghman@meg.abyt.es>

	* res/res_http_post.c: libgen.h is also needed on Darwin for
	  basename(3)

2011-07-07 16:04 +0000 [r326689]  Jonathan Rose <jrose@digium.com>

	* res/res_config_odbc.c: res_odbc patch by tilghman to fix integers
	  with null values Addresses some improper sql statements in
	  res_odbc that would cause an update to fail on realtime peers due
	  to trying to set as "(NULL)" rather than an actual NULL. (closes
	  issue #1922STERISK-17791) Reported by: marcelloceschia Patches:
	  20110505__issue19223.diff.txt uploaded by tilghman (license 14)

2011-07-07 15:28 +0000 [r326681-326683]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: use sips: or sip: depending on the transport
	  in use when building reply digest URIs

	* channels/chan_sip.c: make the uri parameter used in reply digests
	  more standards compliant in certain cases by prepending "sip:" or
	  "sips:" to it

2011-07-06 15:26 +0000 [r326484]  David Vossel <dvossel@digium.com>

	* res/res_timing_timerfd.c: Reverts fix for timerfd locking issue.
	  jrose discovered a performance issue with this fix that prevents
	  his analog phones from working when using timerfd as a timing
	  source. Until it is understood what is causing this performance
	  problem, this patch is being reverted.

2011-07-06 14:35 +0000 [r326411-326469]  Tilghman Lesher <tilghman@meg.abyt.es>

	* pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
	  channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c,
	  channels/chan_jingle.c, channels/chan_dahdi.c,
	  funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
	  funcs/func_aes.c: Removing type attributes, as a change to
	  menuselect makes them no longer necessary.

	* pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
	  channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c,
	  channels/chan_jingle.c, channels/chan_dahdi.c,
	  funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
	  funcs/func_aes.c: Add the attribute "type" to each "<use>" for
	  menuselect. This matters only when autoconf fails to detect that
	  weak linking is supported. External optional dependencies will
	  become optional in both cases, as they are removed at compile
	  time when not detected. However, runtime-optional modules are
	  made mandatory when weak linking is not found. This change
	  affects only the external optional dependencies; previously, they
	  were incorrectly required when weak linking support was not
	  detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt
	  by tilghman (License #5003) Tested by: iasgoscouk

2011-07-05 17:22 +0000 [r326291]  Richard Mudgett <rmudgett@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: Used auth=
	  parameter freed during "sip reload" causes crash. If you use the
	  auth= parameter and do a "sip reload" while there is an ongoing
	  call. The peer->auth data points to free'd memory. The patch does
	  several things: 1) Puts the authentication list into an ao2
	  object for reference counting to fix the reported crash during a
	  SIP reload. 2) Converts the authentication list from open coding
	  to AST list macros. 3) Adds display of the global authentication
	  list in "sip show settings". (closes issue ASTERISK-17939)
	  Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch
	  (license #5621) patch uploaded by rmudgett Review:
	  https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526

2011-07-05 13:23 +0000 [r326209]  Matthew Jordan <mjordan@digium.com>

	* main/file.c: Updated filestream destructor to block until move is
	  complete when cache is used When a cache directory is used, the
	  process is forked and a mv command is executed to move the
	  temporary file to the permanent location. This caused issues with
	  voicemail, where a race condition occurred when the parent
	  expected the file to be in the permanent location prior to the mv
	  command completing. The parent process is now blocked until the
	  mv command completes. (closes issue ASTERISK-17724) Reported by:
	  Adiren P. Tested by: mjordan

2011-07-01 21:07 +0000 [r326144]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Better way to get chan and pvt lock for
	  issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431
	  deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). *
	  Lock the channels in the defined order and avoid the need for a
	  deadlock avoidance loop. * Lock the channel before getting the
	  pointer to the private structure to be sure that the pointer will
	  not change due to a masquerade or channel hangup. * To preserve
	  sanity, check that chan and p->owner are the same. (Pointer
	  rearangements should not happen without the protection of locks
	  because bad things tend to happen otherwise.)

2011-06-30 20:39 +0000 [r325935]  Richard Mudgett <rmudgett@digium.com>

	* configs/sip.conf.sample, channels/chan_sip.c: Misc minor changes
	  in chan_sip. * Add load failure exit if primary SIP container(s)
	  could not get created in chan_sip.c:load_module(). * Removed a
	  redundant static prototype. * Some typos. * Some whitespace.

2011-06-30 20:09 +0000 [r325877]  Matthew Jordan <mjordan@digium.com>

	* apps/app_voicemail.c: Patched voicemail user option for emailbody
	  / emailsubject Incorporated changes per ASTERISK-16795; updated
	  unit tests to check for vmu->emailbody / vmu->emailsubject
	  (closes issue ASTERISK-16795) Reported by: mdeneen Tested by:
	  mjordan

2011-06-30 19:17 +0000 [r325821]  Jonathan Rose <jrose@digium.com>

	* res/res_musiconhold.c: Fixes an issue with Music on Hold classes
	  losing files in playlist when realtime is used. The bug occurs
	  rather intermittently and I relied on the reporters to test the
	  patch. After a sanity check and some testing, I'm giving it an
	  OK. (closes issue ASTERISK-17875) Reported by: David Cunningham
	  Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor
	  Goncharovsky (license #5009)

2011-06-29 21:49 +0000 [r325740]  Kinsey Moore <kmoore@digium.com>

	* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
	  cleanup from the introduction of ast_str Remove the length field
	  from sip_req and sip_pkt in chan_sip since they are redundant
	  (ast_str holds its own length) and refactor the necessary
	  functions. Review: https://reviewboard.asterisk.org/r/1281/

2011-07-11  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.5.0 Released.

	* r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011)

	  Reverts fix for timerfd locking issue.

	  jrose discovered a performance issue with this
	  fix that prevents his analog phones from working
	  when using timerfd as a timing source.  Until
	  it is understood what is causing this performance
	  problem, this patch is being reverted.

2011-06-29  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.5-rc1 Released.

2011-06-29 18:59 +0000 [r325673]  David Vossel <dvossel@digium.com>

	* res/res_timing_timerfd.c: Fixes timerfd locking issue. (closes
	  ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz
	  https://reviewboard.asterisk.org/r/1255/

2011-06-29 18:16 +0000 [r325610-325614]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Fixed some error exit cleanup in app_queue.c. *
	  Fixed error exit cleanup in app_queue.c copy_rules() and
	  reload_queue_rules().

	* apps/app_queue.c: Response to QueueRule manager command does not
	  contain ActionID if it was specified. * Add ActionID support as
	  documented for the QueueRule AMI action. * Remove documentation
	  for ActionID with the Queues AMI action. The output does not
	  follow normal AMI response output and there is no place to put an
	  ActionID header. (closes issue AST-602) Reported by: Vlad
	  Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621)
	  patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett
	  Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575

2011-06-29 16:18 +0000 [r325537-325545]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: make framehooks prevent native bridging (for real
	  this time)

	* apps/app_dial.c, main/rtp_engine.c: don't do native/remote
	  bridging if a framehook is active on the channel

2011-06-28 21:50 +0000 [r325416]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Fix random misspelling noticed on
	  asterisk-users.

2011-06-28 20:31 +0000 [r325339]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes locking inversion caused by holding
	  sip pvt lock during async_goto. (closes ASTERISK-17352)

2011-06-28 20:07 +0000 [r325279]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 325277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r325277 | twilson | 2011-06-28 15:06:16 -0500
	  (Tue, 28 Jun 2011) | 9 lines Merged revisions 325275 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28
	  Jun 2011) | 2 lines Don't leak SIP username information ........
	  ................

2011-06-28 17:30 +0000 [r325212]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Use the device name and not the channel
	  name to initialize the device state. Correct ASTERISK-11323
	  implementation as I don't see how it ever worked as claimed when
	  it used the channel name and not the device name. (issue
	  ASTERISK-11323)

2011-06-28 15:46 +0000 [r325152]  Jonathan Rose <jrose@digium.com>

	* res/res_musiconhold.c: Fixes moh reload breaking custom mode moh
	  classes when the config file is untouched (closes issue
	  ASTERISK-17730) Reported by: sdolloff

2011-06-28 15:12 +0000 [r325091]  Leif Madsen <lmadsen@digium.com>

	* build_tools/prep_tarball: Remove line from prep_tarball that
	  kills mkrelease.

2011-06-27 16:30 +0000 [r324955]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/asterisk.c: Save and restore errno from within signal
	  handlers. This is recommended by the POSIX standard, as well as
	  by the sigaction(2) manpage for various platforms that we support
	  (e.g. Mac OS X).

2011-06-27 15:37 +0000 [r324914]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: When subscribing MWI to an unsolicited
	  mailbox the first notification is incorrect. A remote peer
	  subscribed to MWI with the unsolicited option and a local phone
	  subscribed to the remote mailbox. The notify message-summary
	  events are sent correctly except for the first one when
	  subscribing, which will always be 0. This means the phone MWI
	  indicator will be wrong until the mailbox read/unread count
	  changes and the event is fired. Looks like this is a regression
	  from ASTERISK-16149. * Fix the logic to check the cache and if
	  allowed then fallback to manually counting mailbox messages.
	  (closes issue ASTERISK-17997) Reported by: rsw686 Patches:
	  jira_asterisk_17997_v1.8.patch (license #5621) uploaded by
	  rmudgett Tested by: rsw686 JIRA SWP-3551

2011-06-24 20:46 +0000 [r324849]  Richard Mudgett <rmudgett@digium.com>

	* pbx/pbx_config.c: Syntax errors in dialplan do not display the
	  file name. When issuing the CLI command "dialplan reload" syntax
	  errors and warnings are displayed on the console. The offending
	  line number is displayed on the console, but the file name is not
	  displayed. Errors caught in main/config.c do display the file
	  name. (closes issue ASTERISK-17985) Reported by: ulogic Patches:
	  pbx_config.patch uploaded by ulogic (License #5685) modified
	  format Tested by: rmudgett JIRA SWP-3554

2011-06-24 16:48 +0000 [r324768]  Jonathan Rose <jrose@digium.com>

	* include/asterisk/logger.h: DTMF wasn't being logged on connected
	  consoles when enabled in logger.conf Previously in order for DTMF
	  to be logged in a connected console session, the user would have
	  to do logger set channel DTMF on. This corrects that so that it
	  is on by default. This issue was caused by an off by one error
	  incurred by a logger level count of 6 in logger.h where it should
	  have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H

2011-06-23 18:31 +0000 [r324685]  David Vossel <dvossel@digium.com>

	* channels/sip/reqresp_parser.c: Fixes sip crash when calling
	  remove_uri_parameters with NULL AST-2011-009 (closes issue
	  ASTERISK-18017) Reported by: jaredmauch

2011-06-23 18:29 +0000 [r324678]  Kinsey Moore <kmoore@digium.com>

	* /, channels/chan_sip.c: Merged revisions 324643 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) |
	  4 lines Addresses AST-2011-008, memory corruption and remote
	  crash in SIP driver. AST-2011-008 ........

2011-06-23 18:23 +0000 [r324652]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, include/asterisk/frame.h, /,
	  main/features.c: Merged revisions 324634 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500
	  (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
	  | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
	  Thanks to twilson for identifying the issue and providing the
	  patches. AST-2011-010 ........ ................

2011-06-23 03:10 +0000 [r324557]  Terry Wilson <twilson@digium.com>

	* tests/test_netsock2.c: Remove tests for parsing address with
	  invalid port getaddrinfo on OS X returns with EAI_NONAME error
	  when passed a port greater than 65535. Linux throws no error, so
	  remove the tests for now.

2011-06-22 19:16 +0000 [r324491]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Use correct variable for text SRTP media.

2011-06-22 18:52 +0000 [r324484]  Terry Wilson <twilson@digium.com>

	* include/asterisk/netsock2.h, tests/test_netsock2.c (added),
	  main/netsock2.c, channels/chan_sip.c: Stop sending IPv6
	  link-local scope-ids in SIP messages The idea behind the patch
	  listed below was used, but in a more targeted manner. There are
	  now address stringification functions for addresses that are
	  meant to be sent to a remote party. Link-local scope-ids only
	  make sense on the machine from which they originate and so are
	  stripped in the new functions. There is also a host sanitization
	  function added to chan_sip which is used for when peer and dialog
	  tohost fields or sip_registry hostnames are used to craft a SIP
	  message. Also added are some basic unit tests for netsock2
	  address parsing. (closes issue ASTERISK-17711) Reported by:
	  ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded
	  by ch_djalel (license 1251) Review:
	  https://reviewboard.asterisk.org/r/1278/

2011-06-22 18:41 +0000 [r324479-324481]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Timout or error on INFO or MESSAGE
	  transaction causes call to be lost. When exchanging INFO messages
	  within a call, 4xx error causes the call to be disconnected
	  although RFC 2976 explicitly states that such transactions do not
	  modify the state of the dialog. When exchanging MESSAGE messages
	  within a call, 4xx error causes the call to be disconnected. To
	  provide least surprise, we should not disconnect the call since a
	  MESSAGE is like INFO in this case. (Implied by RFC 3428 Section
	  2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review:
	  https://reviewboard.asterisk.org/r/1257/ Review:
	  https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486

	* channels/chan_sip.c: Comments and whitespace in chan_sip.c

2011-06-21 20:11 +0000 [r324364]  David Vossel <dvossel@digium.com>

	* include/asterisk/pbx.h, main/pbx.c: Fixes locking inversion issue
	  in ast_async_goto() During this function we can not hold the
	  "chan" lock while doing the masquerade, the explicit goto on the
	  tmp chan, or the channel alloc. Instead we need to get the
	  channel lock, store off information about the channel that we
	  need, and then let the channel lock go for the remainder of the
	  function. Review: https://reviewboard.asterisk.org/r/1275/

2011-06-21 16:09 +0000 [r324305]  Kinsey Moore <kmoore@digium.com>

	* apps/app_confbridge.c: ConfBridge does not handle hangup properly
	  When playing back a prompt to a channel, confbridge neglects to
	  check for hangup events causing lockup condititions for hangups
	  that occur before actually joining the conference. This change
	  ensures that the user is removed from the conference in the event
	  of a premature hangup. Review:
	  https://reviewboard.asterisk.org/r/1277/

2011-06-20 18:12 +0000 [r324239-324241]  Leif Madsen <lmadsen@digium.com>

	* configs/queuerules.conf.sample: Remove extra 'the'. Reported by
	  Vlad Povorozniuc

	* configs/queuerules.conf.sample,
	  contrib/scripts/asterisk.logrotate: Revert previous merge which
	  had extra changes.

	* configs/queuerules.conf.sample,
	  contrib/scripts/asterisk.logrotate: Remove extra 'the'. Reported
	  by Vlad Povorozniuc

2011-06-20 17:33 +0000 [r324237]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Ignore media offers with a port of 0 Section
	  5.1 of RFC3264 states: A port number of zero in the offer
	  indicates that the stream is offered but MUST NOT be used.
	  (closes issue ASTERISK-17845) Reported by: jacco Patches:
	  issue19281_2.patch uploaded by jacco (license 1277) Tested by:
	  jacco, twilson

2011-06-17 18:51 +0000 [r324176-324178]  Leif Madsen <lmadsen@digium.com>

	* main/manager.c: Add Username and Secret fields to manager Login
	  action. Pointed out by Vlad Povorozniuc

	* apps/app_meetme.c: Fix typo in documentation. Pointed out by Vlad
	  Povorozniuc

2011-06-17 18:23 +0000 [r324174]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Add header string to libpri debug output.
	  Add header string to libpri debug output so the libpri output can
	  be found/extracted easier from huge debug trace files.

2011-06-17 15:14 +0000 [r324115]  Leif Madsen <lmadsen@digium.com>

	* main/pbx.c: Fix grammar in documentation for Goto() and GotoIf()
	  (closes issue ASTERISK-18023) Reported by: Tim Osman

2011-06-16 22:41 +0000 [r324048-324049]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c: Shame on me

	* include/asterisk/channel.h, main/channel.c,
	  channels/chan_local.c, channels/chan_sip.c: Lock the channel
	  before calling the setoption callback The channel needs to be
	  locked before calling these callback functions. Also,
	  sip_setoption needs to lock the pvt and a check p->rtp is
	  non-null before using it. Review:
	  https://reviewboard.asterisk.org/r/1220/

2011-06-16 18:12 +0000 [r323990]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_event.c: The test_event unit test is occasionally
	  failing. Wait for the special posted event to process before
	  adding a new subscription.

2011-06-16 15:58 +0000 [r323754-323932]  Terry Wilson <twilson@digium.com>

	* Makefile: Don't assume ASTDBDIR exists It most likely doesn't on
	  FreeBSD

	* tests/test_db.c: Remove now-useless cast of ARRAY_LEN

	* include/asterisk/utils.h: Make ARRAY_LEN() return the same type
	  on x86 and x86_64 systems

	* tests/test_db.c: Fix more ARRAY_LEN format string issues

	* /, main/features.c: Merged revisions 323733 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r323733 | twilson | 2011-06-15 13:13:00 -0500
	  (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
	  | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
	  recent DTMF change. This patch makes sure that dynamic features
	  are also checked when deciding whether or not to pass DTMF
	  through or store it for interpreting. (closes issue
	  ASTERISK-17914) Reported by: vrban ........ ................

2011-06-15 17:42 +0000 [r323730]  Jonathan Rose <jrose@digium.com>

	* res/res_config_pgsql.c: Adds locking to find_table in
	  res_configure_pgsql to prevent a crash. Bryonclark described the
	  problem as occuring during this function because of multiple
	  simultaneous database operations causing corruption against a
	  pgsqlConn object. (closes issue ASTERISK-17811) Reported by:
	  byronclark Patches: pgsql_find_table_locking.patch uploaded by
	  byronclark (license 1200)

2011-06-15 17:09 +0000 [r323672]  Terry Wilson <twilson@digium.com>

	* tests/test_db.c: Cast ARRAY_LEN to size_t for ast_logging 32-bit
	  and 64-bit machines return different types for ARRAY_LEN(), so
	  cast it before using in a format string.

2011-06-15 16:43 +0000 [r323669-323670]  Richard Mudgett <rmudgett@digium.com>

	* tests/test_event.c: Add a test to the event unit tests to catch
	  ASTERISK-18002. The new tests check to see if there are ANY
	  subscribers to the event type when ast_event_check_subscriber()
	  is not passed any specific ie values. (issue ASTERISK-18002)

	* main/event.c: [regression] Voicemail MWI is no longer sent. When
	  leaving a voicemail, the MWI message is never sent. The same
	  thing happens when checking a voicemail and marking it as read.
	  If you restart Asterisk, everything comes up at that state
	  correctly, but changes to the messages in voicemail causes the
	  light to not be set appropriately. Very easy to reproduce. * Made
	  ast_event_check_subscriber() return TRUE if there are ANY
	  subscribers to an event type when there are no restricting ie
	  values passed. This allows an event being queued to be queued.
	  (closes issue ASTERISK-18002) Reported by: lmadsen Tested by:
	  lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded
	  by rmudgett (License #5621) (closes issue ASTERISK-18019)

2011-06-15 16:09 +0000 [r323610]  Jonathan Rose <jrose@digium.com>

	* res/res_config_pgsql.c: Adds PQclear calls on result to various
	  parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported
	  by: byronclark Patches: pgsql_pqclear.patch uploaded by
	  byronclark (license 1200)

2011-06-15 15:31 +0000 [r323608]  Sean Bright <sean@malleable.com>

	* main/manager.c, /: Merged revisions 323579 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400
	  (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
	  2011) | 25 lines Resolve a segfault/bus error when we try to map
	  memory that falls on a page boundary. The fix for ASTERISK-15359
	  was incorrect in that it added 1 to the length of the mmap'd
	  region. The problem with this is that reading/writing to that
	  extra byte outside of the bounds of the underlying fd causes a
	  bus error. The real issue is that we are working with both a FILE
	  * and the raw fd underneath it and not synchronizing between
	  them. The code that was removed in ASTERISK-15359 was correct,
	  but we weren't flushing the FILE * before mapping the fd. Looking
	  at the manager code in 1.4 reveals that the FILE * in 'struct
	  mansession' is never used except to create a temporary file that
	  we immediately fdopen. This means we just need to write a 0 byte
	  to the fd and everything will just work. The other branches
	  require a call to fflush() which, while not a guaranteed fix,
	  should reduce the likelihood of a crash. This all makes sense in
	  my head. (closes issue ASTERISK-16460) Reported by:
	  Ravelomanantsoa Hoby (hoby) Patches:
	  issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
	  #5060) ........ ................

2011-06-15 00:50 +0000 [r323392-323456]  Richard Mudgett <rmudgett@digium.com>

	* main/event.c: Add missing break in ast_event_get_cached().

	* main/netsock2.c: Made ast_sockaddr_split_hostport() port warning
	  msgs more meaningful.

	* main/dnsmgr.c: Add more strict hostname checking to
	  ast_dnsmgr_lookup(). Change suggested in review. Review:
	  https://reviewboard.asterisk.org/r/1240/

2011-06-14 16:38 +0000 [r323371]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Changes contact use in build_peer to use the
	  FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this
	  was causing NAT=Yes to always use rport when present which was
	  against 1.6.2 behavior and the check itself was redundant since
	  the only way this segment of code could be reached was if
	  RPORT_PRESENT was already evaluated as true earlier. (closes
	  issue ASTERISK-17789) Reported by: byronclark Patches:
	  use_sip_nat_force_rport.patch uploaded by byronclark (license
	  1200)

2011-06-14 16:33 +0000 [r323370]  Terry Wilson <twilson@digium.com>

	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
	  main/rtp_engine.c, channels/chan_sip.c: Add rtpkeepalives back to
	  1.8 The RTP-engine conversion left out support for handling
	  rtpkeepalives. This patch adds them back. (closes issue
	  ASTERISK-17304) Reported by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/1226/

2011-06-13 20:22 +0000 [r323154-323234]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Additional documentation for bindaddr.
	  Note that bindaddr will only enable UDP instead of both UDP and
	  TCP which is what I would expect for backwards compatibility with
	  systems being upgraded which only support UDP transportation.
	  (closes issue ASTERISK-17976) Reported by: Sean Darcy

	* main/channel.c: Avoid dividing by zero with L() option to Dial()
	  Reported by: nicolasom Patches: issue-17995.patch - nicolasom
	  (License #5994)

	* res/res_agi.c: Tweak documentation for AGI Hangup command.
	  (closes issue ASTERISK-17999) Reported by: Ben Klang Patches:
	  hangup-doc.diff - uploaded by Ben Klang (License #5876)

2011-06-10 19:20 +0000 [r323040]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Unlock the sip channel during fax detection
	  like chan_dahdi does to prevent a deadlock with
	  ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
	  mnicholson

2011-06-10 15:29 +0000 [r322865-322981]  Terry Wilson <twilson@digium.com>

	* main/db.c: Avoid a DB1 infinite loop bug Explicity check the last
	  entry in the DB and make sure that we don't iterate past it.
	  Since there can be no duplicates, this just makes sure that we
	  stop after matching the last key. This patch also refactors the
	  code to get away from some code duplication. A previous patch
	  added many astdb tests and this patch passed them. Review:
	  https://reviewboard.asterisk.org/r/1259/

	* tests/test_db.c (added): Add some astdb unit tests

	* include/asterisk/astdb.h: Correct ast_db_deltree documentation
	  ast_db_deltree returns -1 on error, otherwise the number of
	  deletions

2011-06-09 17:37 +0000 [r322807]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: don't drop any voice frames when checking
	  for T.38 during early media (closes issue ASTERISK-17705) Review:
	  https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
	  oej

2011-06-09 16:31 +0000 [r322749]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/features.h, apps/app_directed_pickup.c,
	  main/features.c: Remove potential deadlock in call pickup race.
	  Deadlock is possible in ast_do_pickup() when holding the target
	  channel lock and trying to get the chan channel lock. Also,
	  holding the target lock when calling ast_channel_masquerade() is
	  not a good idea because that routine does deadlock avoidance. *
	  Removed the need to hold the target lock after marking the target
	  with a datastore and getting the connected line data off of the
	  target channel. * Moved can_pickup() to ast_can_pickup() in
	  features.c. Now all the call pickup methods use the same basic
	  call pickup availability check. Review:
	  https://reviewboard.asterisk.org/r/1234/

2011-06-09 14:06 +0000 [r322585]  Jonathan Rose <jrose@digium.com>

	* main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
	  tests/test_utils.c: Adds ast_escape_encoded utility to properly
	  handle escaping of quoted field before uri. This commit backports
	  a feature in trunk affecting initreqprep so that display name
	  won't be encoded improperly. Also includes unit tests for the
	  ast_escape_quoted function. This patch gives 1.8 a much improved
	  outlook in countries which don't use standard ASCII characters.
	  (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review:
	  https://reviewboard.asterisk.org/r/1235/

2011-06-08 20:46 +0000 [r322425-322484]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c: Ring all queue with more than 255 agents will
	  cause crash. 1. Create a ring-all queue with 500 permanent
	  agents. 2. Call it. 3. Asterisk will crash. The watchers array in
	  app_queue.c has a hard limit of 255. Bounds checking is not done
	  on this array. No sane person should put 255 people in a ring-all
	  queue, but we should not crash anyway. * Added bounds checking to
	  the watchers array. JIRA AST-464 JIRA SWP-2903

	* main/dnsmgr.c: SRV lookup attempted for SIP peers listed as an IP
	  address. Asterisk attempts to SRV lookup a host name even if the
	  host name is an IP address. Regression introduced when IPv6
	  support was added. * Restored the check in ast_dnsmgr_lookup() to
	  see if the given host name is an IP address. The IP address could
	  be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815)
	  Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett
	  Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett
	  (License #5621) Review: https://reviewboard.asterisk.org/r/1240/

2011-06-08 06:18 +0000 [r322322]  Gregory Nietsky <gregory@distrotech.co.za>

	* channels/chan_sip.c: Make handle_request_publish do dialog
	  expiration and destruction. This patch fixes
	  handle_request_publish so that it does dialog expiration and
	  destruction. Without this patch the incoming PUBLISH requests
	  will get stuck in the dialog list. Restarting asterisk is the
	  only way to remove them. Personal observation on one system the
	  server hung up while looping through the channels rendering
	  asterisk unusable and all sip phones unregisterd when they try
	  reregister more requests are added. (closes issue #18898)
	  Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj,
	  irroot Jira:
	  https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review:
	  https://reviewboard.asterisk.org/r/1253

2011-06-07 17:59 +0000 [r322189]  Paul Belanger <pabelanger@digium.com>

	* configs/sip_notify.conf.sample: Use correct syntax for 'sip
	  notify snom-reboot' (closes issue ASTERISK-17915)

2011-06-06 19:07 +0000 [r322069]  Jonathan Rose <jrose@digium.com>

	* main/asterisk.c, include/asterisk/logger.h: Fixes level toggling
	  for logger set levels since it was reversed (closes issue
	  ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H
	  Review: https://reviewboard.asterisk.org/r/1244/

2011-06-03 22:09 +0000 [r321812-321926]  Richard Mudgett <rmudgett@digium.com>

	* cdr/cdr_radius.c, cel/cel_radius.c: Asterisk crash when unloading
	  cdr_radius/cel_radius. The rc_openlog() API call is passed a
	  string that is used by openlog() to format log messages. The
	  openlog() does not copy the string it just keeps a pointer to it.
	  When the module is unloaded, the string is gone from memory.
	  Depending upon module load order and if the other module then has
	  an error, a crash happens. * Pass rc_openlog() a strdup'd string
	  with the understanding that there will be a small memory leak if
	  the cdr_radius/cel_radius modules are unloaded. * Call
	  rc_destroy() to free the rc handle memory when the module is
	  unloaded. JIRA AST-483 JIRA SWP-3062

	* main/ccss.c: Be more explicit for CCSS generic device state event
	  subscription. Make CCSS generic device state event subscription
	  specify the AST_EVENT_IE_STATE ie exists to be safe.

	* main/event.c, tests/test_event.c: Event subscription fixes. Must
	  commit the subscription fixes together with the integration
	  subscription tests. The subscription fixes cause an erroneously
	  passing test to fail. The new subscription tests detect errors
	  without the subscription fixes. * Added missing event_names[]
	  table entry. * Reworked
	  ast_event_check_subscriber()/match_sub_ie_val_to_event() to
	  correctly detect if a subscriber exists for the proposed event. *
	  Made match_ie_val() and match_sub_ie_val_to_event() check the
	  buffer length for RAW payload types. * Fixed error handling
	  memory leak in ast_event_sub_activate(), ast_event_unsubscribe(),
	  and ast_event_queue(). * Made ast_event_new() and
	  ast_event_check_subscriber() better protect themselves from an
	  invalid payload type. * Added container lock protection between
	  removing old cache events and adding the new cached event in
	  ast_event_queue_and_cache()/event_update_cache(). * Added new
	  event subscription tests.

	* main/event.c, include/asterisk/event.h: Constify subscription
	  description parameter string.

	* channels/chan_iax2.c, channels/chan_sip.c: Correct IAX2 and SIP
	  event subscription description string.

2011-06-03 18:32 +0000 [r321753]  Russell Bryant <russell@digium.com>

	* tests/test_astobj2.c: Backport an astobj2 unit test so that it
	  runs on 1.8 as well.

2011-06-03 13:17 +0000 [r321685]  Leif Madsen <lmadsen@digium.com>

	* configs/queues.conf.sample: Also document the 'queue-minute'
	  option. (closes issue #19386) Reported by: juanmol

2011-06-01 23:11 +0000 [r321547]  Richard Mudgett <rmudgett@digium.com>

	* main/cdr.c: CDR comment tweaks.

2011-06-01 20:10 +0000 [r321537]  Brett Bryant <bbryant@digium.com>

	* apps/app_voicemail.c: This patch fixes an issue with using the
	  wrong voicemail folders with greetings. (closes issue #17871)
	  Reported by: edhorton Patches: digium_bug_17871_2 uploaded by
	  fhackenberger (license 592) Tested by: edhorton, fhackenberger

2011-06-01 10:40 +0000 [r321528]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooh245.c: Fix double alerting, add forced
	  alerting before answer Fix double alerting (it wasn't fixed here
	  by issue #18542) Add forced alerting before connect (if it wasn't
	  before) Try to send all packets from outgoing queue rather than
	  one only Call goes into clearing state when disconnect command is
	  received (closes issue #19361) Reported by: vmikhelson Patches:
	  issue19361-3.patch uploaded by may213 (license 454) Tested by:
	  vmikhelson

2011-05-31 20:54 +0000 [r321517]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/dnsmgr.h, include/asterisk/acl.h: Update some
	  comments.

2011-05-31 18:52 +0000 [r321515]  David Vossel <dvossel@digium.com>

	* channels/chan_local.c: Chan_local locking cleanup. This patch
	  removes all of the unnecessary deadlock avoidance loops that
	  occur in chan_local. It also resolves an issue with a deadlock
	  triggered by local channel optimizations. (issue #18028) Review:
	  https://reviewboard.asterisk.org/r/1231/

2011-05-31 16:04 +0000 [r321511]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_sip.c: Enhance NOTICE message to know who couldn't
	  access the dialplan. (closes issue #19390) Reported by: lmadsen
	  Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen
	  (license 10) Tested by: russell

2011-05-28 00:27 +0000 [r321337-321436]  Richard Mudgett <rmudgett@digium.com>

	* res/res_agi.c: Some hagi launch cleanup. Inspired by issue 19256.
	  This patch would also fix the crash.

	* main/srv.c: Crash when using hagi and no servers are available.
	  When none of the servers returned by the SRV querey respond,
	  asterisk crashes. The problem is that if the loop over all the
	  SRV entries finishes then the srv_context has already been
	  cleaned up. * Make ast_srv_cleanup() check to see if the context
	  is already cleaned up. (closes issue #19256) Reported by:
	  byronclark

	* apps/app_privacy.c: The app_privacy args have undocumented
	  "options" position, interferes with "context" position. * Add
	  documention for unused "options" position to match existing code.
	  (closes issue #19273) Reported by: mdavenport

2011-05-27 21:54 +0000 [r321333-321335]  Leif Madsen <lmadsen@digium.com>

	* include/asterisk/frame.h, main/file.c: Fix issue with playback of
	  H.261 video. (closes issue #19379) Reported by: neutrino88
	  Patches: videoprompt.patch uploaded by neutrino88 (license 297)
	  (changes by russell)

	* main/features.c: Allow parking lot hints and musicclass to be
	  set. (closes issue #19378) Reported by: sboily_proformatique
	  Patches: pf_parkinghint_music_fix uploaded by sboily
	  proformatique (license 206) Tested by: russell

2011-05-27 21:31 +0000 [r321330]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_privacy.c: The app_privacy args have undocumented
	  "options" position, interferes with "context" position. * Add
	  documention for unused "options" position to match existing code.
	  The trunk(v1.10) version will remove the unused options position.
	  (closes issue #19273) Reported by: mdavenport

2011-05-27 14:59 +0000 [r321273]  Jonathan Rose <jrose@digium.com>

	* channels/sip/reqresp_parser.c: markm committed a patch I was
	  working on yesterday, this fixes it to mesh up with suggestions
	  by mnicholson.

2011-05-27 08:31 +0000 [r321211]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/features.c: Fix *8 directed pickup locks system during
	  pickupsound play out move playout from sip_pickup_thread to
	  bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
	  threads trying to write audio to same channel. In addition fixes
	  choppy audio beep in issue 19177. (issue #18654) (issue #19177)
	  Reported by: Docent Patches: review1232-1.88888888 alecdavis
	  (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/1232/

2011-05-26 21:48 +0000 [r321100-321155]  Mark Murawki <markm@intellasoft.net>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c: Fixed build
	  problem with dev mode enabled, which was caused by commit 321100.
	  Reformulated patch to be more generic. Moved the sip uri parse
	  variable initalization to parse_uri_full in reqresp_parser.c.
	  This will ensure that any use of parse uri will have null output
	  variables if the parse fails. (closes issue #19346) Reported by:
	  kobaz Tested by: kobaz,JonathanRose Review: [full review board
	  URL with trailing slash]

	* main/netsock2.c, channels/chan_sip.c: ast_sockaddr_resolve() in
	  netsock2.c may deref a null pointer Added a null check in
	  netsock2 ast_sockaddr_resolve() as well as added default
	  initalizers in chan_sip parse_uri_legacy_check() to make sure
	  that invalid uris will make null (and not undefined)
	  user,pass,domain,transport variables (closes issue #19346)
	  Reported by: kobaz Patches: netsock2.patch uploaded by kobaz
	  (license 834) Tested by: kobaz, Marquis

2011-05-26 18:10 +0000 [r321044]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/netsock2.h: Update ast_sockaddr comment with an
	  important note.

2011-05-26 17:29 +0000 [r321042]  Terry Wilson <twilson@digium.com>

	* main/rtp_engine.c: Initialize stack-allocated ast_sockaddrs
	  before use It is important to always initialize ast_sockaddrs
	  before use--even if they are passed to ast_sockaddr_copy as the
	  underlying storage could be bigger than what ends up being
	  copied--leaving part of the data unitialized.

2011-05-26 15:57 +0000 [r320947]  Russell Bryant <russell@digium.com>

	* channels/chan_alsa.c, channels/chan_mgcp.c: Remove some variables
	  that were set but unused.

2011-05-25 22:25 +0000 [r320796-320883]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Native SIP CCSS sends bad CC cancel
	  SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC
	  request has incorrect To/From SIP headers. They are reversed and
	  the dialog tags are the same when they should not be. If pedantic
	  mode was disabled, then the cancel would have succeeded despite
	  the incorrect message. * The SIP_OUTGOING flag was not set
	  correctly for the dialog and I had to move some CC subscribe
	  handling code as a result. * Initialized the dialog subscribed
	  type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE
	  message comes in and the CC instance is not found, the 404
	  response was duplicated. JIRA AST-568 JIRA SWP-3493

	* UPGRADE.txt, CHANGES, apps/app_queue.c, apps/app_dial.c,
	  main/channel.c, main/manager.c, apps/app_meetme.c,
	  apps/app_fax.c, main/features.c: The AMI Newstate event contains
	  different information between v1.4 and v1.8. The addition of
	  connected line support in v1.8 changes the behavior of the
	  channel caller ID somewhat. The channel caller ID value no longer
	  time shares with the connected line ID on outgoing call legs. The
	  timing of some AMI events/responses output the connected line ID
	  as caller ID. These party ID's are now separate. * The
	  ConnectedLineNum and ConnectedLineName headers were added to many
	  AMI events/responses if the CallerIDNum/CallerIDName headers were
	  also present. (closes issue #18252) Reported by: gje Tested by:
	  rmudgett Review: https://reviewboard.asterisk.org/r/1227/

	* include/asterisk/channel.h, main/channel.c, main/features.c: Give
	  zombies a safe channel driver to use. Recent crashes from zombie
	  channels suggests that they need a safe home to goto. When a
	  masquerade happens, the physical part of the zombie channel is
	  hungup. The hangup normally sets the channel private pointer to
	  NULL. If someone then blindly does a callback to the channel
	  driver, a crash is likely because the private pointer is NULL.
	  The masquerade now sets the channel technology of zombie channels
	  to the kill channel driver. Related to the following issues:
	  (issue #19116) (issue #19310) Review:
	  https://reviewboard.asterisk.org/r/1224/

2011-05-25 00:49 +0000 [r320716]  Terry Wilson <twilson@digium.com>

	* addons/chan_mobile.c: Cast data as char * before using S_OR This
	  is required for compiling successfully under dev mode

2011-05-23 17:53 +0000 [r320650]  Richard Mudgett <rmudgett@digium.com>

	* CHANGES, main/manager.c: Add ConnectedLineNum/Name headers to
	  output of AMI action Status. * Add ConnectedLineNum and
	  ConnectedLineName headers to the output of the AMI action Status.
	  This makes it easier to find out who the channel is connected to
	  without having to lookup BridgedChannel or when they are
	  connected to an application (e.g.: VoiceMail) which has no
	  bridged channel. * Bridged channels with no CallerID had ""
	  instead of "<unknown>" output, that might be a bug as "<unknown>"
	  was what older versions used. (closes issue #18158) Reported by:
	  gareth Patches: svn-292308.diff uploaded by gareth (license 208)

2011-05-23 16:19 +0000 [r320573]  Tilghman Lesher <tilghman@meg.abyt.es>

	* configure, configure.ac: GNU libiconv uses symbol "libiconv_open"
	  instead of "iconv_open". (closes issue #19344) Reported by:
	  rohanl Patches: iconv-check.patch uploaded by rohanl (license
	  1284)

2011-05-23 16:18 +0000 [r320568]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, /: Merged revisions 320562 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011)
	  | 9 lines Adds missing part to the ast_tcptls_server_start fails
	  second attempt to bind patch. (closes issue #19289) Reported by:
	  wdoekes Patches:
	  issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
	  wdoekes (license 717) ........

2011-05-23 15:47 +0000 [r320560]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: Don't generate spurious "No: command not
	  found" messages when running the configure script on a system
	  that has neither gmime-config nor pkg-config.

2011-05-23 14:33 +0000 [r320504]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Fixes segfault occuring in chan_sip.c at
	  __set_address_from_contact Checks to see if domain contains
	  anything before sending it off to ast_sockaddr_resolve which is
	  where the segfault was occuring due to null str. (closes issue
	  #18857) Reported by: sybasesql Review:
	  https://reviewboard.asterisk.org/r/1225/

2011-05-22 23:34 +0000 [r320445]  Tilghman Lesher <tilghman@meg.abyt.es>

	* res/res_odbc.c, /: Merged revisions 320444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011)
	  | 8 lines Don't crash when the connection fails. (closes issue
	  #19250) Reported by: seadweller Patches:
	  20110514__issue19250.diff.txt uploaded by tilghman (license 14)
	  Tested by: seadweller, sum ........

2011-05-20 21:39 +0000 [r320338]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, /: Merged revisions 320271 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011)
	  | 8 lines Fixes issue with ast_tcptls_server_start failing on
	  second attempt to bind. (closes issue #19289) Reported by:
	  wdoekes Patches:
	  issue19289_delay_old_address_setting_tcptls.patch uploaded by
	  wdoekes (license 717) ........

2011-05-20 20:49 +0000 [r320237]  Richard Mudgett <rmudgett@digium.com>

	* /, apps/app_meetme.c: Merged revisions 320236 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500
	  (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
	  | 13 lines The meetme CLI command completion leaves conferences
	  mutex locked. When issuing a meetme kick CLI command and an
	  invalid (non-existent) conference number is specified, pressing
	  Tab leaves the conferences mutex locked and, therefore, all
	  conferences deadlock. Add missing unlock. (closes issue #19336)
	  Reported by: zvision Patches: app_meetme.diff uploaded by zvision
	  (license 798) ........ ................

2011-05-20 18:48 +0000 [r320180]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: This commit modifies the way polling is done
	  on TLS sockets. Because of the buffering the TLS layer does,
	  polling is unreliable. If poll is called while there is data
	  waiting to be read in the TLS layer but not at the network layer,
	  the messaging processing engine will not proceed until something
	  else writes data to the socket, which may not occur. This change
	  modifies the logic around TLS sockets to only poll after a failed
	  read on a non-blocking socket. This way we know that there is no
	  data waiting to be read from the buffering layer. (closes issue
	  #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
	  mnicholson (license 96) Tested by: mnicholson

2011-05-20 18:12 +0000 [r320162]  Jonathan Rose <jrose@digium.com>

	* apps/app_voicemail.c: Fixes an imapfolder related crash
	  imapfolders being set in the general section of voicemail would
	  cause the inbox folder name to change. Since sound file names are
	  made based on the names of the folders, this would cause the
	  audio related to that folder name to change and if Asterisk
	  attempted to play it, the channel would instantly hang up when
	  the audio file couldn't be found. This patch searches for the
	  name of the folder first to leave existing behavior in tact and
	  if that fails, it uses the normal inbox name to get the sound
	  file instead. (closes issue #16104) Reported by: blkline Review:
	  https://reviewboard.asterisk.org/r/1215/

2011-05-20 17:03 +0000 [r319997-320059]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Misc comment cleanup in features.c.

	* main/channel.c, main/features.c: Crash while transferring a call
	  during DTMF feature timeout. When a call is being attended
	  transferred during the time between AST_FRAME_DTMF_BEGIN and
	  AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so
	  tech data is not available), making ast_dtmf_stream() segfault
	  when it tries to send the DTMF digit (at least with SIP
	  channels). Patch based on feature-end-zombie.patch uploaded by
	  Irontec (license 1256) * Check for zombies when
	  ast_channel_bridge() returns. * Guarantee that the fo parameter
	  value is initialized in ast_channel_bridge() before any returns.
	  (closes issue #19116) Reported by: Irontec Tested by: rmudgett

	* apps/app_directed_pickup.c, main/features.c: Change some variable
	  names to make pickup code easier to understand.

	* apps/app_directed_pickup.c, main/features.c: Crash when using
	  directed pickup applications. The directed pickup applications
	  can cause a crash if the pickup was successful because the
	  dialplan keeps executing. This patch does the following: *
	  Completes the channel masquerade on a successful pickup before
	  the application returns. The channel is now guaranteed a zombie
	  and must not continue executing the dialplan. * Changes the
	  return value of the directed pickup applications to return zero
	  if the pickup failed and nonzero(-1) if the pickup succeeded. *
	  Made some code optimizations that no longer require re-checking
	  the pickup channel to see if it is still available to pickup.
	  (closes issue #19310) Reported by: remiq Patches:
	  issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
	  Tested by: alecdavis, remiq, rmudgett Review:
	  https://reviewboard.asterisk.org/r/1221/

2011-05-20 13:28 +0000 [r319938]  Jonathan Rose <jrose@digium.com>

	* configs/sip.conf.sample, channels/sip/include/sip.h,
	  channels/chan_sip.c: Adds legacy_useroption_parsing to address
	  interoperability concerns. With the new option engaged, Asterisk
	  should interpret user fields with useroptions contained within
	  the userfield of the uri by stripping them out of the original
	  message whenever a semicolon is encountered in the userfield
	  string. (closes issue #18344) Reported by: danimal Tested by:
	  jrose Review: https://reviewboard.asterisk.org/r/1223/

2011-05-19 23:28 +0000 [r319920]  Terry Wilson <twilson@digium.com>

	* main/bridging.c, include/asterisk/bridging_technology.h,
	  include/asterisk/bridging.h: Revert part of a change to the
	  bridging API code The capabilities used in the bridging API are
	  very different than the ones used for formats. When the
	  conversion was made expanding the bit width of codecs, the
	  bridging code was accidentally accosted in ways that it didn't
	  deserve.

2011-05-19 18:32 +0000 [r319866]  Jonathan Rose <jrose@digium.com>

	* main/features.c: Fix Randomize option on Park() The randomize
	  option was generally not working like it should have at all on
	  Park(). This patch restores intended functionality. (closes issue
	  #18862) Reported by: davidw Tested by: jrose Review:
	  https://reviewboard.asterisk.org/r/1222/

2011-05-19 17:59 +0000 [r319812]  Mark Murawki <markm@intellasoft.net>

	* cel/cel_odbc.c: In cel_odbc, an uninitialized RWLIST is attempted
	  to be locked. Added INIT and DESTROY for the RWLIST odbc_tables
	  (closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch
	  uploaded by kobaz (license 834)

2011-05-19 16:50 +0000 [r319758]  Richard Mudgett <rmudgett@digium.com>

	* main/ccss.c: CCSS generic agent with POTS and ISDN phones fail
	  caller busy call-back test. If the following is true after a CCSS
	  activation: * The generic agent is for an analog phone or ISDN
	  phone. (Caller party) * The called party becomes available. * The
	  caller party is not available. When the caller party becomes
	  available, the caller is not alerted to the called party being
	  available. The generic agent still thinks the caller is busy. *
	  Fixed the generic agent device state event subscription to look
	  for all device states that are considered available. *
	  Encapsulated the device state test for CCSS generic device
	  available in cc_generic_is_device_available(). Made the generic
	  agent and monitor use the new function instead of the manually
	  coded inline equivalent. JIRA AST-559 JIRA SWP-3462

2011-05-18 23:15 +0000 [r319529-319654]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 319653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r319653 | twilson | 2011-05-18 16:11:57 -0700
	  (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
	  | 8 lines Make sure everyone gets an unhold when a transfer
	  succeeds Some phones, like the Snom phones, send a hold to the
	  transfer target after before sending the REFER. We need to make
	  sure that we unhold the parties that are being connected after
	  the masquerade. If Local channels with the /nm option are used
	  when dialing the parties, hold music would still be playing on
	  the transfer target, even after being connected with the
	  transferee. ........ ................

	* channels/chan_sip.c: Unbreak the storing of registrations for
	  restart The fix for issue 18882 broke retrieving non-realtime
	  peers from the ast_db on restart/reload. This patch tries to
	  unbreak things while leaving the intent of the original fix
	  intact. (closes issue #19318) Reported by: remiq Patches:
	  diff.txt uploaded by twilson (license 396) Tested by: lmadsen,
	  remiq

	* apps/app_dial.c, /: Merged revisions 319528 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r319528 | twilson | 2011-05-18 13:02:06 -0700
	  (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
	  | 10 lines Fix app_dial ring groups Revert part of r315643. We
	  need to remove the datastore here as well. The code in bridging
	  code will catch anything that app_dial might miss. (closes issue
	  #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
	  uploaded by elguero (license 37) ........ ................

2011-05-17 21:57 +0000 [r319469]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: Merged revision 319468 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue,
	  17 May 2011) | 15 lines The mISDN HDLC mode is prevented on
	  dialed channels. The use of mISDN HDLC mode is prevented if the
	  mISDN dial technology option 'h1' is used when config option
	  astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which
	  prevents the use of HDLC mode. Instead of setting the channel to
	  HDLC mode it is set to transparent(no dsp, no hdlc), although
	  hdlc is not "no hdlc". I.e the logging message is correct, but
	  the if condition is not. Make check the nodsp and hdlc flags.
	  JIRA ABE-2787 JIRA SWP-3437 ..........

2011-05-17 12:53 +0000 [r319365-319367]  Leif Madsen <lmadsen@digium.com>

	* apps/app_voicemail.c: Don't create [general] voicemail context
	  when using users.conf Prior to this patch, app_voicemail would
	  create a [general] context when parsing users.conf. (closes issue
	  #18891) Reported by: pdugas Patches:
	  app_voicemail-ignore-general.patch uploaded by pdugas (license
	  1222) app_voicemail-ignore-general-style-guidelines.patch
	  uploaded by seanbright (license 71) Tested by: pdugas

	* contrib/init.d/rc.debian.asterisk: Make Debian init script lsb
	  compliant (closes issue #18896) Reported by: manwe Patches:
	  debian_init_lsb.patch uploaded by manwe (license 1223)

2011-05-16 21:00 +0000 [r319261]  Jonathan Rose <jrose@digium.com>

	* main/dsp.c: Makes busy detection in dsp.c always allow for at
	  least one frame (20ms) of error so that 200ms tone lengths don't
	  get ignored by single frame error lengths.

2011-05-16 20:33 +0000 [r319259]  Richard Mudgett <rmudgett@digium.com>

	* main/ccss.c: Deadlock between generic CCSS agent and native ISDN
	  CCSS. Deadlock can occur when the generic CCSS agent is deleting
	  duplicate CC offers and the native ISDN CC driver is processing
	  an incoming CC message. The cc_core_instances container lock
	  cannot be held when an agent or monitor callback is invoked
	  without the possibility of a deadlock. * Make
	  kill_duplicate_offers() remove the reference in cc_core_instances
	  outside of the container lock. JIRA AST-566 JIRA SWP-3469

2011-05-16 18:17 +0000 [r319204]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 319202 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011)
	  | 4 lines Unlink a peer from peers_by_ip when expiring a
	  registration Review: https://reviewboard.asterisk.org/r/1218/
	  ........

2011-05-16 15:57 +0000 [r319145]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 319144 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011)
	  | 2 lines Fixes issue with peer ref-counting during
	  handle_request_subscribe. (closes issue #19293) Reported by:
	  irroot ........

2011-05-16 15:53 +0000 [r319142]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Make sure tcptls_session exists before
	  dereferencing it. (closes issue #19192) Reported by: stknob
	  Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
	  Chainsaw (license 723) Tested by: vois, Chainsaw

2011-05-16 14:35 +0000 [r319085]  Paul Belanger <pabelanger@digium.com>

	* res/res_http_post.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Support gmime-2.4 (closes issue #18863) Reported
	  by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir
	  (license 46) Tested by: tzafrir Review:
	  https://reviewboard.asterisk.org/r/1213/

2011-05-16 14:26 +0000 [r319083]  David Vossel <dvossel@digium.com>

	* formats/format_wav.c: Fixes Big Endian build issue. (closes issue
	  #19298) Reported by: tzafrir

2011-05-13 18:09 +0000 [r318917-318921]  Brett Bryant <bbryant@digium.com>

	* main/channel.c: Fixes a segmentation fault in dynamic hints when
	  a channel technology isn't loaded for a hint. (closes issue
	  #18495) Reported by: bertrand Tested by: bertrand

	* res/res_srtp.c: This patch fixes an issue with SRTP which makes
	  HOLD/UNHOLD impossible when too much time has passed between
	  sending audio. (closes issue #18206) Reported by: bernhardsi
	  Patches: res_srtp_unhold.patch uploaded by bernhards (license
	  1138) Tested by: bernhards, notthematrix

	* channels/chan_sip.c: This patch allows TCP peers into the ast_db
	  where they were previously restricted. (closes issue #18882)
	  Reported by: cmaj Patches:
	  patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
	  uploaded by cmaj (license 830) Tested by: cmaj

2011-05-13 16:28 +0000 [r318783-318868]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: CDR's are being written immediately on caller
	  hangup. CDR's are being written immediately on caller hangup. The
	  dialplan is not able to modify it in the h exten. The h exten in
	  the initial context is not run before closing CDR's when the
	  bridge is unlinked if a macro is active and does not have an h
	  exten. * Make ast_bridge_call() check for an h exten in the
	  current context and if a macro is active then the initial
	  context. The first h exten found is then run before closing the
	  CDR. (closes issue #18212) Reported by: leearcher Patches:
	  issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested
	  by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/

	* channels/sig_pri.c: PRI early media won't ring. And another way
	  to pass early media. Don't indicate that there is inband
	  information present, just assume that the B channel is connected.
	  * Restore clearing the dialing flag Rx squelch unconditionally
	  when a PROCEEDING message comes in. (closes issue #19268)
	  Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
	  rmudgett (license 664) Tested by: tbsky

2011-05-12 23:35 +0000 [r318720]  Matthew Nicholson <mnicholson@digium.com>

	* channels/sip/reqresp_parser.c: Handle ipv6 addresses in the
	  sent-by Via: field. This change fixes a regression in via header
	  parsing and ipv6 handling. (closes issue #18951)

2011-05-12 22:52 +0000 [r318671]  Alec L Davis <sivad.a@paradise.net.nz>

	* include/asterisk/features.h, channels/chan_sip.c,
	  apps/app_directed_pickup.c, main/features.c: Fix directed group
	  pickup feature code *8 with pickupsounds enabled Since 1.6.2, the
	  new pickupsound and pickupfailsound in features.conf cause many
	  issues. 1). chan_sip:handle_request_invite() shouldn't be playing
	  out the fail/success audio, as it has 'netlock' locked. 2).
	  dialplan applications for directed_pickups shouldn't beep. 3).
	  feature code for directed pickup should beep on success/failure
	  if configured. Created a sip_pickup() thread to handle the pickup
	  and playout the audio, spawned from handle_request_invite. Moved
	  app_directed:pickup_do() to features:ast_do_pickup(). Functions
	  below, all now use the new ast_do_pickup() app_directed_pickup.c:
	  pickup_by_channel() pickup_by_exten() pickup_by_mark()
	  pickup_by_part() features.c: ast_pickup_call() (closes issue
	  #18654) Reported by: Docent Patches:
	  ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
	  585) Tested by: lmadsen, francesco_r, amilcar, isis242,
	  alecdavis, irroot, rymkus, loloski, rmudgett Review:
	  https://reviewboard.asterisk.org/r/1185/

2011-05-11 18:47 +0000 [r318549-318550]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Comment out the REF_DEBUG that slipped in
	  during debugging

	* /, channels/chan_sip.c: Merged revisions 318548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011)
	  | 19 lines Clean up several chan_sip reference leaks Several
	  situations in the code could lead to peers or sip_pvt references
	  being leaked. This would cause RTP ports to never be destroyed
	  (leading to exhaustion of all available RTP ports) and memory
	  leaks. The original patch for this issue from rgagnon was the
	  result of an obscene amount of testing and hard work, for which I
	  am very grateful. I did some cleanup and added a few additional
	  refcount fixes that I found. (closes issue #17255) Reported by:
	  kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff
	  uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson,
	  wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/
	  Review: https://reviewboard.asterisk.org/r/1207/ Review:
	  https://reviewboard.asterisk.org/r/1210/ ........

2011-05-10 23:41 +0000 [r318499]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_ss7.c: Unable to pickup
	  DAHDI/PRI call because call state is reported as DIALING. The
	  channel state is not updated to RINGING when an ALERTING message
	  is received. Regression caused when sig_pri.c (also sig_ss7.c)
	  extracted from chan_dahdi.c. * Added missing channel state update
	  to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN
	  and SS7. (closes issue #19257) Reported by: alecdavis Patches:
	  issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
	  Tested by: alecdavis, rmudgett

2011-05-10 18:46 +0000 [r318485]  Leif Madsen <lmadsen@digium.com>

	* main/manager.c: Filter out blacklisted manager events when using
	  eventfilter. Merging change from trunk in revision 306432.
	  (closes issue #19260) Reported by: dhubbard Tested by: dhubbard

2011-05-10 15:13 +0000 [r318436]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: chan_iax2: change LOG_NOTICE to LOG_DEBUG
	  in iax2_read().

2011-05-09 23:15 +0000 [r318351]  Richard Mudgett <rmudgett@digium.com>

	* res/Makefile, res/res_features.exports.in (removed): Remove
	  references to res_features and its export file. The contents of
	  res/res_features.c was moved to into main/features.c awhile ago.
	  There is no longer any need for the res/Makefile to reference
	  res_features or the res_features linker exports file to exist.

2011-05-09 20:23 +0000 [r318337]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 318331 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011)
	  | 12 lines Don't offer video to directmedia callee unless caller
	  offered it as well Make sure that when directmedia is enabled,
	  that video is not offered to the callee even if it supports it.
	  p->vrtp will not exist since the caller didn't offer video.
	  (closes issue #19195) Reported by: one47 Patches:
	  sip_cant_add_video_rtp uploaded by one47 (license 23) ........

2011-05-09 19:07 +0000 [r318282]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Hangup extension executed twice. When a user
	  hangs up a call, in certain circumstances, the hangup extension
	  can end up being executed twice: 1) If a call is bridged and the
	  'h' extension executes the Hangup application, then the 'h'
	  extension will be executed twice. 2) If a call is bridged within
	  a macro (Dial or Queue), it has its own 'h' extension, the main
	  context also has an 'h' extension, and the macro 'h' extension
	  executes the Hangup application, then both 'h' extensions will be
	  executed. * Revert originally commited fix for #16106 and just
	  set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in
	  ast_bridge_call(). The bridge code just executed an 'h' extension
	  so the main PBX loop does not need to execute one as well. (issue
	  #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd

2011-05-09 17:09 +0000 [r318233]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 318230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011)
	  | 7 lines Fixes cases where sip_set_rtp_peer can return too early
	  during media path reset. (closes issue #19225) Reported by: one47
	  Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23)
	  ........

2011-05-09 16:57 +0000 [r318231]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Don't get early media for ISDN on outgoing
	  calls. It looks to be a long-standing misinterpretation of the
	  progress indicator ie values: 1 - Call is not end-to-end ISDN;
	  further call progress information may be available in-band. 8 -
	  In-band information or an appropriate pattern is now available.
	  Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not
	  handled as early media probably because the meaning of the second
	  half of it's description was overlooked. * Test to see if either
	  PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8)
	  bits are set to open the media path. (closes issue #18868)
	  Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded
	  by rmudgett (license 664) Tested by: satish_lx .......... No
	  inband progress on PRI_EVENT_RINGING even if inband flag set. My
	  ISDN-PRI provider sends an ALERTING with "Inband information or
	  appropriate pattern now available", but Asterisk only generates
	  and passes the RING to the SIP extension, not the inband message.
	  Unfortunately, the inband message is not a ringback tone but a
	  prompt that says the number is not in service. The SIP extension
	  then hears two rings and the call is hungup which confuses the
	  caller. * Post an AST_CONTROL_PROGRESS as well as opening the
	  media path if inband audio is indicated with an ALERTING message.
	  (closes issue #19246) Reported by: cristiandimache Patches:
	  issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested
	  by: cristiandimache

2011-05-09 14:18 +0000 [r318148]  Jonathan Rose <jrose@digium.com>

	* configs/features.conf.sample: Documenting an observed behavior of
	  features in features.conf. Since parkinglots use an integer for
	  the parkinglot extensions, leading zeros specified in the
	  configuration file are ignored.

2011-05-09 14:09 +0000 [r318142]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Make indicate/control frames WRITE events on
	  framehooks. Also, if a framehook returns a non-control frame,
	  don't forward it to the channel. (closes issue #19251) Reported
	  by: irroot Patches: (modified) framehook_indicate.patch2 uploaded
	  by irroot (license 52) Tested by: irroot

2011-05-07 23:35 +0000 [r318055-318057]  Russell Bryant <russell@digium.com>

	* res/res_config_curl.c: res_config_curl: fix a crash with static
	  realtime. (closes issue #18413) Reported by: jmls Patches:
	  20101202__issue18413.diff.txt uploaded by tilghman (license 14)
	  Tested by: jmls

	* channels/chan_iax2.c: chan_iax2: Don't overwrite port found with
	  an SRV lookup. (closes issue #17291) Reported by: jcovert
	  Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by
	  jcovert (license 551)

2011-05-06 21:49 +0000 [r317967-317969]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Use the right variable to print the time in a
	  debug message. The original patch also increased some buffer
	  sizes, but that was already done in this version. (closes issue
	  #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch
	  uploaded by sysreq (license 1009)

	* apps/app_meetme.c: Fix some more "set but unused" compiler
	  warnings.

2011-05-06 21:06 +0000 [r317918]  David Vossel <dvossel@digium.com>

	* res/res_rtp_asterisk.c: Fixes missing colon from To/From headers
	  in RTCP manager events. (closes issue #18221) Reported by:
	  clegall_proformatique Patches: 18221_1.patch uploaded by ebroad
	  (license 878)

2011-05-06 21:06 +0000 [r317861-317917]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Fix calculation of free RAM to make minmemfree option
	  work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff
	  uploaded by loic (license 1020)

	* channels/chan_sip.c: chan_sip: Destroy variables on a sip_pvt
	  before copying vars from the sip_peer. Don't duplicate variables
	  on the sip_pvt. Just reset the variable list each time. (closes
	  issue #19202) Reported by: wdoekes Patches:
	  issue19202_destroy_challenged_invite_chanvars.patch uploaded by
	  wdoekes (license 717)

	* channels/chan_sip.c: chan_sip: fix a deadlock in
	  check_rtp_timeout. Don't block doing silly deadlock avoidance.
	  Just return and try again later. The funciton gets called often
	  enough that it's fine. Also, this change was already made in
	  trunk. (closes issue #18791) Reported by: irroot Patches:
	  chan_sip.rtptimeout.patch uploaded by irroot (license 52)

	* channels/chan_sip.c: URI encode less characters in the RPID and
	  Contact headers. If this change causes any problems, we will need
	  to backport the more extensive uri encoding and decoding handling
	  changes that are in trunk/1.10. (closes issue #18686) Reported
	  by: wolfgang Patches: quick-and-dirty.patch uploaded by wdoekes
	  (license 717) Tested by: wdoekes, devellow, wolfgang, mav3rick

2011-05-06 19:31 +0000 [r317858]  Matthew Nicholson <mnicholson@digium.com>

	* pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
	  autoservice in pbx_lua if pbx_lua already started one and don't
	  stop one if we didn't start one. Also start and stop the
	  autoservice when transferring control from and to the pbx.

2011-05-06 19:24 +0000 [r317805-317837]  Russell Bryant <russell@digium.com>

	* addons/app_mysql.c: Fix a crash in the MySQL() application. This
	  code was not handling channel datastores safely. The channel must
	  be locked. (closes issue #17964) Reported by: wuwu Patches:
	  issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license
	  71) Tested by: wuwu

	* contrib/realtime/mysql/sipfriends.sql: Add a new sipfriends.sql
	  for MySQL that has more fields in it. (closes issue #16399)
	  Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by
	  pabelanger (license 224)

2011-05-06 16:19 +0000 [r317670]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Fix SIP connected line updates. This patch
	  fixes a couple SIP connected line update problems: 1) The
	  connected line needs to be updated when the initial INVITE is
	  sent if there is a peer callerid configured. Previously, the
	  connected line information did not get reported until the call
	  was connected so SIP could not report connected line information
	  in ringing or progress messages. 2) The connected line should not
	  be updated on initial connect if there is no connected line
	  information. Previously, all it did was wipe out any default
	  preset CONNECTEDLINE information set by the dialplan with empty
	  strings. (closes issue #18367) Reported by: GeorgeKonopacki
	  Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664)
	  Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1199/

2011-05-06 08:18 +0000 [r317584]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c, /: Merged revisions 317575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r317575 | twilson | 2011-05-06 01:04:17 -0700
	  (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
	  | 6 lines Re-fix queue round-robin This part of the change for
	  r315596 was incorrect. No bridge occurs when doing a roundrobin
	  dial and no one answers, so this code shouldn't have been
	  removed. ........ ................

2011-05-05 23:46 +0000 [r317425-317530]  Russell Bryant <russell@digium.com>

	* Makefile: If the configure script runs, force a rebuild of
	  menuselect-tree. Some contents in the menuselect tree are
	  dependent on configure script parameters, namely
	  --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis
	  Patches: issue_17219.rev1.txt uploaded by russell (license 2)

	* contrib/realtime/mysql/queue_log.sql,
	  contrib/realtime/mysql/sipfriends.sql: Fix some more realtime
	  MySQL schema issues. (closes issue #18537) Reported by: denzs
	  Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182)
	  queue_log.sql.svndiff uploaded by denzs (license 1182)
	  meetme.sql.svndiff uploaded by denzs (license 1182)

	* contrib/realtime/mysql/sipfriends.sql,
	  contrib/realtime/mysql/meetme.sql: Fix some errors in sample
	  MySQL realtime schema files. (closes issue #18915) Reported by:
	  Dovid Patches: sipfriends.patch uploaded by Dovid (license 652)
	  meetme.patch uploaded by Dovid (license 652)

	* cdr/cdr_syslog.c: Don't lose cdr_syslog config on a reload.
	  (closes issue #18679) Reported by: enegaard Patches:
	  issue18679_seanbright.patch uploaded by seanbright (license 71)
	  Tested by: enegaard

	* channels/chan_alsa.c, channels/chan_console.c,
	  channels/chan_oss.c, channels/chan_mgcp.c,
	  channels/misdn_config.c, channels/chan_unistim.c,
	  channels/chan_usbradio.c, channels/chan_dahdi.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c: Fix some consistency issues with
	  jitterbuffer config. Store the defaults noted in the sample
	  config files in the jitterbuffer config data structure. This
	  makes the CLI commands that output these settings show the right
	  thing. Also only show the settings that are relevant in the
	  settings CLI commands, based on which jitterbuffer is selected
	  and whether it's enabled. (closes issue #19083) Reported by:
	  rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by
	  rgagnon (license 1202)

	* pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
	  (closes issue #19055) Reported by: jamhed Patches:
	  lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
	  Tested by: mnicholson, jamhed

	* channels/iax2-provision.c, pbx/pbx_dundi.c,
	  channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c,
	  res/res_jabber.c, res/res_config_sqlite.c, cel/cel_pgsql.c,
	  channels/chan_jingle.c, channels/sip/sdp_crypto.c,
	  res/res_config_odbc.c, channels/chan_sip.c, res/res_crypto.c,
	  pbx/pbx_lua.c: Fix more "set but unused" warnings.

	* main/dsp.c: Only display inband DTMF warning if inband DTMF
	  detection is enabled. (closes issue #18901) Reported by: irroot

	* apps/app_rpt.c: Fix potential memory leak, and use of
	  uninitialized memory. (closes issue #16476) Reported by: junky
	  Patches: M16476.diff uploaded by junky (license 177)

	* main/manager.c: Add missing ActioID handling to Events action.
	  (closes issue #18949) Reported by: edersohe Patches:
	  0018949.patch uploaded by edersohe (license 1228)

2011-05-05 20:25 +0000 [r317370]  Sean Bright <sean@malleable.com>

	* addons/res_config_mysql.c: Don't duplicate our data on the stack
	  and just use the MYSQL_ROW directly. With large result sets we
	  were blowing out the stack. (closes issue #19090) Reported by:
	  mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by
	  seanbright (license 71) Tested by: mickecarlsson

2011-05-05 19:55 +0000 [r317336]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Increase buffer size to be PATH_MAX for a path.
	  (closes issue #19239) Reported by: byronclark Patches:
	  queue_announce_length.patch uploaded by byronclark (license 1200)

2011-05-05 19:09 +0000 [r317283]  Jonathan Rose <jrose@digium.com>

	* channels/chan_sip.c: Resolves a deadlock that occurs during
	  sip_new This is based on an uncommitted patch by jpeeler for the
	  issue. Instead of relocking and then unlocking the channel
	  though, we keep the lock on the channel until we are finished
	  doing what we need to the channel. (closes issue #18441) Reported
	  by: Alric

2011-05-05 18:39 +0000 [r317280-317281]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 317255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r317255 | russell | 2011-05-05 13:29:53 -0500
	  (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
	  | 15 lines chan_sip: fix broken realtime peer count, fix memory
	  leak This patch addresses two bugs in chan_sip: 1) The count of
	  realtime peers and users was off. The increment checked the value
	  of the caching option, while the decrement did not. 2) Add a
	  missing regfree() for a regex. (closes issue #19108) Reported by:
	  vrban Patches: missing_regfree.patch uploaded by vrban (license
	  756) sip_object_counter.patch uploaded by vrban (license 756)
	  ........ ................

	* /: Restore branch-1.6.2-merged and branch-1.6.2-blocked
	  properties.

2011-05-05 18:02 +0000 [r317196]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
	  they eventually go away when a peer abruptly disappears. This
	  mostly occurs after a successful registration. (closes issue
	  #17544) Reported by: marcelloceschia Patches: (modified)
	  tcptls.patch uploaded by st (license 907)

2011-05-05 15:04 +0000 [r317058-317104]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
	  | 8 lines Disable console colourization inside safe_asterisk
	  checks. (closes issue #19213) Reported by: lefoyer Patches:
	  issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
	  wdoekes (license 717) Tested by: wdoekes, lefoyer ........

	* Makefile, configs/cel.conf.sample: Remove unused directory and
	  clear up some documentation. (closes issue #19193) Reported by:
	  bchia Patches: cel-csv.diff uploaded by lathama (license 1028)
	  Tested by: lathama, Marquis42

2011-05-05 02:30 +0000 [r316917-316919]  Sean Bright <sean@malleable.com>

	* main/http.c: Use the correct HTTP method when generating our
	  digest, otherwise we always fail. When calculating the 'A2'
	  portion of our digest for verification, we need the HTTP method
	  that is currently in use. Unfortunately our mapping function was
	  incorrect, resulting in invalid hashes being generated and, in
	  turn, failures in authentication. (closes issue #18598) Reported
	  by: ksn

	* main/utils.c: Look at the correct buffer for our digest info
	  instead of an empty one. (issue #18598) Reported by: ksn

	* main/manager.c: Make sure that tcptls_session is properly
	  initialized. (issue #18598) Reported by: ksn

2011-05-04 20:50 +0000 [r316874]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooSocket.c: Fix trivial bug in ooSocket.c
	  codes Revert condition for result code of ast_gethostbyname
	  (closes issue #19185) Reported by: dswartz Patches:
	  issue19185-patch uploaded by may213 (license 454)

2011-05-04 18:51 +0000 [r316831]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_meetme.c: Wait for leader with Music On Hold allows
	  crosstalk between participants. Parenthesis in the wrong
	  position. Regression from issue #14365 when expanding conference
	  flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan
	  Tested by: rmudgett

2011-05-04 16:15 +0000 [r316663-316709]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c, /: Merged revisions 316708 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400
	  (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May
	  2011) | 8 lines If sox fails when processing a voicemail, don't
	  delete the original file. (closes issue #18111) Reported by:
	  sysreq Patches: issue18111_trunk.patch uploaded by seanbright
	  (license 71) Tested by: seanbright ........ ................

	* main/manager.c: Only return a single error via AMI when
	  requesting a forbidden action. (closes issue #19216) Reported by:
	  oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright
	  (license 71) Tested by: seanbright

2011-05-04 14:25 +0000 [r316617-316650]  David Vossel <dvossel@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 316644 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011)
	  | 9 lines Fixes one-way-audio when chanspy activated with the 'o'
	  option (closes issue #18382) Reported by: jkister Patches:
	  0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
	  uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
	  malin, wdoekes, boroda, dvossel ........

	* /, channels/chan_sip.c: Merged revisions 316616 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011)
	  | 12 lines Fixes session-timers=refuse not being enforced for
	  *caller* During handle_request_invite, the session timer mode was
	  retrieved from a cached variable. This patch forces a peer lookup
	  of the session timer mode in the case of an incoming invite.
	  (closes issue #18804) Reported by: wdoekes Patches:
	  issue18804_session_timer_refuse_caller.patch uploaded by wdoekes
	  (license 717) issue_18804_v2.diff uploaded by dvossel (license
	  671) ........

2011-05-04 02:34 +0000 [r316476]  Sean Bright <sean@malleable.com>

	* /, apps/app_meetme.c: Merged revisions 316475 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May
	  2011) | 10 lines Honor the C option to MeetMe when L is passed.
	  This fixes a case that r304773 and friends missed. (closes issue
	  #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
	  uploaded by var (license 1227) Tested by: seanbright ........

2011-05-04 00:12 +0000 [r316429]  Tilghman Lesher <tilghman@meg.abyt.es>

	* addons/cdr_mysql.c, addons/res_config_mysql.c: Escape column
	  names in case they contain illegal characters ('-') or reserved
	  words. (closes issue #19063) Reported by: festr Patches: patch
	  uploaded by festr (license 443)

2011-05-03 22:13 +0000 [r316336]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c, channels/chan_mgcp.c, channels/chan_skinny.c:
	  Use htons() instead of ntohs() in some places. (closes issue
	  #19200) Reported by: wdoekes Patches: issue19200-trunk.patch
	  uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded
	  by wdoekes (license 717)

2011-05-03 22:05 +0000 [r316334]  David Vossel <dvossel@digium.com>

	* main/channel.c: Fixes framehook segfault on indicate (closes
	  issue #19215) Reported by: irroot Patches:
	  framehook_indicate.patch uploaded by irroot (license 52)

2011-05-03 21:41 +0000 [r316331]  Russell Bryant <russell@digium.com>

	* apps/app_minivm.c: Resolve another warning.

2011-05-03 21:37 +0000 [r316330]  David Vossel <dvossel@digium.com>

	* channels/chan_local.c, /: Merged revisions 316329 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500
	  (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011)
	  | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ
	  for tracking down the issue and submitting the patch. (closes
	  issue #19053) Reported by: oej Tested by: oej Review:
	  https://reviewboard.asterisk.org/r/1158/ ........
	  ................

2011-05-03 19:55 +0000 [r316265]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c, apps/app_ices.c, apps/app_followme.c,
	  main/config.c, main/cdr.c, main/channel.c, channels/chan_phone.c,
	  funcs/func_enum.c, main/manager.c, channels/chan_skinny.c,
	  res/res_agi.c, main/plc.c, main/features.c, apps/app_minivm.c,
	  apps/app_amd.c, main/pbx.c, res/res_fax.c, formats/format_wav.c,
	  apps/app_festival.c, channels/chan_agent.c, apps/app_originate.c,
	  apps/app_queue.c, codecs/lpc10/dyptrk.c,
	  include/asterisk/linkedlists.h, main/audiohook.c,
	  pbx/pbx_config.c, main/asterisk.c, main/dsp.c,
	  res/res_calendar.c, apps/app_voicemail.c, main/udptl.c,
	  channels/chan_unistim.c, main/fskmodem_float.c,
	  main/rtp_engine.c: Fix a bunch of compiler warnings generated by
	  gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there
	  were a few others mixed in here, as well.

2011-05-03 19:18 +0000 [r316224]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/chan_dahdi.c, channels/sig_analog.c:
	  The dahdi_hangup() call does not clean up the channel fully.
	  After dahdi_hangup() has supposedly hungup an ISDN channel there
	  is still traffic on the S0-bus because the channel was not
	  cleaned up fully. Shuffled the hangup code to include some
	  missing cleanup. Also fixed some code formatting in the area. I
	  think the primary missing clean up code was the call to
	  tone_zone_play_tone() to turn off any active tones on the
	  channel. (closes issue #19188) Reported by: jg1234 Patches:
	  issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested
	  by: jg1234

2011-05-03 18:59 +0000 [r316215-316217]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Never put the Require: timer header in an
	  Invite. This has already been discussed and should have been
	  resolved earlier. View revsion 285565's log for more information
	  about why it is important to not put timer in the Require header.
	  (closes issue #18704) Reported by: mfrager

	* res/res_odbc.c: Fixes a random crash (NULL reference) in
	  res_odbc.c. (closes issue #19180) Reported by: pruiz Patches:
	  tmp.diff uploaded by pruiz (license 1152) Tested by: pruiz,
	  seanbright

2011-05-03 18:17 +0000 [r316206]  Sean Bright <sean@malleable.com>

	* main/manager.c: If we aren't interested in events, don't generate
	  the FullyBooted event on AMI login. (closes issue #19089)
	  Reported by: bklang Patches: issue19089-1.8-r316204.patch
	  uploaded by seanbright (license 71) Tested by: seanbright

2011-05-03 10:57 +0000 [r316193]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* autoconf/ast_check_pwlib.m4, configure: Re-fix bashism in
	  ./configure: s/let/$(( ))/ A forward-port in r278985 accidentally
	  re-introduced issue 17485. Fixing it. Thanks to Jilles Tjoelker
	  for the good report. (closes issue #17485)

2011-05-02 19:09 +0000 [r316094]  Tilghman Lesher <tilghman@meg.abyt.es>

	* funcs/func_curl.c, /: Merged revisions 316093 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011)
	  | 8 lines More possible crashes based upon invalid inputs.
	  (closes issue #18161) Reported by: wdoekes Patches:
	  20110301__issue18161.diff.txt uploaded by tilghman (license 14)
	  Tested by: wdoekes ........

2011-04-27 19:14 +0000 [r315894]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged
	  revisions 315893 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500
	  (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
	  2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
	  This change optimizes the free_via() function and removes some
	  redundant null checking. It also fixes compliance with RFC 3261
	  section 18.2.2 by always using the port specified in the Via
	  header for routing responses (even when maddr is not set). Also
	  the htons() function is now used when setting the port.
	  Additional documentation comments have been added in various
	  places to make the logic in the code clearer. (closes issue
	  #18951) Reported by: jmls Patches:
	  issue18951_set_proper_port_from_via.patch uploaded by wdoekes
	  (license 717) (modified) ........ ................

2011-04-27 15:55 +0000 [r315810]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Set the copyright year to 2011 in the startup
	  message.

2011-04-27 12:36 +0000 [r315765]  Leif Madsen <lmadsen@digium.com>

	* sounds/sounds.xml, sounds/Makefile: Enable Russian core sound
	  selection in menuselect. (closes issue #18724) Reported by:
	  pbxware

2011-04-26 22:56 +0000 [r315673]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 315672 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315672 | twilson | 2011-04-26 15:52:25 -0700
	  (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
	  | 11 lines Make sure unregistering a peer unlinks it from the
	  peer container Instead of mostly copying the code from
	  expire_register, just use the function that "does the right
	  thing". (closes issue #16033) Reported by: kkm Patches:
	  016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
	  Tested by: kkm, tilghman, twilson ........ ................

2011-04-26 22:14 +0000 [r315645]  Richard Mudgett <rmudgett@digium.com>

	* main/pbx.c: The 'e' special extension fails to trigger in at
	  least two cases. The 'e' extension is a fall back for the 'i',
	  't', or 'T' extensions if any of them do not exist. Many of the
	  places the 'e' extension was supposed to be invoked fail because
	  the priority was set wrong. There were two places where the 'e'
	  extension was not even checked for fall back. * Made invoke the
	  'e' extension similarly to the previous 'i', 't', or 'T'
	  extension check and added the 'e' extension as a fall back to the
	  two missing locations. * Prioritized and optimized some hangup
	  tests associated with the 'e' extension. (closes issue #19136)
	  Reported by: kshumard Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1196/

2011-04-26 21:39 +0000 [r315644]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged
	  revisions 315643 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315643 | twilson | 2011-04-26 14:27:44 -0700
	  (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
	  | 18 lines Allow transfer loops without allowing forwarding loops
	  We try to avoid the situation where two phones may be forwarded
	  to each other causing an infinite loop by storing each dialed
	  interface in a channel datastore and checking the list before
	  dialing out. This works, but currently breaks situations like A
	  calls B, A transfers B to C, B transfers C to A, and A transfers
	  C to B. Since human interaction is happening here and not an
	  automated forwarding loop, it should be allowed. This patch
	  removes the dialed_interfaces datastore when a call is bridged (a
	  suggestion from the brilliant mmichelson). If a call is being
	  bridged, it should be safe to assume that we aren't stuck in a
	  loop. Since we are now handling this is the bridge code, the
	  previous attempts at handling it in app_dial and app_queue are
	  removed. Review: https://reviewboard.asterisk.org/r/1195/
	  ........ ................

2011-04-26 19:32 +0000 [r315503]  Tilghman Lesher <tilghman@meg.abyt.es>

	* include/asterisk/select.h, /: Merged revisions 315502 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500
	  (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
	  | 14 lines Fix the bounds-checking code. The code that set the
	  bit within the select bitfield was correct, but the
	  bounds-checking code was not. The change to that line uses the
	  new _bitsize macro for clarity. Also, FD_ZERO macro did not
	  zero-out anything but the first word of the bitfield, so this
	  could have caused problems with modules using that macro with the
	  expanded bitfield. (closes issue #18773) Reported by: jamicque
	  Patches: 20110423__issue18773.diff.txt uploaded by tilghman
	  (license 14) Tested by: chris-mac ........ ................

2011-04-26 18:00 +0000 [r315452]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Add missing set of name valid flag when dialing.

2011-04-26 17:40 +0000 [r315446]  Russell Bryant <russell@digium.com>

	* channels/chan_local.c: chan_local: resolve a deadlock. This patch
	  resolves a fairly complex deadlock that can occur with the
	  combination of chan_local and a dialplan switch, such as dynamic
	  realtime extensions, which pulls autoservice into the picture
	  when doing a dialplan lookup. (closes issue #18818) Reported by:
	  nic Patches: issue18818.patch uploaded by jthurman (license 614)
	  18818.v1.txt uploaded by russell (license 2) Tested by: nic,
	  jthurman, kterzi, steve-howes, sysreq, IshMalik

2011-04-26 02:18 +0000 [r315394]  Paul Belanger <pabelanger@digium.com>

	* pbx/pbx_config.c, /: Merged revisions 315393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr
	  2011) | 7 lines Add back CLI command 'dialplan save' (closes
	  issue #19140) Reported by: lmadsen Patches:
	  __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
	  10) ........

2011-04-25 21:49 +0000 [r315349]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_mgcp.c: When using MGCP realtime gateway
	  definitions, random crashes occur. Fixed incorrect linked list
	  node removal for realtime gateways. (closes issue #18291)
	  Reported by: nahuelgreco Patches:
	  dangling-pointers-when-pruning.patch uploaded by nahuelgreco
	  (license 162)

2011-04-25 19:37 +0000 [r315213-315259]  Russell Bryant <russell@digium.com>

	* /, formats/format_wav.c: Merged revisions 315258 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315258 | russell | 2011-04-25 14:31:44 -0500
	  (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011)
	  | 10 lines Be more flexible with unknown chunks in wav files.
	  This patch makes format_wav ignore unknown chunks instead of
	  erroring out on them. (closes issue #18306) Reported by: jhirsch
	  Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
	  (license 1156) ........ ................

	* /, channels/chan_sip.c: Merged revisions 315212 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011)
	  | 7 lines Don't link non-cached realtime peers into the
	  peers_by_ip container. (closes issue #18924) Reported by: wdoekes
	  Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch
	  uploaded by wdoekes (license 717) ........

2011-04-25 07:14 +0000 [r315053]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_local.c, /: Merged revisions 315052 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200
	  (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr
	  2011) | 11 lines chan_local:check_bridge() misplaced misplaced
	  ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't
	  followed, brigde remains locked. (closes issue #19176) Reported
	  by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis
	  (license 585) ........ ................

2011-04-22 22:59 +0000 [r315001]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_dahdi.c: chan_dahdi: Can't return to normal ring
	  after distinctive ring on FXS clear a previous distinctivering
	  pattern before each new call (closes issue #18985) Reported by:
	  bromont Patches: bug18985.diff.txt uploaded by alecdavis (license
	  585) Tested by: alecdavis, bromont

2011-04-22 21:20 +0000 [r314959]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 314958 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500
	  (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
	  2011) | 4 lines Don't hold the pvt lock while streaming a file.
	  ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
	  -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
	  the app threads from using the asterisk channel at the same time.
	  ABE-2756 ........ ................

2011-04-22 14:02 +0000 [r314780]  Russell Bryant <russell@digium.com>

	* /, res/res_agi.c: Merged revisions 314778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011)
	  | 11 lines Initialize buffers in getvar and getvarfull.
	  Initialize the buffers used to hold the result from GET VARIABLE
	  or GET VARIABLE FULL. The bug report shows func_read returning
	  garbage in the result. It assumed that the buffer passed in was
	  initialized, like many other functions do. In the more common
	  code path (through the dialplan), it is initialized, so just
	  initialize it here too. (closes issue #19050) Reported by: johnz
	  ........

2011-04-22 13:59 +0000 [r314779]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* res/res_fax_spandsp.c, channels/chan_unistim.c: Fix a few typos
	  (shown by Lintian)

2011-04-22 13:35 +0000 [r314777]  Russell Bryant <russell@digium.com>

	* /: Recorded merge of revisions 314776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r314776 | russell | 2011-04-22 08:35:22 -0500 (Fri, 22 Apr 2011)
	  | 10 lines Fix handling of some call parking config options. This
	  patch adjusts the handling of some call parking config options to
	  fix some issues that have already been addressed in 1.8 and
	  trunk. (closes issue #19167) Reported by: bluecrow76 Patches:
	  asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
	  uploaded by bluecrow76 (license 270) ........

2011-04-21 22:38 +0000 [r314732]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Correct DAHDIShowChannels XML
	  documentation.

2011-04-21 18:24 +0000 [r314628]  Matthew Nicholson <mnicholson@digium.com>

	* configs/sip.conf.sample, configs/skinny.conf.sample,
	  channels/sip/include/sip.h, configs/http.conf.sample,
	  main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
	  main/http.c: Merged revisions 314620 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500
	  (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
	  2011) | 14 lines Added limits to the number of unauthenticated
	  sessions TCP based protocols are allowed to have open
	  simultaneously. Also added timeouts for unauthenticated sessions
	  where it made sense to do so. Unrelated, the manager interface
	  now properly checks if the user has the "system" privilege before
	  executing shell commands via the Originate action. AST-2011-005
	  AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
	  issue #18996) Reported by: tzafrir ........ ................

2011-04-21 00:23 +0000 [r314550]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 314549 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011)
	  | 6 lines Don't allocate more space than necessary for a sip_pkt
	  This extra allocation is a hold-over from when pkt->data was a
	  character array. Now that it is an allocated string, just
	  allocate enough for the sip_pkt. ........

2011-04-20 16:54 +0000 [r314417]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/frame.h: AST_CONTROL_XXX comment changes.

2011-04-20 05:25 +0000 [r314358]  Terry Wilson <twilson@digium.com>

	* main/lock.c: Initialize track pointer ast_reentrancy_init checks
	  to see if it is NULL before initializing with calloc

2011-04-19 15:42 +0000 [r314203-314251]  Leif Madsen <lmadsen@digium.com>

	* main/tcptls.c: Use SSLv23_client_method instead of old SSLv2
	  only. (closes issue #19095) (closes issue #19138) Reported by:
	  tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46)
	  Tested by: russell, chazzam

	* /, funcs/func_channel.c: Merged revisions 314205 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19
	  Apr 2011) | 6 lines Remove duplicate documentation from
	  func_channel.c (closes issue #18970) Reported by: IgorG Patches:
	  func_channel.c.doc.diff uploaded by IgorG (license 20) ........

	* apps/app_dial.c, /: Merged revisions 314202 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011)
	  | 7 lines Update seconds to milliseconds in ast_verb output.
	  (closes issue #19084) Reported by: smurfix Patches:
	  app_dial.patch uploaded by smurfix (license 547) Tested by:
	  lmadsen, smurfix ........

2011-04-18 16:10 +0000 [r314068-314069]  Richard Mudgett <rmudgett@digium.com>

	* res/res_agi.c: The AsyncAGI command loop is lax in the value it
	  returns for the return status. * Return correct status:
	  SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the
	  command loop such as hangup would return SUCCESS. * The "asyncagi
	  break" command now returns SUCCESS and is now the only way to
	  break the command loop with that status. Previously, it returned
	  FAILED. * The AMI event AsyncAGI End is no longer sent if the
	  AsyncAGI Start event is not sent. Previously, this happened
	  because of an error setting up the AGI pipes. * All executed AGI
	  commands now get an AsyncAGI Exec result event. Previously, if
	  the command returned failure (because of hangup), the command
	  loop just exited with FAILURE and did not send the AsyncAGI Exec
	  result event. * Makes sure that the channel frame queue is empty
	  on hangup. Review: https://reviewboard.asterisk.org/r/1183/

	* apps/app_dial.c: Unclear code in app_dial.c. Make code formatting
	  clear. (closes issue #19134) Reported by: oej

2011-04-18 15:23 +0000 [r314017-314067]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Remove the need for deadlock avoidance in
	  chan_sip do_monitor. Deadlock avoidance between the sip pvt and
	  the pvt->owner is very difficult. Now that channel's are ao2
	  objects, this complication is no longer necessary. It turns out
	  the pvt's msg queue only exists because of deadlock avoidance
	  (when deadlock avoidance fails msgs were added to a queue to be
	  processed later), so this goes away as well. The technique used
	  in the new sip_lock_pvt_full() function should be used as a
	  template for replacing all locations where deadlock avoidance
	  occurs between a channel tech_pvt and the pvt's owner. My hope is
	  that this will begin a reversal of the invalid channel driver
	  locking architecture we have been using for so long. This patch
	  also resolves an issue where the pvt->owner gets unlocked during
	  processing the msg queue. (closes issue #18690) Reported by:
	  dvossel Review: https://reviewboard.asterisk.org/r/1182/

	* include/asterisk/rtp_engine.h, main/rtp_engine.c,
	  channels/chan_sip.c: sip codec negotiation of dynamic rtp
	  payloads error fix This patch fixes how chan_sip handles dynamic
	  rtp payload types it does not understand. At the moment if a
	  dynamic payload's mime type does not match one we understand, the
	  payload does not get removed from our payload table. As a result
	  of this, the payload is set to whatever dynamic codec we use
	  internally for that payload number on outgoing INVITES. This is
	  incorrect. This patch fixes this by properly checking the rtpmap
	  set function's return code to make sure it was found. The
	  function can return both -1 and -2 depending on the source of the
	  mismatch. We were just checking -1 explicitly. Review:
	  https://reviewboard.asterisk.org/r/1169/

2011-04-15 15:08 +0000 [r313860]  Jonathan Rose <jrose@digium.com>

	* main/cli.c, /: Merged revisions 313859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) |
	  10 lines Fix a Tab Completion bug that occurs due to multiple
	  matches on a substring. Makes word_match function in cli.c repeat
	  a search for a command string until a proper match is found or
	  the string is searched to the last point. (closes issue #17494)
	  Reported by: ffossard Review:
	  https://reviewboard.asterisk.org/r/1180/ ........

2011-04-14 20:59 +0000 [r313517-313780]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Leftover debug messages unconditionally
	  sent to the console. Executing Dial(DAHDI/1/18475551212,300,)
	  with the echotraining config option enabled outputs the following
	  debug messages unconditionally: Dialing T1847555121 on 1 Dialing
	  www2w on 1 * Made debug messages in my_dial_digits() normal debug
	  messages that do not get output unless enabled. * Reworded some
	  debug messages in my_dial_digits() to be clearer. * Replace
	  strncpy() with ast_copy_string() in my_dial_digits() which does
	  the same job better. (closes issue #18847) Reported by:
	  vmikhelson Tested by: rmudgett

	* res/res_agi.c: Revert flushing stale AsyncAGI commands from
	  -r313615. It looks like it was intentional to leave any commands
	  or in-flight commands in the queue in case Async AGI is run again
	  on the call.

	* res/res_agi.c: Miscellaneous AGI diagnostic message cleanup and
	  code optimization.

	* res/res_agi.c: * Add missing channel lock to
	  handle_cli_agi_add_cmd(). * Flush any Async AGI commands left
	  over from earlier Async AGI control of the call.

	* main/channel.c, /, res/res_agi.c: Merged revisions 313579 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500
	  (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
	  | 41 lines Asterisk does not hangup a channel after endpoint
	  hangs up. If the call that the dialplan started an AGI script for
	  is hungup while the AGI script is in the middle of a command then
	  the AGI script is not notified of the hangup. There are many AGI
	  Exec commands that this can happen with. The reported
	  applications have been: Background, Wait, Read, and Dial. Also
	  the AGI Get Data command. * Don't wait on the Asterisk channel
	  after it has hung up. The channel is likely to never need
	  servicing again. * Restored the AGI script's ability to return
	  the AGI_RESULT_HANGUP value in run_agi(). It previously only
	  could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
	  DeadAGI and AGI applications were merged. (closes issue #17954)
	  Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
	  rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
	  rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
	  (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
	  #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
	  (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
	  rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
	  Tested by: rmudgett JIRA SWP-2727 Review:
	  https://reviewboard.asterisk.org/r/1165/ ........
	  ................

	* apps/app_dumpchan.c: Bring the dumpchan application inline with
	  "core show channel". * Added fields that are in "core show
	  channel" to dumpchan output. * Fixed reuse of formatbuf before
	  the previous string stored there was used by snprintf. All output
	  strings now have their own buffer. * Adjusted the buffer sizes to
	  not be so abusive of the stack now that there are more buffers.
	  Change requested by oej.

2011-04-12 18:47 +0000 [r313434-313436]  Jonathan Rose <jrose@digium.com>

	* channels/chan_dahdi.c, /: fixing stupid mistake with putting code
	  before variable declaration ........ Merged revisions 313435 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
	  14 lines reload Chan_dahdi memory leak caused by variables
	  chan_dahdi reloading with variables set via setvar in
	  chan_dahdi.conf would stay in the dahdi_pvt structs for
	  individual channels (causing them to just continue adding the new
	  ones to the list) and also there was a memory leak causes by the
	  conf objects. This patch resolves both of these by using
	  ast_variables_destroy during the loading process. (closes issue
	  #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
	  jrose (license 1225) Tested by: tilghman, jrose Review:
	  https://reviewboard.asterisk.org/r/1170/ ........ ........

	* channels/chan_dahdi.c, /: Merged revisions 313432 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr
	  2011) | 14 lines reload Chan_dahdi memory leak caused by
	  variables chan_dahdi reloading with variables set via setvar in
	  chan_dahdi.conf would stay in the dahdi_pvt structs for
	  individual channels (causing them to just continue adding the new
	  ones to the list) and also there was a memory leak causes by the
	  conf objects. This patch resolves both of these by using
	  ast_variables_destroy during the loading process. (closes issue
	  #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
	  jrose (license 1225) Tested by: tilghman, jrose Review:
	  https://reviewboard.asterisk.org/r/1170/ ........

2011-04-11 23:08 +0000 [r313366-313369]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Frames from the inbound channel should go to all
	  outbound channels in app_dial.c. In app_dial.c:wait_for_answer()
	  frames from the inbound channel should be sent to all outbound
	  channels instead of only if there is just one outbound channel.
	  Control frames like AST_CONTROL_CONNECTED_LINE need to be passed
	  to all of the the outbound channels. This can happen if a blond
	  transfer is done by a remote switch on the inbound channel. JIRA
	  AST-443 JIRA SWP-2730

	* apps/app_dial.c: Backport a restructuring change from trunk to
	  make the next change stand out.

	* main/cli.c: Added "Connected Line ID" and "Connected Line ID
	  Name" to "core show channel" output.

2011-04-11 19:36 +0000 [r313279]  Leif Madsen <lmadsen@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 313278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500
	  (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011)
	  | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093)
	  Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by
	  tzafrir (license 46) ........ ................

2011-04-11 15:40 +0000 [r313190]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  313189 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500
	  (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011)
	  | 25 lines Stuck channel using FEATD_MF if caller hangs up at the
	  right time. The cause was actually a caller hanging up just at
	  the end of the Feature Group D DTMF tones that setup the call.
	  The reason for this is a "guard timer" that's implemented using
	  ast_safe_sleep(100). If the caller happens to hang up AFTER the
	  final tone of the DTMF string but BEFORE the end of that
	  ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
	  This causes the code to bounce to the end of ss_thread(), but it
	  does NOT tear down the call properly. This should be a rare
	  occurrence because the caller has to hang up at EXACTLY the right
	  time. Nonetheless, it was happening quite regularly on the
	  reporter's system. It's not easily reproducible, unless you
	  purposely increase the guard-time to 2000 or more. Once you do
	  that, you can reproduce it every time by watching the DTMF debug
	  and hanging up just as it ends. Simply add an ast_hangup() before
	  goto quit. (closes issue #15671) Reported by: jcromes Patches:
	  issue15671.patch uploaded by pabelanger (license 224) Tested by:
	  jcromes ........ ................

2011-04-09 20:56 +0000 [r313142]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: fix trivial bug in ooh323_indicate on
	  AST_CONTROL_SRC... check p->rtp is not null

2011-04-07 13:35 +0000 [r313048]  Jonathan Rose <jrose@digium.com>

	* /, main/features.c: Merged revisions 313047 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) |
	  9 lines Makes parking lots clear and rebuild properly when
	  features reload is invoked from CLI Before, default parkinglot in
	  context parkedcalls with ext 700 would always be present and when
	  reload was invoked, the previous parkinglots would not be
	  cleared. (closes issue #18801) Reported by: mickecarlsson Review:
	  https://reviewboard.asterisk.org/r/1161/ ........

2011-04-07 10:24 +0000 [r313001-313002]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_voicemail.c: app_voicemail: close_mailbox change
	  LOG_WARNING to LOG_NOTICE

	* channels/sig_pri.c: Fix ISDN calling subaddr User Specified
	  Odd/Even Flag Calculation of the Odd/Even flag was wrong.
	  Implement correct algo, and set odd/even=0 if data would be
	  truncated. Only allow automatic calculation of the O/E flag,
	  don't let dialplan influence. (closes issue #19062) Reported by:
	  festr Patches: bug19062.diff2.txt uploaded by alecdavis (license
	  585) Tested by: festr, alecdavis, rmudgett

2011-04-05 18:45 +0000 [r312866-312949]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  Crash if ISDN span layer 1 is down on initial load. Regression
	  from -r312575 B channel shifting during negotiation. * Also
	  combine updating the alarm flag with clearing the resetting flag.

	* channels/chan_sip.c: Add 416 response to OPTIONS packet. RFC3261
	  Section 11.2 says the response code to an OPTIONS packet needs to
	  be the same as if it were an INVITE.

	* channels/chan_sip.c: Responding to OPTIONS packet with 404
	  because Asterisk not looking for "s" extension. The
	  get_destination() function was not using the "s" extension when
	  the request URI did not specify an extension. This is a
	  regression caused when the URI parsing code was extracted into
	  parse_uri(). Made get_destination() substitute the "s" extension
	  when the parsed URI results in an empty string. (closes issue
	  #18348) Reported by: shmaize Patches: issue18348_v1.8.patch
	  uploaded by rmudgett (license 664) Tested by: shmaize

2011-04-05 14:14 +0000 [r312766]  Matthew Nicholson <mnicholson@digium.com>

	* configs/manager.conf.sample, main/manager.c, /: Merged revisions
	  312764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500
	  (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr
	  2011) | 8 lines Limit the number of unauthenticated manager
	  sessions and also limit the time they have to authenticate.
	  AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested
	  by: mnicholson ........ ................

2011-04-05 14:13 +0000 [r312765]  Jonathan Rose <jrose@digium.com>

	* /, apps/app_meetme.c: Merged revisions 312762 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r312762 | jrose | 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) |
	  1 line Backporting trunk change to add verbosity to 'L' option in
	  meetme ........

2011-04-04 16:10 +0000 [r312575]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c, /:
	  Merged revisions 312574 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500
	  (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011)
	  | 38 lines Issues with ISDN calls changing B channels during call
	  negotiations. The handling of the PROCEEDING message was not
	  using the correct call structure if the B channel was changed.
	  (The same for PROGRESS.) The call was also not hungup if the new
	  B channel is not provisioned or is busy. * Made all call
	  connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS,
	  ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
	  using the correct structure and B channel. If there is any
	  problem with the operations then the call is now hungup with an
	  appropriate cause code. * Made miscellaneous messages
	  (INFORMATION, FACILITY, NOTIFY) find the correct structure by
	  looking for the call and not using the channel ID. NOTIFY is an
	  exception with versions of libpri before v1.4.11 because a call
	  pointer is not available for Asterisk to use. * Made all hangup
	  messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct
	  structure by looking for the call and not using the channel ID.
	  (closes issue #18313) Reported by: destiny6628 Tested by:
	  rmudgett JIRA SWP-2620 (closes issue #18231) Reported by:
	  destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue
	  #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The
	  issues fixed here are most likely causing this JIRA issue.) JIRA
	  DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
	  ........ ................

2011-04-01 23:15 +0000 [r312461-312509]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: When a call going out an NT-PTMP port gets
	  rejected, Asterisk crashes. If a call is sent to an ISDN phone
	  that rejects the call with RELEASE_COMPLETE(cause: call
	  reject(21), or busy(17)) Asterisk crashes. I could not get my
	  setup to crash. However, I could see the possibility from a race
	  condition between queuing an AST_CONTROL_BUSY to the core and
	  then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is
	  processed before the AST_CONTROL_HANGUP is queued, the
	  ast_channel could be destroyed out from under chan_misdn. Avoid
	  this particular crash scenario by not queueing the
	  AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes
	  issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch
	  uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy
	  JIRA SWP-2679

	* main/ccss.c: CallCompletionRequest()/CallCompletionCancel() exit
	  non-zero if fail. The
	  CallCompletionRequest()/CallCompletionCancel() dialplan
	  applications exit nonzero on normal failure conditions. The
	  nonzero exit causes the dialplan to hangup immediately. The
	  dialplan author has no opportunity to report success/failure to
	  the user. * Made always return zero so the dialplan can continue.
	  * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
	  CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.
	  Also documented the values set. * Reduced the warning about no
	  core instance in CallCompletionCancel() to a debug message. It is
	  a normal event and should not be output at the WARNING level.
	  (closes issue #18763) Reported by: p_lindheimer Patches:
	  ccss.patch uploaded by p lindheimer (license 558) Modified Tested
	  by: p_lindheimer, rmudgett JIRA SWP-3042

2011-04-01 10:58 +0000 [r312286-312288]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/asterisk.c, include/asterisk/select.h, /: Merged revisions
	  312287 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500
	  (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011)
	  | 7 lines Found some leaking file descriptors while looking at
	  ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej
	  Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman
	  (license 14) ........ ................

	* addons/cdr_mysql.c: Reload must react correctly against a
	  possibly changed table, so dropping the conditional reload flag.

2011-04-01 09:03 +0000 [r312117-312211]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_voicemail.c, /: Merged revisions 312210 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300
	  (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr
	  2011) | 23 lines voicemail: get real last_message_index and
	  count_messages, ODBC resequence change last_message_index to read
	  the max msgnum stored in the database change count_messages to
	  actually count the number of messages. last_message_index change:
	  This fixed overwriting of the last message if msgnum=0 was
	  missing. Previously every incoming message would overwrite
	  msgnum=1. count_messages change: allows us to detect when
	  requencing is required in opneA_mailbox. resequence enabled for
	  ODBC storage: Assists with fixing up corrupt databases with gaps,
	  but only when a user actively opens there mailboxes. (closes
	  issue #18692,#18582,#19032) Reported by: elguero Patches: based
	  on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
	  37) Tested by: elguero, nivek, alecdavis Review:
	  https://reviewboard.asterisk.org/r/1153/ ........
	  ................

	* apps/app_voicemail.c, /: Merged revisions 312103 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300
	  (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr
	  2011) | 16 lines app_voicemail: close_mailbox needs to respect
	  additional messages while mailbox is open. close_mailbox leave
	  gaps in message sequence if messages are deleted and new messages
	  arrive during this time, this is because the shuffle down to slot
	  0, only shuffles the number of pre-existing messages when mailbox
	  is opened, ignoring new arrivals. Fix: in close_mailbox
	  re-evaluate number of messages before the shuffle, this then
	  includes new arrivals. Happens on filebased or ODBC storage.
	  (issues #19032,#18582,#18692,#18998) Reported by:
	  alecdavis,tootai,afosorio Review:
	  https://reviewboard.asterisk.org/r/1153/ ........
	  ................

2011-03-31 20:11 +0000 [r312022]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: chan_misdn segfaults when DEBUG_THREADS is
	  enabled. The segfault happens because jb->mutexjb is
	  uninitialized from the ast_malloc(). The internals of
	  ast_mutex_init() were assuming a nonzero value meant mutex
	  tracking initialization had already happened. Recent changes to
	  mutex tracking code to reduce excessive memory consumption
	  exposed this uninitialized value. Converted misdn_jb_init() to
	  use ast_calloc() instead of ast_malloc(). Also eliminated
	  redundant zero initialization code in the routine. (closes issue
	  #18975) Reported by: irroot

2011-03-31 06:43 +0000 [r311930]  Tilghman Lesher <tilghman@meg.abyt.es>

	* configs/cdr_mysql.conf.sample: Incorrect default example; the
	  field is actually internally named "clid", not "callerid".
	  (closes issue #19040) Reported by: wcselby Tested by: tilghman

2011-03-30 01:56 +0000 [r311874]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Update some setup_dahdi_int() comments.

2011-03-29 07:08 +0000 [r311799]  Tilghman Lesher <tilghman@meg.abyt.es>

	* cel/cel_odbc.c: Remove extraneous check from integer-type fields.
	  (closes issue #19027) Reported by: mlehner Review:
	  https://reviewboard.asterisk.org/r/1149/

2011-03-28 22:00 +0000 [r311751]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Cross-reference VoiceMail() and
	  VoiceMailMain() in the xml docs.

2011-03-27 21:47 +0000 [r311687]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: correct return values in ooh323_indicate
	  for AST_CONTROL_T38_PARAMETERS

2011-03-23 21:54 +0000 [r311612-311615]  Brett Bryant <bbryant@digium.com>

	* apps/app_meetme.c: This patch fixes a bug with MeetMe behavior
	  where the 'P' option for always prompting for a pin is ignored
	  for the first caller. (closes issue #18070) Reported by: mav3rick
	  Review: https://reviewboard.asterisk.org/r/1132/

	* channels/sip/reqresp_parser.c: Fix a possible crash in
	  sip/reqresp_parser.c that is caused by a possible null value.
	  (closes issue #18821) Reported by: cmaj Patches:
	  patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
	  uploaded by cmaj (license 830)

2011-03-23 02:24 +0000 [r311558]  Terry Wilson <twilson@digium.com>

	* channels/sip/reqresp_parser.c: Don't use static declared buf in
	  parse_name_andor_addr This function isn't used anywhere yet, but
	  we definitely don't want to keep the same value for buf between
	  calls to the function.

2011-03-22 15:25 +0000 [r311497]  David Vossel <dvossel@digium.com>

	* /, apps/app_meetme.c: Merged revisions 311496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011)
	  | 2 lines Fixes memory leak in MeetMe AMI action ........

2011-03-18 16:19 +0000 [r311352]  Jonathan Rose <jrose@digium.com>

	* res/res_jabber.c, channels/chan_sip.c, res/res_fax.c: Changes
	  some print statements/events to use a blank string in place of
	  NULL if the string in question is NULL. This is supposed to
	  improve Solaris compatibility since Solaris goes berserk when
	  trying to output NULL strings. (closes issue #18759) Reported by:
	  bklang Patches: null-strings.patch uploaded by bklang (license
	  919)

2011-03-18 16:02 +0000 [r311342]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Properly populate the LOCALSTATIONID channel
	  variable.

2011-03-18 02:59 +0000 [r311295-311297]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Race condition when ISDN
	  CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY
	  could sometimes be processed before the call_forward dial string
	  is recognized. * Moved setting the call_forwarding dial string
	  after sending a response to the initiator and just queue an empty
	  frame to wake up the media thread instead of an AST_CONTROL_BUSY.
	  * Added check for empty rerouting/deflection number and respond
	  with an error.

	* apps/app_dial.c: Merged revision 310986 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed,
	  16 Mar 2011) | 28 lines Dial() o option broke when connected line
	  feature added. The patch restores the o option behavior and adds
	  the ability to specify the CallerID. The Dial o and f options are
	  complementary to each other. The o option stores the CallerID on
	  the outgoing channel as the channel's CallerID. The f option
	  forces the CallerID sent by the outgoing channel. o(x) - The
	  argument 'x' is optional. If not present, then specify that the
	  CallerID that was present on the *calling* channel be stored as
	  the CallerID on the *called* channel. This was the behavior of
	  Asterisk 1.0 and earlier. If present, then specify the CallerID
	  stored on the *called* channel. Note that o(${CALLERID(all)}) is
	  similar to option o without parameters. f(x) - The argument 'x'
	  is optional and its presence changes the behavior of this option.
	  If not present, then force the outgoing CallerID on a
	  call-forward or deflection to the dialplan extension for this
	  Dial() using a dialplan 'hint'. For example, some PSTNs do not
	  allow CallerID to be set to anything other than the numbers
	  assigned to you. If present, then force the outgoing CallerID to
	  'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by
	  rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA
	  SWP-3096 ..........

2011-03-17 19:03 +0000 [r311197]  Jonathan Rose <jrose@digium.com>

	* apps/app_chanspy.c: This fixes a nasty chanspy bug which was
	  causing a channel leak every time a spied on channel made a call.
	  In addition to the above, it makes certain channel destruction
	  occurs so that applications don't get stuck waiting for datastore
	  destruction while monitored by chanspy. (closes issue #18742)
	  Reported by: jkister Tested by: jkister, jcovert, jrose Review:
	  http://reviewboard.digium.internal/r/106/

2011-03-17 15:00 +0000 [r311141]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c, /: Merged revisions 311140 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar
	  2011) | 4 lines Don't write items to the manager socket twice.
	  AST-2011-003 (closes issue 0018987) Reported by: ks-steven
	  ........

2011-03-17 10:49 +0000 [r311050]  Alec L Davis <sivad.a@paradise.net.nz>

	* /, configs/indications.conf.sample: Merged revisions 311049 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300
	  (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar
	  2011) | 12 lines Remove extra quote in indications.conf Picking
	  low hanging fruit. (closes issue #18971) Reported by: IgorG
	  Patches: based on indications.conf.sample.diff uploaded by IgorG
	  (license 20) Tested by: IgorG ........ ................

2011-03-16 19:47 +0000 [r310902-310999]  Terry Wilson <twilson@digium.com>

	* main/tcptls.c, /: Merged revisions 310998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011)
	  | 11 lines Fix crash on fdopen failure See security advisory
	  AST-2011-004 (closes issue #18845) Reported by: cmaj Patches:
	  patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
	  uploaded by cmaj (license 830)
	  patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
	  uploaded by cmaj (license 830) Tested by: cmaj, twilson ........

	* main/manager.c, /: Merged revisions 310992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011)
	  | 4 lines Don't keep trying to write to a closed connection See
	  security advisory AST-2011-003. ........

	* /, main/features.c: Merged revisions 310889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r310889 | twilson | 2011-03-16 12:03:27 -0500
	  (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011)
	  | 29 lines Don't delay DTMF in core bridge while listening for
	  DTMF features This patch is mostly the work of Olle Johansson. I
	  did some cleanup and added the silence generating code if
	  transmit_silence is set. When a channel listens for DTMF in the
	  core bridge, the outbound DTMF is not sent until we have received
	  DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds
	  of DTMF to Asterisk, which sends no audio for those 4 seconds.
	  Some products see this delay and the time skew on RTP packets
	  that results and start ignoring the audio that is sent afterward.
	  With this change, the DTMF_BEGIN frame is inspected and checked.
	  If it matches a feature code, we wait for DTMF_END and activate
	  the feature as before. If transmit_silence=yes in asterisk.conf,
	  silence is sent if we paritally match a multi-digit feature. If
	  it doesn't match a feature, the frame is forwarded along with the
	  DTMF_END without delay. By doing it this way, DTMF is not
	  delayed. (closes issue #15642) Reported by: jasonshugart Patches:
	  issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license
	  396) Tested by: globalnetinc, jde (closes issue #16625) Reported
	  by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/
	  Review: https://reviewboard.asterisk.org/r/1125/ ........
	  ................

2011-03-15 01:48 +0000 [r310834]  Tilghman Lesher <tilghman@meg.abyt.es>

	* addons/chan_ooh323.c: Fix branch compile.

2011-03-15 01:00 +0000 [r310781]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/utils.c: core show locks: display ThreadID in hexadecimal
	  Allow easier cross referencing of thread ID's with GDB backtraces
	  (closes issue #18968) Reported by: alecdavis Patches:
	  bug18968.diff.txt uploaded by alecdavis (license 585)

2011-03-14 21:45 +0000 [r310734]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c,
	  addons/ooh323c/src/ooCalls.h: Introduce t.38 parameters control
	  functionality not full but enough for Send/RcvFax support
	  Introduce t.38 controls between asterisk core and channel/proto
	  layers. Not all parameters are transferred from proto layers but
	  *Fax apps tested and work ok. (issue #18693) Reported by:
	  benngard2 Patches: issue-18693.patch uploaded by may213 (license
	  454)

2011-03-14 21:30 +0000 [r310726-310733]  Jonathan Rose <jrose@digium.com>

	* main/channel.c: Undoes 310726 for further analysis

	* main/channel.c: Moves data store destruction from channel
	  destruction to hangup in channel.c This moves the data store
	  destruction and app signaling events for a call to ast_hangup so
	  that threads which wait for data store destruction don't become
	  stuck forever when attached to an application/function/etc that
	  keeps the channel open. (closes issue #18742) Reported by:
	  jkister Patches: patch.diff uploaded by jrose (license 1225)
	  Tested by: jkister, jcovert, jrose Review:
	  https://reviewboard.asterisk.org/r/1136/

2011-03-14 16:50 +0000 [r310636]  Richard Mudgett <rmudgett@digium.com>

	* /, main/callerid.c: Merged revisions 310635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500
	  (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011)
	  | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and
	  TDM410 The last character in the caller id message is getting a
	  framing error. The checksum is the last character in the message.
	  A framing error in the checksum could be because: 1) The sender
	  did not send a full stop bit. 2) The sender cut off the FSK
	  carrier too soon. 3) The sender opted to send zero of the
	  specified zero to 10 trailing mark bits and round-off errors in
	  the code resulted in the code not being where it thought it was
	  in the demodulated bit stream. Bit 8 of 'b' is set when parity
	  error. Bit 9 of 'b' is set when framing error. Made ignore the
	  framing and parity error bits if the errored character is the
	  checksum. We can tolerate a framing/parity error there. The
	  checksum character validates the message. (closes issue #18474)
	  Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
	  (license 636) (with modifications) Tested by: nivek ........
	  ................

2011-03-14 15:27 +0000 [r310587]  Jonathan Rose <jrose@digium.com>

	* /, funcs/func_volume.c: Merged revisions 310585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) |
	  8 lines Adds 'p' as an option to func_volume. When it is on, the
	  old behavior with DTMF controlling volume adjustment will be
	  enforced. When it is off, DTMF will not be processed by the
	  function. Programmed by Jonathan Rose Reviewed by David Vossel,
	  Leif Madsen, and Russell Bryant
	  http://reviewboard.digium.internal/r/93/ ........

2011-03-12 20:27 +0000 [r310415-310462]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, pbx/pbx_ael.c: Merged revisions 310448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600
	  (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011)
	  | 31 lines Add AELSub, which provides a stable entry point into
	  AEL subroutines. This commit needs some explanation, given that
	  we're adding a new application into an existing release branch.
	  This is generally a violation of our release policy, except in
	  very limited circumstances, and I believe this is one of those
	  circumstances. The problem that this solves is one of the sanity
	  of using multiple dialplan languages to define a dialplan. In the
	  case of the reporter, he or she is using AEL is define
	  subroutines, while using Realtime extensions to invoke those
	  subroutines. While you can do this, it's based upon the reality
	  of AEL using actual dialplan extensions; however, there is no
	  guarantee that the details of _how_ AEL is compiled into
	  extensions will remain stable. In fact, at the time of this
	  commit, it has already changed twice, once in a fundamental way.
	  Now normally, a new application would only be added to trunk.
	  However, this application is explicitly to create a stable
	  user-level API between versions, and adding it to trunk only will
	  not solve the user's problem of switching between 1.6.2 and 1.8,
	  nor will it help anybody switching from 1.8 to 1.10. Therefore,
	  it needs to go into existing release branches. For the sake of
	  consistency, and also because one of the changes was between 1.4
	  and 1.6.x, I am also electing to commit this to 1.4. (closes
	  issue #18910) Reported by: alexandrekeller Patches:
	  20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
	  (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
	  tilghman (license 14) Tested by: alexandrekeller ........
	  ................

	* /, funcs/func_odbc.c: Merged revisions 310414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011)
	  | 7 lines Transactional handles should be used for the insertbuf,
	  if available. Also, fix a possible resource leak. (closes issue
	  #18943) Reported by: irroot ........

2011-03-11 06:47 +0000 [r310287]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/rtp_engine.c: remote_bridge_loop: prevent segfault when
	  after transfer of IAX2 of DAHDI call If the channel condition is
	  one of the following after breaking out of the loop, don't try to
	  update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx
	  3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes
	  issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt
	  uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81
	  Review: https://reviewboard.asterisk.org/r/1128/

2011-03-10 16:05 +0000 [r310240]  Terry Wilson <twilson@digium.com>

	* main/manager.c, res/res_phoneprov.c: Add \r\n to remaining http
	  headers passed to ast_http_send r309204 changed the behavior of
	  ast_http_send. It now requires headers to be passed with trailing
	  \r\n. This change updates the remaining instances in the code
	  that did not pass the \r\n. (closes issue #18186) Reported by:
	  nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama
	  (license 1028) manager.diff.txt uploaded by twilson (license 396)
	  Tested by: lathama

2011-03-10 15:17 +0000 [r310231]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Be more tolerant of what URI we accept for
	  call completion PUBLISH requests. (closes issue #18946) Reported
	  by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson
	  (license 60) Tested by: GeorgeKonopacki

2011-03-10 05:53 +0000 [r310142]  Tilghman Lesher <tilghman@meg.abyt.es>

	* apps/app_voicemail.c, res/res_config_odbc.c, /,
	  funcs/func_odbc.c: Merged revisions 310141 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600
	  (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011)
	  | 5 lines Initialize column size to 0 to deal with a potential
	  UnixODBC bug on 64-bit systems. (closes issue #18295) Reported
	  by: pruiz ........ ................

2011-03-08 20:19 +0000 [r310088]  Jonathan Rose <jrose@digium.com>

	* channels/sip/dialplan_functions.c: Returns with an error notice
	  if CHANNEL function of SIP channel is read without arguments.
	  (Closes issue #18653) Reported by: wuwu Patches: diff.patch
	  uploaded by jrose (license 1225) Tested by: jrose

2011-03-08 18:10 +0000 [r310039]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: Spelling fix in "calendar show calendar"
	  s/Cartegories/Catagories/ (closes issue #18931) Reported by:
	  pdugas Patches: res_calendar.c.patch uploaded by pdugas (license
	  1222)

2011-03-08 16:37 +0000 [r309994]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Make pri parameter description consistent.

2011-03-07 22:07 +0000 [r309858]  Jonathan Rose <jrose@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 309857 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r309857 | jrose | 2011-03-07 16:04:44 -0600
	  (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) |
	  8 lines Bug fix for MixMonitor involving filenames with '.' not
	  in the extension Closes issue #18391) Reported by: pabelanger
	  Patches: bugfix.patch uploaded by jrose (license 1225) Tested by:
	  jrose ........ ................

2011-03-07 00:54 +0000 [r309808]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/ast_expr2.fl, channels/chan_dahdi.c, /, configure,
	  include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
	  configure.ac: Merged revisions 309251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011)
	  | 7 lines Revert previous 2 commits, and instead conditionally
	  redefine the same macro used in flex 2.5.35 that clashed with our
	  workaround. Not surprisingly, the workaround was exactly the same
	  code as was provided by the Flex maintainers, albeit in two
	  different places, in different macros. This should fix the
	  FreeBSD builds, which have an older version of Flex. ........

2011-03-07 00:13 +0000 [r309765]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample: Indicate that Asterisk uses the Allow
	  header to determine if MESSAGE requests should be sent.

2011-03-05 17:44 +0000 [r309720]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: Fix caller id passed to
	  openr2_chan_make_call (closes issue #18894) Reported by: malufrj
	  Tested by: moy

2011-03-05 10:29 +0000 [r309678]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/asterisk.c, /: Merged revisions 309677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011)
	  | 7 lines Missed part of the conversion when we started passing
	  ppid to astcanary. (closes issue #18850) Reported by: viraptor
	  Patches: canary_ppid.patch uploaded by viraptor (license 543)
	  ........

2011-03-04 19:38 +0000 [r309448-309585]  Matthew Nicholson <mnicholson@digium.com>

	* /, pbx/pbx_lua.c: Merged revisions 309584 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar
	  2011) | 2 lines Restore mysterious lua_pushvalue() call removed
	  in r309494. The mystery has been solved. ........

	* /, pbx/pbx_lua.c: Merged revisions 309541 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar
	  2011) | 4 lines Check for errors from fseek() when loading config
	  file, properly abort on errors from fread(), and supply a
	  traceback for errors generated when loading the config file.
	  Also, prepend a newline to traceback output so that the main
	  error message is on it's own line. ........

	* /, pbx/pbx_lua.c: Merged revisions 309494 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar
	  2011) | 2 lines remove mysterious lua_pushvalue() that is never
	  used ........

	* pbx/pbx_lua.c: Export global symbols from pbx_lua to allow
	  modules to be loaded. Fixes a regression introduced in r278132.
	  (closes issue #18671) Reported by: Igels Patches:
	  pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
	  Tested by: Igels

2011-03-04 15:22 +0000 [r309445]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt, channels/sig_pri.c, channels/sig_pri.h,
	  channels/chan_dahdi.c, funcs/func_channel.c: Get real channel of
	  a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name
	  format was changed for ISDN calls to:
	  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There
	  were several reasons that the channel name had to change. 1) Call
	  completion requires a device state for ISDN phones. The generic
	  device state uses the channel name. 2) Calls do not necessarily
	  have B channels. Calls placed on hold by an ISDN phone do not
	  have B channels. 3) The B channel a call initially requests may
	  not be the B channel the call ultimately uses. Changes to the
	  internal implementation of the Asterisk master channel list
	  caused deadlock problems for chan_dahdi if it needed to change
	  the channel name. Chan_dahdi no longer changes the channel name.
	  4) DTMF attended transfers now work with ISDN phones because the
	  channel name is "dialable" like the chan_sip channel names. For
	  various reasons, some people need to know which B channel a DAHDI
	  call is using. * Added CHANNEL(dahdi_span),
	  CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan
	  can determine the B channel currently in use by the channel. Use
	  CHANNEL(no_media_path) to determine if the channel even has a B
	  channel. * Added AMI event DAHDIChannel to associate a DAHDI
	  channel with an Asterisk channel so AMI applications can
	  passively determine the B channel currently in use. Calls with
	  "no-media" as the DAHDIChannel do not have an associated B
	  channel. No-media calls are either on hold or call-waiting.
	  (closes issue #17683) Reported by: mrwho Tested by: rmudgett
	  (closes issue #18603) Reported by: arjankroon Patches:
	  issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
	  Tested by: stever28, rmudgett

2011-03-04 01:50 +0000 [r309403]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c, /: Merged revisions 309356 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500
	  (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar
	  2011) | 9 lines fix small memory leak fix small memory leak
	  caused by a string allocation that wasn't freed (closes issue
	  #18907) Reported by: andy11 Patches:
	  asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
	  (license 1224) ........ ................

2011-03-02 19:54 +0000 [r309204-309256]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 309255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) |
	  8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in
	  users.conf for SIP. Since it's a duplicate, nothing is going to
	  be done, so delme doesn't need to be set at all. Strangely, when
	  this was added, this was being set to 1 in 1.6, and 0 in trunk.
	  (issue AST-439) ........

	* main/http.c: Fix consistency of CRLFs on HTTP headers that get
	  sent out. (closes issue #18186) Reported by: nivaldomjunior
	  Patches: 18186-httpheadernewline.diff uploaded by qwell (license
	  4)

2011-03-01 21:57 +0000 [r309126-309170]  Richard Mudgett <rmudgett@digium.com>

	* funcs/func_channel.c: Document CHANNEL(keypad_digits) and
	  CHANNEL(no_media_path). * Added XML documentation for
	  CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML
	  documentation for CHANNEL(reversecharge).

	* channels/sig_analog.c: Chan_dahdi does not retain CID when
	  detecting DTMF CID without polarity reversal. Looks like an
	  unintended change when sig_analog.c was extracted from
	  chan_dahdi.c. Removed useless conditional around needed code and
	  fixed resulting compiler warning. (closes issue #18667) Reported
	  by: enegaard Patches: issue18667.patch uploaded by enegaard
	  (license 1197) Tested by: enegaard JIRA SWP-2965

2011-03-01 16:09 +0000 [r309084]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 309083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011)
	  | 9 lines Fixes thread blocking issue in the sip TCP/TLS
	  implementation. (closes issue #18497) Reported by: vois Patches:
	  issues_18497.diff uploaded by dvossel (license 671) Tested by:
	  vois, rossbeer, kowalma, Freddi_Fonet ........

2011-02-28 11:10 +0000 [r308991-309035]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/ast_expr2.fl, /, configure,
	  include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
	  configure.ac: Merged revisions 309033-309034 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011)
	  | 4 lines A later version of flex already includes the fwrite
	  workaround code, which if used twice causes a compilation error.
	  Detect whether Flex will compile without the workaround; if so,
	  suppress our workaround code. ........ r309034 | tilghman |
	  2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify
	  meaning, removing double negative (stupid!) ........

	* /, funcs/func_odbc.c: Merged revisions 308990 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011)
	  | 7 lines Statements updating zero rows may return SQL_NO_DATA.
	  This is fine; it's handled. (closes issue #18815) Reported by:
	  irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot
	  (license 52) ........

2011-02-25 18:52 +0000 [r308945]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_sip.c: Fix Deadlock with attended transfer of SIP
	  call Call path sip_set_rtp_peer (locks chan then pvt)
	  transmit_reinvite_with_sdp try_suggested_sip_codec
	  pbx_builtin_getvar_helper (locks p->owner) But by the time
	  p->owner lock was attempted, seems as though chan and p->owner
	  were different. So in sip_set_rtp_peer, lock pvt first then lock
	  p->owner using deadlocking methods. (closes issue #18837)
	  Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded
	  by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81,
	  cmaj Review: [https://reviewboard.asterisk.org/r/1126/]

2011-02-24 21:38 +0000 [r308903]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Invalid read in ast_channel_set_caller_event().
	  Valgrind reported that ast_channel_set_caller_event() was reading
	  data from a freed buffer when using the pre_set structure.
	  Rearange things to pre-calculate the name and number pointer
	  before updating the caller party structure to see if the name or
	  number was changed.

2011-02-24 17:57 +0000 [r308815]  Terry Wilson <twilson@digium.com>

	* main/manager.c, /: Merged revisions 308814 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r308814 | twilson | 2011-02-24 11:54:49 -0600
	  (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011)
	  | 12 lines Don't broadcast FullyBooted to every AMI connection
	  The FullyBooted event should not be sent to every AMI connection
	  every time someone connects via AMI. It should only be sent to
	  the user who just connected. (closes issue #18168) Reported by:
	  FeyFre Patches: bug0018168.patch uploaded by FeyFre (license
	  1142) Tested by: FeyFre, twilson ........ ................

2011-02-24 15:06 +0000 [r308723]  Matthew Nicholson <mnicholson@digium.com>

	* main/udptl.c, /: Merged revisions 308722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600
	  (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu,
	  24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........
	  ................

2011-02-24 03:41 +0000 [r308679]  Terry Wilson <twilson@digium.com>

	* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
	  308678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011)
	  | 8 lines Use remotesecret to authenticate with a remote party
	  The remotesecret option was only being used for outbound
	  registration and not for placing calls. This patch uses
	  remotesecret on outbound calls if it is set, otherwise secret is
	  still used. Review: https://reviewboard.asterisk.org/r/1107/
	  ........

2011-05-09  Leif Madsen <lmadsen@digium.com>

	* Asteris 1.8.4 Released.

2011-04-25  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.4-rc3 Released.

	* Use SSLv23_client_method instead of old SSLv2 only.

		(closes issue 0019095)
		(closes issue 0019138)
		Reported by: tzafrir
		Patches: 
		      no_ssl2.diff uploaded by tzafrir (license 46)
		      Tested by: russell, chazzam

	* Resolve crash in ast_mutex_init()

2011-02-25  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.4-rc2 Released.

	* Fix Deadlock with attended transfer of SIP call
	  (Closes issue #18837. Reported, patched by alecdavis. Tested by
	  alecdavid, Irontec, ZX81, cmaj)

2011-02-23  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.4-rc1 Released.

2011-02-23 23:38 +0000 [r308622]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: sig_pri_new_ast_channel() should return NULL
	  when new_ast_channel() fails. (closes issue #18874) Reported by:
	  cmaj Patches:
	  patch-sig_pri-crash-possible-null-channel-pointer.diff.txt
	  uploaded by cmaj (license 830) JIRA SWP-3172

2011-02-22 15:31 +0000 [r308526]  Andrew Latham <lathama@gmail.com>

	* main/http.c: Use ast_debug for console logging Guessed the log
	  levels based on info that level 3 is the soft roof. Can we create
	  a page / document to define the levels?

2011-02-21 15:02 +0000 [r308416]  Matthew Nicholson <mnicholson@digium.com>

	* main/udptl.c, /: Merged revisions 308414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600
	  (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
	  2011) | 5 lines Properly check the bounds of arrays when decoding
	  UDPTL packets. Also, remove broken support for receiving UDPTL
	  packets larger than 16k. That shouldn't ever happen anyway.
	  AST-2011-002 FAX-281 ........ ................

2011-02-21 14:24 +0000 [r308393]  Andrew Latham <lathama@gmail.com>

	* main/http.c: Add HTTP URI Debug logging and update notice enable
	  reporting of the request URI / URL in debugging change funny
	  debug note to a serious note.

2011-02-19 14:06 +0000 [r308330]  Andrew Latham <lathama@gmail.com>

	* main/http.c: Add CSS MIME Type Modern browsers are checking for
	  the MIME Type of pages and in some cases will not load a file if
	  the type is wrong.

2011-02-19 11:02 +0000 [r308288]  Tilghman Lesher <tilghman@meg.abyt.es>

	* utils: A few more (copies of) files to ignore in this directory.

2011-02-18 00:07 +0000 [r308242]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323cDriver.c, addons/ooh323cDriver.h,
	  addons/chan_ooh323.c: added g729onlyA option for announce only
	  AnnexA g.729 codec in h.323 capabilities. Option can be global or
	  per user/peer.

2011-02-16 20:21 +0000 [r308150]  Paul Belanger <pabelanger@digium.com>

	* addons/ooh323c/src/ooSocket.c: Fix FreeBSD builds.

2011-02-16 07:57 +0000 [r308098]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooSocket.c: ifdef __linux__ keepalive
	  variables also

2011-02-15 23:34 +0000 [r308010]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c, /: Merged revisions 308007 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r308007 | qwell | 2011-02-15 17:33:24 -0600
	  (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
	  10 lines Fix regression that changed behavior of queues when
	  ringing a queue member. This reverts r298596, which was to fix a
	  highly bizarre and contrived issue with a queue member that
	  called into his own queue being transferred back into his own
	  queue. I couldn't reproduce that issue in any way. I think one of
	  the other recent transfer fixes actually fixed this. (closes
	  issue #18747) Reported by: vrban ........ ................

2011-02-15 23:08 +0000 [r307970]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls
	  only on linux, freebsd and others don't have these options on
	  sockets.

2011-02-15 19:52 +0000 [r307879-307962]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Don't crash when forcing caller id.

	* channels/sig_pri.c, include/asterisk/ccss.h, channels/sig_pri.h,
	  channels/chan_dahdi.c, channels/chan_sip.c, main/ccss.c: No
	  response sent for SIP CC subscribe/resubscribe request. Asterisk
	  does not send a response if we try to subscribe for call
	  completion after we have received a 180 Ringing. You can only
	  subscribe for call completion when the call has been cleared.
	  When we receive the 180 Ringing, for this call, its
	  call-completion state is 'CC_AVAILABLE'. If we then send a
	  subscribe message to Asterisk, it trys to change the
	  call-completion state to 'CC_CALLER_REQUESTED'. Because this is
	  an invalid state change, it just ignores the message. The only
	  state Asterisk will accept our subscribe message is in the
	  'CC_CALLER_OFFERED' state. Asterisk will go into the
	  'CC_CALLER_OFFERED' when the SIP client clears the call by
	  sending a CANCEL. Asterisk should always send a response. Even if
	  its a negative one. The fix is to allow for the CCSS core to
	  notify a CC agent that a failure has occurred when CC is
	  requested. The "ack" callback is replaced with a "respond"
	  callback. The "respond" callback has a parameter indicating
	  either a successful response or a specific type of failure that
	  may need to be communicated to the requester. (closes issue
	  #18336) Reported by: GeorgeKonopacki Tested by: mmichelson,
	  rmudgett JIRA SWP-2633 (closes issue #18337) Reported by:
	  GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634

2011-02-15 07:02 +0000 [r307750-307837]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, funcs/func_odbc.c: Merged revisions 307836 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011)
	  | 8 lines Need to retrieve the rows affected before using the
	  associated variable. (closes issue #18795) Reported by: irroot
	  Patches: 20110211__issue18795.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman ........

	* res/res_odbc.c, /: Merged revisions 307792 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011)
	  | 8 lines Increment usage count at first reference, to avoid a
	  race condition with many threads creating connections all at
	  once. (issue #18156) Reported by: asgaroth Patches:
	  20110214__issue18156.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman ........

	* apps/app_queue.c, apps/app_dial.c: Calling a gosub routine
	  defined in AEL from Dial/Queue ceased to work. A bug in AEL did
	  not distinguish between the "s" extension generated by AEL and an
	  "s" extension that was required to exist by the chan_dahdi (or
	  another channel) that was not supplied with a starting extension.
	  Therefore, AEL made incorrect assumptions about what commands
	  were permissable in the context. This was fixed by making AEL
	  generate a different extension name. However, Dial and Queue make
	  additional assumptions about the name of the default gosub
	  extension. Therefore, they needed to be brought into line with a
	  "macro" rendered by AEL (as a gosub), without breaking
	  traditional dialplans written without the aid of AEL. Related to
	  (issue #18480) Reported by: nivek (closes issue #18729) Reported
	  by: kkm Patches: 20110209__issue18729.diff.txt uploaded by
	  tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded
	  by kkm (license 888) Tested by: kkm

2011-02-10 22:39 +0000 [r307536]  Jason Parker <jparker@digium.com>

	* main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
	  revisions 307535 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r307535 | qwell | 2011-02-10 16:35:49 -0600
	  (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
	  8 lines Remove color when executing commands via a remote
	  console. Essentially this makes '-x' imply '-n' on rasterisk.
	  This was done in a different and incomplete way previously, which
	  I'm reverting here. (issue #18776) Reported by: alecdavis
	  ........ ................

2011-02-10 18:50 +0000 [r307509]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h:
	  Corrections for properly work with H.323v2 (older) endpoints and
	  other small fixes. Interpret remote side H.225 version.
	  Corrections for H.323v2 endpoints: don't start TCS and MSD before
	  connect, don't start TCS and MSD by accepting H.245 connection,
	  start TCS and MSD by StartH245 facility message. Other fixes: fix
	  non zeroended remoteDisplayName issue, small fixes in call
	  clearing by closing H.245 connection, tcp keepalive introduced on
	  TCP connections (now is hardcoded, will be configurable in the
	  future), don't force H.245tunneling if FastStart is active, don't
	  send Alerting singal more than once per call. (issue 0018542)
	  Reported by: vmikhelson Patches: issue18542-final-3.patch
	  uploaded by may213 (license 454) Tested by: vmikhelson

2011-02-10 17:44 +0000 [r307467]  Mark Michelson <mmichelson@digium.com>

	* configs/ccss.conf.sample: Fix a gaffe in the CCSS sample
	  configuration. Discovered by Philippe Lindheimer and pointed out
	  on #asterisk-dev

2011-02-09 21:44 +0000 [r307314]  Andrew Latham <lathama@gmail.com>

	* contrib/init.d/rc.debian.asterisk: Disable color during running
	  test (closes issue #18776) Reported by: alecdavis Patches:
	  ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
	  andrel, lathama

2011-02-09 21:06 +0000 [r307228-307273]  Jeff Peeler <jpeeler@digium.com>

	* main/astobj2.c: Add missing debug info for ao2_link for use with
	  REF_DEBUG in ao2 callback. (closes issue #18758) Reported by:
	  rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by
	  rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by
	  rgagnon (license 1202)

	* /, main/features.c: Merged revisions 307227 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011)
	  | 11 lines Make sure to set parking dial context for non-default
	  parking lots. Since parking_con_dial isn't settable, set all
	  parking lots to "park-dial". (closes issue #17946) Reported by:
	  bluecrow76 Patches:
	  asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
	  bluecrow76 (license 270) modified by me ........

2011-02-09 05:39 +0000 [r307142]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/lock.c: Initialize tracking variable in structure properly.
	  Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by
	  me.)

2011-02-08 21:24 +0000 [r307092]  Jason Parker <jparker@digium.com>

	* main/logger.c: Fix issue with verbose messages not showing on
	  remote console. This code was reworked recently, and since the
	  logchannel list hadn't been created yet at this point, and it was
	  a verbose message, it was being dropped on the floor. Now it'll
	  continue on to where it should be handled. (closes issue #18580)
	  Reported by: pabelanger

2011-02-08 21:13 +0000 [r307065]  Mark Michelson <mmichelson@digium.com>

	* main/ccss.c: Add a couple of useful channel variables for the CC
	  recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine
	  the channel and context that will be called when the recall
	  occurs.

2011-02-08 20:22 +0000 [r306999]  Andrew Latham <lathama@gmail.com>

	* doc/asterisk.sgml, doc/asterisk.8, configs/asterisk.conf.sample,
	  configs/voicemail.conf.sample: Documentation Updates Note default
	  polling setting in voicemail.conf Add missing config to
	  asterisk.conf Update manpage (issue #16505) Reported by: tzafrir
	  Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir
	  (license 46) Tested by: lathama, tzafrir

2011-02-08 20:18 +0000 [r306979]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 306973 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306973 | twilson | 2011-02-08 12:14:09 -0800
	  (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08
	  Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with
	  pedantic=yes ........ ................

2011-02-08 19:41 +0000 [r306866-306967]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 306966 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600
	  (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
	  Feb 2011) | 1 line fix this line again ........ ................

	* apps/app_voicemail.c, /: Merged revisions 306961 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600
	  (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011)
	  | 9 lines Backup file storing message duration is not used with
	  IMAP_STORAGE, remove code. The message duration is stored in the
	  body of the email when using IMAP_STORAGE, so nothing needs to
	  happen with the backup file. (closes issue #18718) Reported by:
	  kerframil ........ ................

	* apps/app_voicemail.c, /: Merged revisions 306865 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600
	  (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
	  Feb 2011) | 1 line make this safer and fully correct, pointed out
	  by Steve Davis ........ ................

2011-02-08 01:45 +0000 [r306826]  Andrew Latham <lathama@gmail.com>

	* UPGRADE.txt, include/asterisk/manager.h, doc/asterisk.sgml,
	  include/asterisk/doxygen/mantisworkflow.h: Documentation Updates.
	  More updates to the removed doc folder and start updates to the
	  man page. (issue #16505) Reported by: tzafrir Tested by: lathama

2011-02-07 22:43 +0000 [r306619-306674]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: Merged revisions 306673 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306673 | twilson | 2011-02-07 14:40:20 -0800
	  (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
	  | 10 lines Don't try to pickup a call in the middle of a
	  masquerade If A calls B which doesn't answer and C & D both try
	  to do a call pickup, it is possible for ast_pickup_call to answer
	  the call, then fail to masquerade one of the calls because the
	  other one is already in the process of masquerading. This patch
	  checks to see if the channel is in the process of masquerading
	  before call before selecting it for a pickup. Review:
	  https://reviewboard.asterisk.org/r/1094/ ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 306618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306618 | twilson | 2011-02-07 13:59:54 -0800
	  (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
	  | 10 lines Don't allow a REFER w/replaces to replace its own
	  dialog Asterisk currently accepts a REFER with a Refer-To with an
	  embedded Replaces header that matches the dialog of the REFER.
	  This would be a situation like A calls B, A calls C, A transfers
	  B to A, which is just silly. This patch makes the transfer fail
	  instead of making Asterisk freak out and forget to hang other
	  channels up. Review: https://reviewboard.asterisk.org/r/1093/
	  ........ ................

2011-02-07 17:36 +0000 [r306575]  Mark Michelson <mmichelson@digium.com>

	* main/ccss.c: Rearrange a bit of code in the generic CC recall
	  operation. By waiting to call the callback macro after the
	  CC_INTERFACES, extension, priority, and context have been set,
	  this information can be accessed more easily within the callback
	  macro. Reported by Philippe Lindheimer.

2011-02-04 19:24 +0000 [r306356]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c, /: Merged revisions 306346 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) |
	  9 lines Don't fallthrough to 'unknown' in the 'ringing' case.
	  This could cause improper exits from the queue. (closes issue
	  #18499) Reported by: zaltar Patches: app_queue.patch uploaded by
	  zaltar (license 1148) ........

2011-02-04 18:53 +0000 [r306324]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, apps/app_dial.c: Don't send redirecting updates
	  to the caller if the dialplan forked the call. Each fork in the
	  dial could be redirected and confuse the caller. For ISDN the
	  DivLeg1 and DivLeg3 messages would get confused because ISDN
	  redirects calls in sequence not in parallel. * Also fixed a
	  formatting inconsistency in app_dial.c and make a warning message
	  more useful about what frame type could not be written.

2011-02-03 23:49 +0000 [r306215]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Fix SIP deadlock involving state changes.
	  Once again a call to pbx_builtin_getvar_helper (and
	  pbx_builtin_setvar_helper) has caused locking problems. Both of
	  these functions lock the channel when the channel argument is
	  passed in! In this case, the suspected problem (the backtrace
	  makes it impossible to tell) was the private being locked in
	  sip_set_rtp_peer and then: transmit_reinvite_with_sdp
	  try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to
	  verify that the fix was only required in 1.8 and later.) (closes
	  issue #18491) Reported by: cmaj Patches:
	  chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license
	  830) Tested by: cmaj

2011-02-03 21:03 +0000 [r306127]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, /: Merged revisions 306126 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r306126 | twilson | 2011-02-03 12:56:00 -0800
	  (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011)
	  | 9 lines Set hangup cause in local_hangup When a call involves a
	  local channel (like SIP -> Local -> SIP), the hangup cause was
	  not being set. This resulted in SIP channels sometimes getting a
	  503 error instead of a 486 when the far side sent a busy. In
	  Asterisk 1.8+ this also can cause issues with CCSS that involve a
	  local channel. This patch sets the hangupcause for one side of
	  the local channel to the other in local_hangup for outbound
	  calls. ........ ................

2011-02-03 20:50 +0000 [r306124]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 306123 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011)
	  | 10 lines Set exception on channel in parking thread when
	  POLLPRI event detected. This is done just to make the code be
	  equivalent to the old select code. As noted in 303106 the same
	  issue was already fixed in this branch, but the exception was not
	  set on the channel in the case of POLLPRI. The reason that this
	  did not cause a problem here is because in 122923 the check in
	  __ast_read to check the exception flag was removed. (related to
	  #18637) ........

2011-02-03 15:50 +0000 [r305987]  Andrew Latham <lathama@gmail.com>

	* phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample, /:
	  res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
	  (issue #18713) Reported by: lathama Patches: snom_dir.diff
	  uploaded by lathama (license 1028) Tested by: lathama

2011-02-03 00:24 +0000 [r305923]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, main/manager.c, /, channels/chan_sip.c,
	  apps/app_sendtext.c: Merged revisions 305889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600
	  (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
	  | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
	  terminator in the buffer length. When the frame is queued it is
	  copied. If the null terminator is not part of the frame buffer
	  length, the receiver could see garbage appended onto it. * Add
	  channel lock protection with ast_sendtext(). * Fixed AMI SendText
	  action ast_sendtext() return value check. ........
	  ................

2011-02-02 20:05 +0000 [r305844]  Tilghman Lesher <tilghman@meg.abyt.es>

	* funcs/func_env.c: Eliminate a file descriptor leak when using the
	  FILE() dialplan function. (closes issue #18731) Reported by:
	  marioabajo

2011-02-02 19:27 +0000 [r305753-305838]  Andrew Latham <lathama@gmail.com>

	* apps/app_externalivr.c, configs/sip.conf.sample,
	  configs/skinny.conf.sample, configs/h323.conf.sample,
	  configs/sla.conf.sample, apps/app_voicemail.c,
	  configs/iax.conf.sample, funcs/func_enum.c,
	  configs/dundi.conf.sample, funcs/func_callcompletion.c,
	  configs/mgcp.conf.sample, configs/iaxprov.conf.sample,
	  configs/unistim.conf.sample: Replacing doc/* and asterisk.pdf
	  with wiki links Adding links to http(s)://wiki.asterisk.org

	* configs/ccss.conf.sample, configs/sip.conf.sample,
	  configs/skinny.conf.sample, main/config.c,
	  configs/h323.conf.sample, configs/sla.conf.sample,
	  main/ast_expr2.fl, res/res_srtp.c,
	  configs/chan_dahdi.conf.sample, configs/extconfig.conf.sample,
	  configs/res_snmp.conf.sample, main/ast_expr2f.c,
	  res/res_timing_dahdi.c: Replacing doc/* with wiki links Adding
	  links to http(s)://wiki.asterisk.org

	* channels/chan_sip.c: Replace link to old doc with new wiki page.
	  Link to
	  https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

2011-02-01 22:48 +0000 [r305692]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Reverse sense of an error test when reading
	  from astdb. (closes issue #18545) Reported by: jcovert Patches:
	  chan_iax2.c.patch uploaded by jcovert (license 551)

2011-02-01 21:14 +0000 [r305649]  Andrew Latham <lathama@gmail.com>

	* configs/sip.conf.sample: SIP Configuration Documentation sip show
	  settings reports qualifyfreq in milliseconds. sip.conf configures
	  qualifyfreg in seconds.

2011-02-01 19:23 +0000 [r305603]  Brett Bryant <bbryant@digium.com>

	* cel/cel_pgsql.c: Add a possible solution to a customer problem
	  with reloading cel_pgsql.so quickly.

2011-02-01 18:02 +0000 [r305560]  Andrew Latham <lathama@gmail.com>

	* CHANGES, Makefile, README, /: doc/tex dir removed, but
	  corresponding entries still exists Update README, CHANGES, and
	  Makefile. Direct users to http://wiki.asterisk.org for
	  documentation or to the AST.txt and AST.pdf included in the
	  tarball. (closes issue #18443) Reported by: bas Patches:
	  changes.diff uploaded by lathama (license 1028) readme.diff
	  uploaded by lathama (license 1028) Tested by: lathama bas

2011-02-01 17:04 +0000 [r305473]  Jason Parker <jparker@digium.com>

	* res/res_musiconhold.c, /: Merged revisions 305472 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r305472 | qwell | 2011-02-01 11:02:09 -0600
	  (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) |
	  9 lines Close file descriptor for timing source when a MOH class
	  gets destroyed. (closes issue #18457) Reported by: mcallist
	  Patches: 18457-closetimer.diff uploaded by qwell (license 4)
	  18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested
	  by: qwell, loloski ........ ................

2011-02-01 00:01 +0000 [r305343]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, /: Merged revisions 305342 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600
	  (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011)
	  | 7 lines Obtain the pri lock for PRI queue counters. Need to
	  obtain the pri lock when calling pri_dump_info_str() to avoid a
	  reentrancy problem when calculating the Q.921 Q count statistic.
	  JIRA AST-484 ........ ................

2011-01-31 23:07 +0000 [r305131-305254]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305253
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r305253 | qwell | 2011-01-31 16:59:34 -0600
	  (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
	  10 lines Prevent a crash when dialing a technology with no
	  destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
	  already had code to prevent this. The attempt that app_dial was
	  making to prevent it was not correct, so I fixed that. (closes
	  issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
	  gbour (license 1162) ........ ................

	* configs/sip.conf.sample, main/tcptls.c: Add alternative name for
	  config option. The SIP sample configuration had "tlscadir" as the
	  option name, but chan_sip used the more correct "tlscapath". Now
	  both are accepted. Discovered (sort of) by a user on IRC in
	  #asterisk

	* res/res_musiconhold.c: Fix compile error. pseudofd no longer
	  exists.

	* res/res_musiconhold.c, /: Merged revisions 305130 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r305130 | qwell | 2011-01-31 14:59:37 -0600
	  (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
	  2011) | 2 lines Set file descriptors to -1 on creation, so that
	  we don't see weirdness later. ........ ................

2011-01-31 13:56 +0000 [r305083]  Andrew Latham <lathama@gmail.com>

	* main/http.c: Asterisk HTTP response Content-type Address content
	  type for BSD and other platforms (closes issue #18456) Reported
	  by: alexo Patches: asterisk18_http.patch uploaded by alexo
	  (license 1175) Tested by: alexo

2011-01-31 07:51 +0000 [r304950-305040]  Tilghman Lesher <tilghman@meg.abyt.es>

	* include/asterisk/lock.h: Use the non-specific API aliases, to
	  avoid a problem with building the utils directory.

	* apps/app_voicemail.c, /: Merged revisions 304978 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600
	  (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
	  Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
	  ........ ................

	* main/utils.c, include/asterisk/lock.h, .cleancount, main/lock.c,
	  main/heap.c: Change mutex tracking so that it only consumes
	  memory in the core mutex object when it's actually being used.
	  This reduces the overall size of a mutex which was 3016 bytes
	  before this back down to 216 bytes (this is on 64-bit Linux with
	  a glibc-implemented mutex). The exactness of the numbers here may
	  vary slightly based upon how mutexes are implemented on a
	  platform, but the long and short of it is that prior to this
	  commit, chan_iax2 held down 98MB of memory on a 64-bit system for
	  nothing more than a table of 32767 locks. After this commit, the
	  same table occupies a mere 7MB of memory. (closes issue #18194)
	  Reported by: job Patches: 20110124__issue18194.diff.txt uploaded
	  by tilghman (license 14) Tested by: tilghman Review:
	  https://reviewboard.asterisk.org/r/1066

2011-01-30 00:11 +0000 [r304908]  Andrew Latham <lathama@gmail.com>

	* apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c,
	  funcs/func_realtime.c, res/res_calendar.c,
	  funcs/func_callcompletion.c: Add Function and Application
	  Relationships to documentation Add and extend the see-also
	  sections to the documentation for applications and functions in
	  an effort to expand the online documentation of the wiki. Also
	  check for and update any links to moved documentation in the doc
	  folder.

2011-01-29 23:07 +0000 [r304638-304866]  Sean Bright <sean@malleable.com>

	* res/res_config_ldap.c, /: Merged revisions 304865 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat,
	  29 Jan 2011) | 7 lines Plug some memory leaks in the LDAP
	  realtime driver. (closes issue #18435) Reported by: zaltar
	  Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
	  ........

	* /, apps/app_meetme.c: Merged revisions 304776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan
	  2011) | 15 lines If we fail to allocate our announcement objects,
	  make sure we don't leak objects. The majority of this patch was
	  committed already in r304726 and r304729. (issue #18225) Reported
	  by: kenji (issue #18444) Reported by: junky (closes issue #18343)
	  Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz
	  (license 834) ........

	* /, apps/app_meetme.c: Merged revisions 304773 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan
	  2011) | 9 lines When we pass the S() or L() options to MeetMe,
	  make sure that we honor C as well. Without this patch, if the
	  user was kicked from the conference via the S() or L() mechanism,
	  we would just hang up on them even if we also passed C (continue
	  in dialplan when kicked). With this patch we honor the C flag in
	  those cases. (closes issue #17317) Reported by: var ........

	* /, apps/app_meetme.c: Merged revisions 304729 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan
	  2011) | 15 lines Make sure that we unref the correct object when
	  ejecting the most recent caller. Currently, when we kick the last
	  user to enter, we decrement our own reference count which results
	  in a crash when we kick another user or when we exit the
	  conference ourselves. This will fix #18225 in 1.8 and trunk, but
	  that particular bug does not exist in 1.6.2. (closes issue
	  #18225) Reported by: kenji Patches: issue18225.patch uploaded by
	  seanbright (license 71) Tested by: seanbright ........

	* /, apps/app_meetme.c: Merged revisions 304726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan
	  2011) | 9 lines Fix user reference leak in MeetMe. We were
	  unlinking the user from the conferences user container, but not
	  decrementing the reference count of the user as well, resulting
	  in a leak. (closes issue #18444) Reported by: junky Tested by:
	  seanbright ........

	* /, apps/app_meetme.c: Merged revisions 304659,304682 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri,
	  28 Jan 2011) | 5 lines Don't leak references if we can't create a
	  pseudo channel for mixing in MeetMe. If there was a problem
	  allocating a pseudo channel when building our meetme, we weren't
	  destroying our user container or destroying the mutexes that we
	  created. ........ r304682 | seanbright | 2011-01-28 17:38:05
	  -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous
	  commit that snuck in. ........

	* main/acl.c: Restore some conditionals that we lost in r277814.
	  There are some cases where ast_append_ha() is called with a NULL
	  instead of a valid int pointer. So if we get a NULL, don't try to
	  dereference it. (closes issue #18162) Reported by: imcdona
	  Patches: issue0018162.patch uploaded by pabelanger (license 224)
	  Tested by: enegaard

2011-01-27 19:08 +0000 [r304554]  Richard Mudgett <rmudgett@digium.com>

	* main/ccss.c: Warning message if CALLCOMPLETION(cc_callback_macro
	  or cc_agent_dialstring) are empty. Test if the value pointer is
	  not NULL instead of not ast_strlen_zero().

2011-01-27 17:03 +0000 [r304462-304466]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac: Merged revisions 304465 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304465 | qwell | 2011-01-27 11:01:24 -0600
	  (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) |
	  9 lines Fix default prefix=/usr regression on non-Linux systems.
	  This partially reverts a change made in branches/1.4/ r267759,
	  which will cause issue #17013 to be reopened. This issue was
	  pointed out by a user on #asterisk, who helpfully discovered that
	  paths were being set incorrectly. To truly understand what was
	  wrong, one should run: svn diff --force -c<this revision>
	  configure ........ ................

	* /, configure: Merged revisions 304461 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304461 | qwell | 2011-01-27 10:48:00 -0600
	  (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan
	  2011) | 1 line Rerun bootstrap.sh with no changes, so that it is
	  more obvious what my next commit changes. ........
	  ................

2011-01-26 22:27 +0000 [r304339]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 304338 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011)
	  | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR
	  to commas to match 76703. ........

2011-01-26 21:02 +0000 [r304251]  Mark Michelson <mmichelson@digium.com>

	* main/udptl.c, /: Merged revisions 304250 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600
	  (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed,
	  26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in
	  ast_udptl ........ ................

2011-01-26 20:43 +0000 [r304245]  Matthew Nicholson <mnicholson@digium.com>

	* channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
	  channels/sip/reqresp_parser.c: Merged revisions 304244 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600
	  (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
	  2011) | 6 lines This patch modifies chan_sip to route responses
	  to the address the request came from. It also modifies chan_sip
	  to respect the maddr parameter in the Via header. ABE-2664
	  Review: https://reviewboard.asterisk.org/r/1059/ ........
	  ................

2011-01-26 20:23 +0000 [r304186]  Sean Bright <sean@malleable.com>

	* /, configs/queues.conf.sample: Merged revisions 304181 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500
	  (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed,
	  26 Jan 2011) | 1 line Make sure the sample queues.conf is
	  properly commented. ........ ................

2011-01-26 19:39 +0000 [r304150]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 304149 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600
	  (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
	  26 Jan 2011) | 2 lines Update documentation for
	  DAHDISendCallreroutingFacility() application. ..........
	  ................

2011-01-26 01:26 +0000 [r304097]  Sean Bright <sean@malleable.com>

	* /, main/file.c: Merged revisions 304096 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan
	  2011) | 12 lines Per the man page, setvbuf() must be called
	  before any other operation on an open file. We use setvbuf() to
	  associate a buffer with a stream, but we have already written to
	  the open file. This works (by chance) on Linux, but fails on
	  other platforms, such as OpenSolaris. (closes issue #16610)
	  Reported by: bklang Patches: setvbuf.patch uploaded by crjw
	  (license 963) Tested by: bklang, asgaroth, efutch ........

2011-01-25 23:28 +0000 [r304007]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 304006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600
	  (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
	  | 8 lines DTMF attended transfers sometimes fail for no apparent
	  reason. The loop in feature_request_and_dial() can exit when
	  Party C has answered without processing an AST_CONTROL_ANSWER.
	  Also sometimes an AST_CONTROL_ANSWER never happens even though
	  Party C has answered. Don't hangup Party C if he is up or we
	  receive an AST_CONTROL_ANSWER. ........ ................

2011-01-25 22:09 +0000 [r303962]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 303960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303960 | twilson | 2011-01-25 16:02:42 -0600
	  (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
	  | 16 lines Guard against retransmitting BYEs indefinitely In the
	  case of an attended transfer (A calls B, A atxfers to C) where A
	  becomes unreachable before replying to Asterisk's BYE, Asterisk
	  can sometimes retransmit the BYE indefinitely. This is because
	  __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
	  SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
	  out, it will not ever be marked as ALREADYGONE, so when
	  __sip_autodestruct is called again, we end up starting the cycle
	  over. This patch adds a call to sip_alreadygone(pkt->owner) in
	  retrans_pkt in the case of a BYE that has timed out. This should
	  prevent Asterisk from trying to transmit new BYE messages in the
	  future. Review: https://reviewboard.asterisk.org/r/1077/ ........
	  ................

2011-01-25 20:56 +0000 [r303907]  Matthew Nicholson <mnicholson@digium.com>

	* include/asterisk/res_fax.h, res/res_fax.c: Reimplemented fax
	  session reservation to reverse the ABI breakage introduced in
	  r297486.

2011-01-25 18:55 +0000 [r303860]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, channels/chan_sip.c: Merged revisions 303858 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011)
	  | 5 lines Fix "sip show user <tab>", so that it actually shows
	  results, instead of just completing the last entry. (closes issue
	  #16675) Reported by: pj ........

2011-01-25 17:49 +0000 [r303771]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
	  channels/chan_dahdi.c, channels/sig_ss7.h, /: Merged revisions
	  303769 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600
	  (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011)
	  | 40 lines Sending out unnecessary PROCEEDING messages breaks
	  overlap dialing. Issue #16789 was a good idea. Unfortunately, it
	  breaks overlap dialing through Asterisk. There is not enough
	  information available at this point to know if dialing is
	  complete. The ast_exists_extension(), ast_matchmore_extension(),
	  and ast_canmatch_extension() calls are not adequate to detect a
	  dial through extension pattern of "_9!". Workaround is to use the
	  dialplan Proceeding() application early in non-dial through
	  extensions. * Effectively revert issue #16789. * Allow outgoing
	  overlap dialing to hear dialtone and other early media. A
	  PROGRESS "inband-information is now available" message is now
	  sent after the SETUP_ACKNOWLEDGE message for non-digital calls.
	  An AST_CONTROL_PROGRESS is now generated for incoming
	  SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of
	  the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent
	  with the cause codes. * Added better protection from sending out
	  of sequence messages by combining several flags into a single
	  enum value representing call progress level. * Added diagnostic
	  messages for deferred overlap digits handling corner cases.
	  (closes issue #17085) Reported by: shawkris (closes issue #18509)
	  Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch
	  uploaded by rmudgett (license 664) Expanded upon
	  issue18509_early_media_v1.8_v3.patch to include analog and SS7
	  because of backporting requirements. Tested by: wimpy, rmudgett
	  ........ ................

2011-01-25 17:02 +0000 [r303678]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 303677 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600
	  (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011)
	  | 20 lines Fix voicemail sequencing for file based storage. A
	  previous change was made to account for when the number of
	  voicemail messages exceeds the max limit to be handled properly,
	  but it caused gaps in the messages to not be properly handled.
	  This has now been resolved. In later non 1.4 branches, it appears
	  that resequencing wasn't even occurring due from what appears and
	  accidental code removal. (closes issue #18498) Reported by:
	  JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license
	  325) (closes issue #18486) Reported by: bluefox Patches:
	  bug18486.patch uploaded by jpeeler (license 325) ........
	  ................

2011-01-24 20:51 +0000 [r303549]  Russell Bryant <russell@digium.com>

	* include/asterisk/channel.h, main/channel.c, main/pbx.c, /,
	  apps/app_meetme.c, main/features.c: Merged revisions 303548 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303548 | russell | 2011-01-24 14:49:53 -0600
	  (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
	  | 31 lines Fix channel redirect out of MeetMe() and other issues
	  with channel softhangup. Mantis issue #18585 reports that a
	  channel redirect out of MeetMe() stopped working properly. This
	  issue includes a patch that resolves the issue by removing a call
	  to ast_check_hangup() from app_meetme.c. I left that in my patch,
	  as it doesn't need to be there. However, the rest of the patch
	  fixes this problem with or without the change to app_meetme. The
	  key difference between what happens before and after this patch
	  is the effect of the END_OF_Q control frame. After END_OF_Q is
	  hit in ast_read(), ast_read() will return NULL. With the
	  ast_check_hangup() removed, app_meetme sees this which causes it
	  to exit as intended. Checking ast_check_hangup() caused
	  app_meetme to exit earlier in the process, and the target of the
	  redirect saw the condition where ast_read() returned NULL.
	  Removing ast_check_hangup() works around the issue in app_meetme,
	  but doesn't solve the issue if another application did the same
	  thing. There are also other edge cases where if an application
	  finishes at the same time that a redirect happens, the target of
	  the redirect will think that the channel hung up. So, I made some
	  changes in pbx.c to resolve it at a deeper level. There are
	  already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
	  attempt to abort the hangup process. My patch extends this to
	  remove the END_OF_Q frame from the channel's read queue, making
	  the "abort hangup" more complete. This same technique was used in
	  every place where a softhangup flag was cleared. (closes issue
	  #18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
	  https://reviewboard.asterisk.org/r/1082/ ........
	  ................

2011-01-24 17:20 +0000 [r303467]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
	  (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
	  8 lines Reset configuration before parsing users.conf. Some
	  values configured in chan_dahdi.conf were able to leak in to
	  users.conf configuration. This was surprising users, and
	  potentially setting non-sane "defaults". ASTNOW-125 ........
	  ................

2011-01-21 23:11 +0000 [r303286-303375]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c, /: Temporarily revert r303286

	* channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303285 | qwell | 2011-01-21 15:48:09 -0600
	  (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
	  8 lines Reset configuration before parsing users.conf. Some
	  values configured in chan_dahdi.conf were able to leak in to
	  users.conf configuration. This was surprising users, and
	  potentially setting non-sane "defaults". ASTNOW-125 ........
	  ................

2011-01-20 20:31 +0000 [r303153]  Richard Mudgett <rmudgett@digium.com>

	* main/ccss.c: Merged revision 303098 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu,
	  20 Jan 2011) | 15 lines CC_INTERFACES does not get built
	  correctly with local channels. If local channels are used with
	  CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall
	  fails. Also CC_INTERFACES gets "&(null)" appended to it. *
	  Initialize the buffer to eliminate the prepended garbage. *
	  Filter out the empty interface strings to eliminate the latter. *
	  Added a diagnostic message if the CC_INTERFACES is ever empty.
	  JIRA ABE-2740 JIRA SWP-2848 ..........

2011-01-20 19:57 +0000 [r303107]  Shaun Ruffell <sruffell@digium.com>

	* /, main/features.c: Merged revisions 303106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011)
	  | 15 lines main/features: Use POLLPRI when waiting for events on
	  parked channels. This change resolves a regression in the 1.6.2
	  when converting from select to poll. The DAHDI timers use POLLPRI
	  to indicate that the timer fired, but features was not waiting
	  for that flag. The result was no audio for MOH when a call was
	  parked and res_timing_dahdi was in use. This patch is slightly
	  modified from the one on the mantis issue. It does not set an
	  exception on the channel if the POLLPRI flag is set. (closes
	  issue #18262) Reported by: francesco_r Patches:
	  patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
	  Tested by: francesco_r, rfrantik, one47 ........

2011-01-20 17:10 +0000 [r303009]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
	  303008 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600
	  (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
	  | 8 lines Add new queue strategy to preserve behavior for when
	  queue members moved to ao2. Add queue strategy called "rrordered"
	  to mimic old behavior from when queue members were stored in a
	  linked list. ABE-2707 ........ ................

2011-01-20 16:12 +0000 [r302921]  Russell Bryant <russell@digium.com>

	* /, apps/app_privacy.c: Merged revisions 302920 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011)
	  | 2 lines Resolve a compiler warning. ........

2011-01-20 15:45 +0000 [r302918]  Leif Madsen <lmadsen@digium.com>

	* apps/app_dial.c, /: Merged revisions 302917 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011)
	  | 8 lines Option L() is milliseconds, not seconds. > Change the
	  verbose output of option L() to say milliseconds and not seconds
	  > as the value is in milliseconds. > > (closes issue #18264) >
	  Reported by: jacco > Patches: > app_dial_patch.txt uploaded by
	  lmadsen (license 10) ........

2011-01-19 23:56 +0000 [r302837]  Russell Bryant <russell@digium.com>

	* main/manager.c: Only check container count if it exists.

2011-01-19 23:49 +0000 [r302834]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c, /: Merged revisions 302833 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed,
	  19 Jan 2011) | 7 lines Support greetingsfolder as documented in
	  voicemail.conf.sample. (closes issue #17870) Reported by:
	  edhorton Patches:
	  __20100816-app_voicemail-greetingsfolder-support.txt uploaded by
	  lmadsen (license 10) ........

2011-01-19 23:29 +0000 [r302831]  Paul Belanger <pabelanger@digium.com>

	* contrib/scripts/install_prereq: Add binutils-dev for
	  BETTER_BACKTRACES

2011-01-19 23:06 +0000 [r302785-302789]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 302788 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011)
	  | 4 lines Turn a noisy verbose message into a debug message. This
	  can drown your console if you're using the AMI over HTTP.
	  ........

	* main/manager.c: Resolve a memory leak with the manager interface
	  is disabled. The intent of this check as it stands in previous
	  versions of Asterisk was to check if there are any active
	  sessions. If there were no sessions, then the function would
	  return immediately and not bother with queueing up the manager
	  event to be processed. Since the conversion of storing sessions
	  in an astobj2 container, this check will always pass. I changed
	  it to go back to checking what was intended. The side effect of
	  this was that if the AMI is disabled, the manager event queue is
	  populated anyway, but the code that runs to clear out the queue
	  never runs. A producer with no consumer is a bad thing. Reported
	  internally by kmorgan.

2011-01-19 21:29 +0000 [r302713]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 302693 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600
	  (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
	  | 15 lines DTMF transfer plays the wrong sounds for wrong number
	  or other call failure. * Set the default for features.conf.sample
	  xferfailsound option to "beeperr" as documented instead of
	  "pbx-invalid" and corrected the use of it in DTMF blind transfer
	  (#1). * Improved DTMF blind transfer handling of wrong numbers.
	  Most of the concerns in this issue were taken care of by the
	  patch for issue 17999: Issues with DTMF triggered attended
	  transfers. (closes issue #18379) Reported by: gincantalupo Tested
	  by: rmudgett ........ ................

2011-01-19 21:23 +0000 [r302634-302680]  Tilghman Lesher <tilghman@meg.abyt.es>

	* include/asterisk/astdb.h, /: Merged revisions 302675 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600
	  (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
	  Jan 2011) | 2 lines Add some API documentation ........
	  ................

	* main/app.c, /: Merged revisions 302599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011)
	  | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero
	  argument, we're expected to reap our own zombies. On a zero
	  argument, however, the zombies are only reaped when there aren't
	  any non-zero forked children alive. At other times, we accumulate
	  zombies. This code is forward ported from res_agi in 1.4, so that
	  forked children are always reaped, thus preventing an
	  accumulation of zombie processes. (closes issue #18515) Reported
	  by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
	  tilghman (license 14) Tested by: ernied ........

2011-01-19 20:14 +0000 [r302600]  Jason Parker <jparker@digium.com>

	* res/res_fax.c: Fix typo pointed out on asterisk-users list.

2011-01-19 19:03 +0000 [r302505-302555]  Sean Bright <sean@malleable.com>

	* main/utils.c, /: Merged revisions 302554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan
	  2011) | 7 lines Don't call strlen() when we only need to look at
	  the next character or two. (closes issue #18042) Reported by:
	  wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
	  by wdoekes (license 717) ........

	* /, main/features.c: Merged revisions 302551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan
	  2011) | 7 lines Remove an extraneous \r\n at the end of a parking
	  manager events. (closes issue #18363) Reported by:
	  clegall_proformatique Patches:
	  asterisk_1.8_295998_parking_manager_events_format.patch uploaded
	  by clegall proformatique (license 1139) ........

	* /, res/res_agi.c: Merged revisions 302548 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan
	  2011) | 10 lines Properly handle partial reads from fgets() when
	  handling AGIs. When fgets() failed with EAGAIN, we were
	  continually decrementing the available space left in our buffer,
	  resulting in botched command handling. (closes issue #16032)
	  Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
	  fnordian (license 110) ........

	* main/utils.c, /: Merged revisions 302504 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan
	  2011) | 7 lines Make sure that h_length is set when we
	  short-circuit out of ast_gethostbyname. (closes issue #16135)
	  Reported by: thedavidfactor Patches: utils.patch uploaded by
	  thedavidfactor (license 903) ........

2011-01-19 17:09 +0000 [r302462]  Paul Belanger <pabelanger@digium.com>

	* /, res/res_timing_timerfd.c: Merged revisions 302461 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed,
	  19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable'
	  error more gracefully. ........

2011-01-19 15:53 +0000 [r302412-302417]  Sean Bright <sean@malleable.com>

	* configs/extensions.conf.sample, /: Merged revisions 302416 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan
	  2011) | 9 lines Remove references to priorityjumping from the
	  sample extensions.conf. Priority jumping was removed from
	  pbx_config in r68970. (closes issue #18622) Reported by: kshumard
	  Patches: extensions.conf.sample.patch uploaded by kshumard
	  (license 92) ........

	* channels/chan_sip.c: Initialize an uninitialized variable.
	  (closes issue #18640) Reported by: jcovert Patches:
	  chan_sip.c.patch uploaded by jcovert (license 551)

	* channels/chan_local.c: Use appropriate type for requested format
	  in chan_local. We were passing and storing the requested format
	  as an int instead of format_t resulting in truncation. (closes
	  issue #18238) Reported by: whizemen Patches:
	  0018238_speex16.patch uploaded by whizemen (license 1143)

2011-01-18 22:04 +0000 [r302318]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Use the expanded format type instead of plain
	  int.

2011-01-18 21:43 +0000 [r302314]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 302313 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600
	  (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
	  2011) | 4 lines URI encode the user part of the contact header.
	  ABE-2705 ........ ................

2011-01-18 20:19 +0000 [r302267]  Russell Bryant <russell@digium.com>

	* main/astobj2.c: Don't enable AO2_DEBUG by default if AST_DEVMODE
	  is on. AO2_DEBUG is not important and is causing a false compiler
	  warning to be generated on my Ubuntu Natty dev box.

2011-01-18 20:19 +0000 [r302266]  Jeff Peeler <jpeeler@digium.com>

	* main/pbx.c, /: Merged revisions 302265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011)
	  | 27 lines Convert device state callbacks to ao2 objects to fix a
	  deadlock in chan_sip. Lock scenario presented here: Thread 1
	  holds ast_rdlock_contexts &conlock holds handle_statechange hints
	  holds handle_statechange hint waiting for cb_extensionstate
	  Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
	  handle_request_do &netlock holds find_call sip_pvt_ptr waiting
	  for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
	  (ast_rdlock_contexts) Chan_sip has an established locking order
	  of locking the sip_pvt and then getting the context lock. So the
	  as stated by the summary, the operations in thread 2 have been
	  modified to no longer require the context lock. (closes issue
	  #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
	  uploaded by one47 (license 23), modified by me Review:
	  https://reviewboard.asterisk.org/r/1072/ ........

2011-01-18 18:11 +0000 [r302174]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 302173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
	  (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
	  | 88 lines Issues with DTMF triggered attended transfers. Issue
	  #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
	  features.conf for attended transfer). 3) A hears MOH. B dial
	  number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
	  MOH. C ringing. 6) A hangup. C still ringing until
	  "atxfernoanswertimeout" expires. For v1.4 C will ring forever
	  until C answers the dead line. (Issue #17096) Problem: When A and
	  B hangup, C is still ringing. Issue #18395 SIP call limit of B is
	  1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
	  ringing 4. Timeout waiting for C to answer 5. Recall to B fails
	  because B has reached its call limit. Because B reached its call
	  limit, it cannot do anything until the transfer it started
	  completes. Issue #17273 Same scenario as issue 18395 but party B
	  is an FXS port. Party B cannot do anything until the transfer it
	  started completes. If B goes back off hook before C answers, B
	  hears ringback instead of the expected dialtone. ********** Note
	  for the issue #17273 and #18395 fix: DTMF attended transfer works
	  within the channel bridge. Unfortunately, when either party A or
	  B in the channel bridge hangs up, that channel is not completely
	  hung up until the transfer completes. This is a real problem
	  depending upon the channel technology involved. For chan_dahdi,
	  the channel is crippled until the hangup is complete. Either the
	  channel is not useable (analog) or the protocol disconnect
	  messages are held up (PRI/BRI/SS7) and the media is not released.
	  For chan_sip, a call limit of one is going to block that endpoint
	  from any further calls until the hangup is complete. For party A
	  this is a minor problem. The party A channel will only be in this
	  condition while party B is dialing and when party B and C are
	  conferring. The conversation between party B and C is expected to
	  be a short one. Party B is either asking a question of party C or
	  announcing party A. Also party A does not have much incentive to
	  hangup at this point. For party B this can be a major problem
	  during a blonde transfer. (A blonde transfer is our term for an
	  attended transfer that is converted into a blind transfer. :))
	  Party B could be the operator. When party B hangs up, he assumes
	  that he is out of the original call entirely. The party B channel
	  will be in this condition while party C is ringing, while
	  attempting to recall party B, and while waiting between call
	  attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
	  fix the problem. It will replace the party B channel technology
	  with a NULL channel driver to complete hanging up the party B
	  channel technology. The consequences of this code is that the 'h'
	  extension will not be able to access any channel technology
	  specific information like SIP statistics for the call.
	  ATXFER_NULL_TECH is not defined by default. ********** (closes
	  issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
	  SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
	  rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
	  shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
	  grecco Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1047/ ........
	  ................

2011-02-22  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.3 Released.

	* Merged changes related to AST-2011-002

2011-02-16  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.3-rc3 Released.

	------------------------------------------------------------------------
	r301790 | jpeeler | 2011-01-14 11:32:53 -0600 (Fri, 14 Jan 2011) | 42 lines

	Resolve deadlock involving REFER.

	(closes issue 0018403)
	Reported by: jthurman
	Patches: 
	      20110103-blind_deadlock.diff uploaded by jthurman (license 614)
	            issue18403.patch uploaded by jpeeler (license 325)
		    Tested by: jthurman

	------------------------------------------------------------------------

	------------------------------------------------------------------------
	r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10
	lines

	Fix regression that changed behavior of queues when ringing a queue
	member.

	This reverts r298596, which was to fix a highly bizarre and contrived
	issue with a queue member that called into his own queue being 
	transferred back into his own queue. I couldn't reproduce that issue in 
	any way. I think one of the other recent transfer fixes actually fixed 
	this.

	(closes issue 0018747)
	Reported by: vrban

	------------------------------------------------------------------------

2011-01-20  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.3-rc2 Released.

	------------------------------------------------------------------------
	r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2 
	lines

	Reimplemented fax session reservation to reverse the ABI breakage 
	introduced in r297486.
	------------------------------------------------------------------------

	------------------------------------------------------------------------
	r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
	lines

	main/features: Use POLLPRI when waiting for events on parked channels.

	This change resolves a regression in the 1.6.2 when converting from
	select to poll. The DAHDI timers use POLLPRI to indicate that the
	timer
	fired, but features was not waiting for that flag. The result was no
	audio for MOH when a call was parked and res_timing_dahdi was in use.

	This patch is slightly modified from the one on the mantis issue. It
	does
	not set an exception on the channel if the POLLPRI flag is set.

	(closes issue 0018262)
	Reported by: francesco_r
	Patches:
	      patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
	            Tested by: francesco_r, rfrantik, one47
	------------------------------------------------------------------------

	------------------------------------------------------------------------
	r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15
	lines

	Resolve a memory leak with the manager interface is disabled.

	The intent of this check as it stands in previous versions of Asterisk
	was to
	check if there are any active sessions.  If there were no sessions,
	then the
	function would return immediately and not bother with queueing up the
	manager
	event to be processed.  Since the conversion of storing sessions in an
	astobj2
	container, this check will always pass.  I changed it to go back to
	checking
	what was intended.

	The side effect of this was that if the AMI is disabled, the manager
	event
	queue is populated anyway, but the code that runs to clear out the
	queue
	never runs.  A producer with no consumer is a bad thing.

	Reported internally by kmorgan.

	------------------------------------------------------------------------

	------------------------------------------------------------------------
	r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2
	lines

	Only check container count if it exists.

	------------------------------------------------------------------------

2011-01-17  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.3-rc1 Released.

2011-01-18 18:11 +0000 [r302174]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 302173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
	  (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
	  | 88 lines Issues with DTMF triggered attended transfers. Issue
	  #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
	  features.conf for attended transfer). 3) A hears MOH. B dial
	  number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
	  MOH. C ringing. 6) A hangup. C still ringing until
	  "atxfernoanswertimeout" expires. For v1.4 C will ring forever
	  until C answers the dead line. (Issue #17096) Problem: When A and
	  B hangup, C is still ringing. Issue #18395 SIP call limit of B is
	  1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
	  ringing 4. Timeout waiting for C to answer 5. Recall to B fails
	  because B has reached its call limit. Because B reached its call
	  limit, it cannot do anything until the transfer it started
	  completes. Issue #17273 Same scenario as issue 18395 but party B
	  is an FXS port. Party B cannot do anything until the transfer it
	  started completes. If B goes back off hook before C answers, B
	  hears ringback instead of the expected dialtone. ********** Note
	  for the issue #17273 and #18395 fix: DTMF attended transfer works
	  within the channel bridge. Unfortunately, when either party A or
	  B in the channel bridge hangs up, that channel is not completely
	  hung up until the transfer completes. This is a real problem
	  depending upon the channel technology involved. For chan_dahdi,
	  the channel is crippled until the hangup is complete. Either the
	  channel is not useable (analog) or the protocol disconnect
	  messages are held up (PRI/BRI/SS7) and the media is not released.
	  For chan_sip, a call limit of one is going to block that endpoint
	  from any further calls until the hangup is complete. For party A
	  this is a minor problem. The party A channel will only be in this
	  condition while party B is dialing and when party B and C are
	  conferring. The conversation between party B and C is expected to
	  be a short one. Party B is either asking a question of party C or
	  announcing party A. Also party A does not have much incentive to
	  hangup at this point. For party B this can be a major problem
	  during a blonde transfer. (A blonde transfer is our term for an
	  attended transfer that is converted into a blind transfer. :))
	  Party B could be the operator. When party B hangs up, he assumes
	  that he is out of the original call entirely. The party B channel
	  will be in this condition while party C is ringing, while
	  attempting to recall party B, and while waiting between call
	  attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
	  fix the problem. It will replace the party B channel technology
	  with a NULL channel driver to complete hanging up the party B
	  channel technology. The consequences of this code is that the 'h'
	  extension will not be able to access any channel technology
	  specific information like SIP statistics for the call.
	  ATXFER_NULL_TECH is not defined by default. ********** (closes
	  issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
	  SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
	  rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
	  shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
	  grecco Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/1047/ ........
	  ................

2011-01-17 15:04 +0000 [r302005]  Terry Wilson <twilson@digium.com>

	* configs/sip.conf.sample: Document "encryption" option in
	  sip.conf.sample

2011-01-14 21:09 +0000 [r301946]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Deadlock between dahdi_request() and
	  pri_dchannel() processing an incomming call. The
	  sig_pri_new_ast_channel() is called with the channel private lock
	  held when pri_dchannel() calls it and no channel private lock
	  held when dahdi_request() calls it. The use of pri_grab() in
	  sig_pri_new_ast_channel() could leave the channel private lock
	  held when it returns if the lock was not held before calling it.
	  Make sig_pri_new_ast_channel() just lock the PRI span lock
	  instead of using pri_grab(). It is safe to do this because
	  dahdi_request() does not have the channel private lock and the
	  deadlock potential with the PRI span lock is only between
	  pri_dchannel() and other threads.

2011-01-14 20:11 +0000 [r301851]  Brett Bryant <bbryant@digium.com>

	* channels/chan_multicast_rtp.c: Changing previous revisions
	  301845/301847 to use ast_sockaddr_setnull() instead of setting
	  the field manually to avoid uninitialized data. Review:
	  https://reviewboard.asterisk.org/r/1076/

2011-01-14 20:05 +0000 [r301849]  Andrew Latham <lathama@gmail.com>

	* funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
	  function documentation. Fix amatuer type mistake

2011-01-14 19:35 +0000 [r301845]  Brett Bryant <bbryant@digium.com>

	* channels/chan_multicast_rtp.c: Fix for a consistent MulticastRTP
	  channel driver crash due to use of unitilized data. (closes issue
	  #18290) (closes issue #18602) Reported by: voipgate, wybecom
	  Review: https://reviewboard.asterisk.org/r/1076/

2011-01-14 19:35 +0000 [r301844]  Andrew Latham <lathama@gmail.com>

	* funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
	  function documentation.

2011-01-14 17:32 +0000 [r301790]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Resolve deadlock involving REFER. Two fixes:
	  1) One must always have the private unlocked before calling
	  pbx_builtin_setvar_helper to not invalidate locking order since
	  it locks the channel. 2) Unlock the channel before calling
	  pbx_find_extension, which starts and stops autoservice during the
	  lookup. The problem scenario as illustrated by the reporter:
	  Thread: do_monitor ----------------------- handle_request_do
	  handle_incoming handle_request_refer ast_parking_ext_valid
	  pbx_find_extension ast_autoservice_stop while (chan_list_state ==
	  as_chan_list_state) { usleep(1000); } Thread: autoservice_run
	  ----------------------- autoservice_run chan = ast_waitfor_n
	  ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
	  complex (depending on your system) ast_channel_lock(c[x]);
	  handle_request_do and schedule_process_request_queue locks the
	  owner if it exists. The autoservice thread is waiting for the
	  channel lock, which wasn't ever released since the do_monitor
	  thread was waiting for autoservice operations to complete. Solved
	  by unlocking the channel but keeping a reference to guarantee
	  safety. (closes issue #18403) Reported by: jthurman Patches:
	  20110103-blind_deadlock.diff uploaded by jthurman (license 614)
	  issue18403.patch uploaded by jpeeler (license 325) Tested by:
	  jthurman

2011-01-13 17:01 +0000 [r301731]  Leif Madsen <lmadsen@digium.com>

	* configs/phoneprov.conf.sample, /: Merged revisions 301730 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
	  | 7 lines Add static entry for split Polycom 332 firmware.
	  (closes issue #18607) Reported by: cjacobsen Patches:
	  polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
	  lathama ........

2011-01-12 21:19 +0000 [r301683]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 301682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
	  | 9 lines Don't reject all SUBSCRIBE auth requests When merging
	  another SUBSCRIBE fix from 1.4, some braces were put in the wrong
	  place. This patch fixes that. (closes issue #18597) Reported by:
	  thsgmbh ........

2011-01-12 18:51 +0000 [r301595]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c, /: Merged revisions 301594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
	  (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
	  be necessary in session_do, and removed the ms_t member from the
	  mansession_session structure. Merged revisions 301591 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
	  2011) | 5 lines Don't store the thread id for the manager session
	  in the structure we pass to the thread for the manager session.
	  ABE-2543 ........ ................

2011-01-12 18:12 +0000 [r301504]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 301503 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
	  (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
	  | 12 lines Fix CPU spike when pressing DTMF after agent login.
	  The problem here is that DTMF was being continuously deferred and
	  requeued since ast_safe_sleep is called in a loop. There are
	  serveral other places in the code that sleeps and then loops in a
	  similar fashion. Because of this fact I opted to not defer DTMF
	  any more, which will not affect the original fix:
	  https://reviewboard.asterisk.org/r/674 (closes issue #18130)
	  Reported by: rgj ........ ................

2011-01-12 16:05 +0000 [r301446]  David Vossel <dvossel@digium.com>

	* main/file.c: Removal of unused variables so Asterisk will
	  compile.

2011-01-12 15:57 +0000 [r301444]  Stefan Schmidt <sst@sil.at>

	* Makefile: fix wrong text of rerun menuselect after user interface
	  warning the warning, if no user interface for menuselect warning
	  was found is not right. you have to rerun configure before make
	  menuselect after installing a proper user interface. (closes
	  issue #18594) Reported by: Dovid

2011-01-12 00:26 +0000 [r301402]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/file.c: Call execl() directly for a better solution for
	  paths with spaces. (closes issue #18600) Reported by: ebroad
	  Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman
	  (license 14)

2011-01-11 19:16 +0000 [r301311]  Paul Belanger <pabelanger@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 301310 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan
	  2011) | 2 lines Fix a logic issue when passing context ARG
	  ........

2011-01-11 18:51 +0000 [r301308]  Matthew Nicholson <mnicholson@digium.com>

	* main/utils.c, /: Merged revisions 301307 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600
	  (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
	  2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
	  ABE-2705 ........ ................

2011-01-10 22:39 +0000 [r301263]  Tilghman Lesher <tilghman@meg.abyt.es>

	* main/strcompat.c: Little endian machines were not converted
	  properly. (closes issue #18583) Reported by: jcovert Patches:
	  20110110__issue18583.diff.txt uploaded by tilghman (license 14)
	  Tested by: jcovert

2011-01-09 21:40 +0000 [r301177-301221]  Paul Belanger <pabelanger@digium.com>

	* autoconf/ast_ext_lib.m4, /, configure, configure.ac: Merged
	  revisions 301220 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
	  2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds
	  files included in the Asterisk tarball were being ignored and
	  re-downloaded. Users wanting to cache the files can still
	  override the setting using the --with-sounds-cache option.
	  (closes issue #18589) Reported by: pabelanger Patches:
	  issue18589.patch uploaded by pabelanger (license 224) Tested by:
	  pabelanger Review: https://reviewboard.asterisk.org/r/1074/
	  ........

	* apps/app_verbose.c, /: Merged revisions 301176 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan
	  2011) | 7 lines Indicate log level argument for Log() is not
	  optional (closes issue #18586) Reported by: kshumard Patches:
	  app_verbose.c.patch uploaded by kshumard (license 92) ........

2011-01-08 01:11 +0000 [r301134]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: The DTMF attended transfer feature cannot
	  callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is
	  not dialable. Make a channel name like DAHDI/i3/400-12 dialable
	  when the sequence number is stripped off of the name.

2011-01-07 20:53 +0000 [r301090]  Jason Parker <jparker@digium.com>

	* /, apps/app_meetme.c: Merged revisions 301089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) |
	  8 lines Initialize useropts/adminopts in case there is no column
	  in the realtime DB. (closes issue #18182) Reported by: dimas
	  Patches: v1-18182.patch uploaded by dimas (license 88) Tested by:
	  dimas ........

2011-01-07 19:58 +0000 [r300955-301047]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 301046 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07
	  Jan 2011) | 8 lines Fix regression causing forwarding voicemails
	  to not work with file storage. I had actually already fixed this
	  in 295200 in 1.4 and thought it wasn't missing in the other
	  branches for some reason. (closes issue #18358) Reported by:
	  cabal95 ........

	* apps/app_voicemail.c, /: Merged revisions 300951 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600
	  (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011)
	  | 7 lines Ensure good bye prompt in voicemail is played at the
	  correct time. Specifically in the case of timing out but not
	  leaving voicemail nothing should be heard. And when leaving
	  voicemail it should be heard. ABE-2647 ........ ................

2011-01-06 06:28 +0000 [r300798]  Tilghman Lesher <tilghman@meg.abyt.es>

	* addons/res_config_mysql.c: Don't destroy handle not created by
	  use (because the caller will). (closes issue #18526) Reported by:
	  makoto Patches: res-config-mysql-include.patch uploaded by makoto
	  (license 38) Tested by: makoto

2011-01-05 20:54 +0000 [r300714]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Merged revision 300711 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed,
	  05 Jan 2011) | 14 lines A call retrieved from hold may wind up
	  with no audio. If the retrieved call is natively bridged then the
	  call may not have any audio path. The following warning message
	  is given: "Failed to add <dfd> to conference <chan>/<chan>:
	  Invalid argument". * Open the media on a B channel when
	  pri_fixup_principle() moves the call from a no_b_channel channel
	  to a real channel. * Added lock protection while
	  pri_fixup_principle() moves a call from one private structure to
	  another. * Made some pri_fixup_principle() messages more
	  meaningful. ..........

2011-01-05 18:56 +0000 [r300623]  Tilghman Lesher <tilghman@meg.abyt.es>

	* res/res_odbc.c, /: Merged revisions 300622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600
	  (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
	  | 10 lines Use the sanity check in place of the
	  disconnect/connect cycle. The disconnect/connect cycle has the
	  potential to cause random crashes. (closes issue #18243) Reported
	  by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
	  Tested by: ks3 ........ ................

2011-01-05 16:29 +0000 [r300575]  Paul Belanger <pabelanger@digium.com>

	* /, cdr/cdr_sqlite.c: Merged revisions 300574 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan
	  2011) | 6 lines Change deprecated message to LOG_WARNING Also
	  removed latter part of message Discussed on #asterisk-dev
	  ........

2011-01-04 21:53 +0000 [r300433-300521]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_iax2.c, main/xmldoc.c, /, channels/chan_sip.c,
	  channels/chan_agent.c: Merged revisions 300520 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011)
	  | 9 lines Fix backwards and broken XML documentation. (closes
	  issue #18547) Reported by: jcovert Patches: xmldoc.c.patch
	  uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded
	  by jcovert (license 551) chan_sip.c.patch uploaded by jcovert
	  (license 551) chan_agent.c.patch uploaded by jcovert (license
	  551) ........

	* configs/users.conf.sample, /: Merged revisions 300431 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011)
	  | 7 lines Add some documentation to users.conf.sample. (closes
	  issue #18531) Reported by: lathama Patches:
	  users.conf.sample2.diff uploaded by lathama (license 1028) Tested
	  by: lathama ........

2011-01-04 21:00 +0000 [r300430]  Russell Bryant <russell@digium.com>

	* contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
	  Merged revisions 300429 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r300429 | russell | 2011-01-04 14:59:56 -0600
	  (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
	  | 4 lines Update the autosupport script from Digium support.
	  (closes AST-395) ........ ................

2011-01-04 19:45 +0000 [r300384]  Leif Madsen <lmadsen@digium.com>

	* phoneprov/000000000000.cfg: Update STAT() to use the comma
	  instead of the pipe. (closes issue #18503) Reported by: cjacobsen
	  Patches: old_separator.diff uploaded by cjacobsen (license 1029)
	  Tested by: lathama

2011-01-04 17:54 +0000 [r300301]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 300298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r300298 | twilson | 2011-01-04 11:37:26 -0600
	  (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
	  | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
	  only skips authentication on retransmissions that are already
	  authenticated. A similar method is already used for INVITES. This
	  is the kind of thing we end up having to do when we don't have a
	  transaction layer... (closes issue #18075) Reported by: mdu113
	  Patches: diff.txt uploaded by twilson (license 396) Tested by:
	  twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
	  ........ ................

2011-01-04 17:01 +0000 [r300214]  Jan Kalab <pitlicek@gmail.com>

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c: Memory
	  leaking in calendars ne_request_destroy() was missing in
	  icalendar and exchange calendar modules, causing memory leak.
	  (closes issue #18521) Review:
	  https://reviewboard.asterisk.org/r/1068/

2011-01-03 23:14 +0000 [r300166]  Richard Mudgett <rmudgett@digium.com>

	* /, main/features.c: Merged revisions 300165 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011)
	  | 4 lines Use correct variable for atxfercallbackretries config
	  option. * Misc formatting changes. ........

2011-01-03 13:14 +0000 [r300082]  Leif Madsen <lmadsen@digium.com>

	* pbx/pbx_dundi.c: Increase side of mapping response field. I've
	  increased the size of the response field in a DUNDi mapping
	  because of some documentation I'm writing. Previously it was set
	  to AST_MAX_EXTENSION which is only 80 characters, which is far
	  too small when you're using some dialplan functions to craft a
	  response. The example I'm using is: extensions =>
	  RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial

2010-12-29 22:02 +0000 [r299989]  Tilghman Lesher <tilghman@meg.abyt.es>

	* apps/app_voicemail.c, main/file.c: Quote arguments, just in case
	  there's a space in a pathname. (Diagnosed by pabelanger on
	  #asterisk-dev, fixed by me.)

2010-12-29 19:28 +0000 [r299865-299948]  Paul Belanger <pabelanger@digium.com>

	* sounds/Makefile: Only remove /tmp/astdatadir, not
	  /var/lib/asterisk

	* build_tools/make_sample_voicemail, sounds/Makefile, Makefile:
	  Properly quote varibles for MAC OS X

	* apps/app_chanspy.c, /: Merged revisions 299864 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec
	  2010) | 2 lines Documentation typo ........

2010-12-27 21:23 +0000 [r299752-299820]  Tilghman Lesher <tilghman@meg.abyt.es>

	* sounds/Makefile: More space-in-pathname issues.

	* sounds/Makefile, Makefile, Makefile.moddir_rules: Mac OS X
	  spaces-in-pathnames fix.

	* configure: Regen configure

	* configure.ac: Properly quote path on Darwin.

2010-12-25 16:12 +0000 [r299711]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Change
	  order of sending TCS and MSD packets Change order of sending
	  Terminal Capability Set and MasterSlave Determination packets,
	  MSD send when TCS exchange procedure is done (we send tcs ack to
	  remote and we have remote tcs ack already or we receive tcs ack
	  from remote and we have send our tcs ack to remote already). Some
	  endpoints can work in this sequence only, i suggest they can't
	  work with both (tcs and msd) exchange procedures simultaneously.
	  Also changed StartH245 facility message sending. It send on
	  incoming calls only due to some endpoints can't proccess properly
	  this facility messages on their incoming calls. (issue #18433)
	  Reported by: MrHanMan Patches: tcs-msd-h245-3.patch uploaded by
	  may213 (license 454) Tested by: MrHanMan, may213

2010-12-25 10:07 +0000 [r299583-299626]  Tilghman Lesher <tilghman@meg.abyt.es>

	* channels/chan_local.c, /: Merged revisions 299625 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600
	  (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010)
	  | 5 lines Move check for extension existence below variable
	  inheritance, due to the possible use of an eswitch. (closes issue
	  #16228) Reported by: jlaguilar ........ ................

	* addons/res_config_mysql.c: Reset 'first' variable after usage.
	  (closes issue #18525) Reported by: makoto Patches:
	  res-config-mysql-update2.patch uploaded by makoto (license 38)

2010-12-23 02:53 +0000 [r299531]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c, /: Merged revisions 299530 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r299530 | moy | 2010-12-22 21:28:37 -0500 (Wed, 22 Dec
	  2010) | 7 lines Enqueue AST_CONTROL_PROGRESS after
	  AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
	  #18438) Reported by: mariner7 Tested by: moy ........

2010-12-22 20:05 +0000 [r299449]  Tilghman Lesher <tilghman@meg.abyt.es>

	* pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
	  pbx/ael/ael-test/ref.ael-vtest25,
	  pbx/ael/ael-test/ref.ael-vtest17, /,
	  pbx/ael/ael-test/ref.ael-test3: Merged revisions 299448 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010)
	  | 8 lines Resolve warnings by disambiguating the "s" extension as
	  used by chan_dahdi from the "s" extension as used by the AEL
	  macros. (closes issue #18480) Reported by: nivek Patches:
	  20101215__issue18480__2.diff.txt uploaded by tilghman (license
	  14) Tested by: nivek ........

2010-12-22 02:10 +0000 [r299405]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Chan_dahdi sends an empty COLP on the bridged
	  channel. Chan_dahdi always inserts a connected party IE when you
	  call from one dahdi channel to another dahdi channel, even if no
	  such information was received on the 2nd channel. This clears the
	  display of many phones. * Removed leftover artifact from before
	  the valid flag was added. * Updated all of the channel's caller
	  id information with the new connected line information instead of
	  just the string parts. (closes issue #18508) Reported by: wimpy
	  Patches: issue18508_trunk.patch uploaded by rmudgett (license
	  664) Tested by: wimpy, rmudgett

2010-12-21 15:25 +0000 [r299353]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 299242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600
	  (Mon, 20 Dec 2010) | 23 lines Merged revisions
	  299194,299198,299220 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
	  2010) | 6 lines Respond as soon as possible with a 202 Accepted
	  to refer requests. This change also plugs a few memory leaks that
	  can occur when parking sip calls. ABE-2656 ........ r299198 |
	  mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
	  lines Remove changes to via processing that were not supposed to
	  go into the last commit. ........ r299220 | mnicholson |
	  2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
	  ast_free() instead of free() ABE-2656 ........ ................

2010-12-21 00:44 +0000 [r299312]  Paul Belanger <pabelanger@digium.com>

	* configs/cel.conf.sample: Correct typo with USER_DEFINED event.
	  (closes issue #18461) Reported by: joscas Patches:
	  cel.conf.sample.diff uploaded by lathama (license 1028) Tested
	  by: lathama, joscas

2010-12-20 21:38 +0000 [r299248]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a couple of CCSS issues. * Make sure to
	  allocate a cc_params structure when creating autopeers. * Use
	  sip_uri_cmp when retrieving SIP CC agents and monitors in case
	  parameters appear in the URI. (closes issue #18504) Reported by:
	  kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches:
	  18338.diff uploaded by mmichelson (license 60) Tested by:
	  GeorgeKonopacki

2010-12-20 18:17 +0000 [r299131-299138]  Tilghman Lesher <tilghman@meg.abyt.es>

	* sample.call, /: Merged revisions 299136 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 Dec 2010)
	  | 2 lines Documentation fix ........

	* cdr/cdr_pgsql.c, /: Merged revisions 299130 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010)
	  | 11 lines If a call was not answered, then the billsec was
	  calculated unusually large. Also, due to a copy and paste error,
	  a request for the answer field would have given the start value,
	  instead. (closes issue #18460) Reported by: joscas Patches:
	  20101215__issue18460.diff.txt uploaded by tilghman (license 14)
	  Tested by: joscas ........

2010-12-20 16:18 +0000 [r299088]  Leif Madsen <lmadsen@digium.com>

	* /, main/features.c: Merged revisions 299087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010)
	  | 5 lines Note that Park() timeout is milliseconds. (closes issue
	  #15758) Reported by: mmurdock Tested by: mmurdock, seanbright
	  ........

2010-12-20 09:14 +0000 [r299004]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/aoc.c, channels/sig_pri.h, channels/chan_sip.c: Typos:
	  recieved => received

2010-12-18 00:09 +0000 [r298818-298963]  Tilghman Lesher <tilghman@meg.abyt.es>

	* /, main/say.c: Merged revisions 298962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r298962 | tilghman | 2010-12-17 18:08:57 -0600 (Fri, 17 Dec 2010)
	  | 2 lines Remove backtrace used for testing merge process
	  ........

	* main/utils.c, main/astobj2.c, utils/conf2ael.c,
	  include/asterisk/logger.h, configure,
	  build_tools/menuselect-deps.in, main/logger.c, utils/ael_main.c,
	  utils/hashtest2.c, makeopts.in, utils/check_expr.c,
	  utils/refcounter.c, include/asterisk/utils.h,
	  build_tools/cflags-devmode.xml, /,
	  include/asterisk/autoconfig.h.in, main/Makefile, main/say.c,
	  configure.ac, utils/hashtest.c: Merged revisions 298957 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600
	  (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
	  | 6 lines Let Asterisk find better backtrace information with
	  libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
	  use libbfd to search for better symbol information within both
	  the Asterisk binary, as well as loaded modules, to assist when
	  using inline backtraces to track down problems. ........
	  ................

	* contrib/init.d/rc.debian.asterisk: -v implies -f, so override
	  with -F. (closes issue #18446) Reported by: lathama Patches:
	  rc.debian.asterisk.diff uploaded by lathama (license 1028) Tested
	  by: lathama

	* /, configure, configure.ac: Merged revisions 298817 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17
	  Dec 2010) | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS
	  for SQLite 3, as it's needed on some platforms. (closes issue
	  #18493) Reported by: pprindeville Patches:
	  asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
	  Tested by: pprindeville ........

2010-12-17 17:26 +0000 [r298773]  Brad Watkins <Marquis42@gmail.com>

	* configs/sip.conf.sample, channels/chan_sip.c: Fix parsing of mwi
	  => lines in sip.conf Reworking parsing of mwi => lines to resolve
	  a segfault. Also add a set of unit tests for the function that
	  does the parsing. (closes issue #18350) Reported by: gbour Tested
	  by: Marquis, gbour Review:
	  https://reviewboard.asterisk.org/r/1053/

2010-12-16 23:31 +0000 [r298598-298685]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 298684 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600
	  (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
	  Dec 2010) | 2 lines After recording only silence for a voicemail
	  prepending, restore backup files. ........ ................

	* apps/app_queue.c, /: Merged revisions 298597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600
	  (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
	  | 7 lines Fix improper hangup when doing an attended transfer to
	  queue. Had to indicate ringing in wait_for_answer so the attended
	  transfer code would not try and hang up the local channel it
	  created, which would kill the call. ABE-2624 ........
	  ................

2010-12-16 09:28 +0000 [r298394-298539]  Tilghman Lesher <tilghman@meg.abyt.es>

	* channels/chan_sip.c: Ensure the ipaddr field in realtime is large
	  enough to handle IPv6 addresses. (closes issue #18464) Reported
	  by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG
	  (license 20) (plus a few additional lines by tilghman)

	* res/res_config_odbc.c, /: Merged revisions 298481 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600
	  (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010)
	  | 14 lines Only increment the pointer once per loop, otherwise we
	  corrupt the value. (closes issue #18251) Reported by: bcnit
	  Patches: 20101110__issue18251.diff.txt uploaded by tilghman
	  (license 14) Tested by: trev, jthurman, elguero (closes issue
	  #18279) Reported by: zerohalo Patches:
	  20101109__issue18279.diff.txt uploaded by tilghman (license 14)
	  Tested by: zerohalo ........ ................

	* /, funcs/func_dialgroup.c: Merged revisions 298477 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16
	  Dec 2010) | 8 lines Eliminate duplicates from container. (closes
	  issue #18091) Reported by: bunny Patches:
	  20101006__issue18091.diff.txt uploaded by tilghman (license 14)
	  Tested by: bunny ........

	* /, cdr/cdr_sqlite.c: Merged revisions 298393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600
	  (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
	  | 8 lines Unregister before shutting down the connection, to
	  avoid a race. (closes issue #18481) Reported by: pabelanger
	  Patches: 20101215__issue18481.diff.txt uploaded by tilghman
	  (license 14) Tested by: pabelanger ........ ................

2010-12-13 17:11 +0000 [r298195]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/chan_dahdi.c, /: Merged revisions
	  298194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600
	  (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010)
	  | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered
	  transfers. Outgoing PRI/BRI calls cannot do DTMF triggered
	  transfers if a PROCEEDING message is not received. The debug
	  output shows that the DTMF begin event is seen, but the DTMF end
	  event is missing. When the DTMF begin happens, the call is muted
	  so we now have one way audio (until a DTMF end event is somehow
	  seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER
	  event is received. * Made absorb the DTMF begin and DTMF end
	  events if we are overlap dialing and have not seen a PROCEEDING
	  message. * Added a debug message when absorbing a DTMF event.
	  JIRA SWP-2690 JIRA ABE-2697 ........ ................

2011-01-12  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.2 Released.

	* Merge in a change in the configure script to fix an issue for
	  Debian packagers.

	  ------------------------------------------------------------------------
	  r301221 | pabelanger | 2011-01-09 15:40:35 -0600 (Sun, 09 Jan 2011)
	  | 21 lines

	  Merged revisions 301220 via svnmerge from 
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 [^]

	  ........
	    r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
	    2011) | 14 lines
	    SOUND_CACHE_DIR now defaults to empty

	    Sounds files included in the Asterisk tarball were being
	    ignored and
	    re-downloaded. Users wanting to cache the files can
	    still override the setting
	    using the --with-sounds-cache option.

	    (closes issue 0018589)
	    Reported by: pabelanger
	       Patches:
	        issue18589.patch uploaded by
		pabelanger (license 224)
	        Tested by: pabelanger

	    Review:
	    https://reviewboard.asterisk.org/r/1074/

	    ------------------------------------------------------------------------

2010-12-13  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.2-rc1 Released.

2010-12-11 21:45 +0000 [r298099]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooGkClient.c: Correction to work with
	  gatekeeper which don't send GK ID Don't use GK ID if it's not
	  presented in GK replies Extract GK ID not only in GK confirm but
	  in GK register confirm also (issue #18401) Reported by: MrHanMan
	  Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested
	  by: may213, MrHanMan

2010-12-10 16:52 +0000 [r298054]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Prevent a memcpy overlap in
	  GENERIC_FAX_EXEC_SET_VARS

2010-12-10 16:26 +0000 [r298051]  Tilghman Lesher <tlesher@digium.com>

	* main/netsock.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 298050 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010)
	  | 11 lines Portability issue on OpenSolaris. Also detect the
	  required structure element, because OpenSolaris defines
	  SIOCGIFHWADDR, but without support for IP sockets. (closes issue
	  #18442) Reported by: ranjtech Patches:
	  20101209__issue18442.diff.txt uploaded by tilghman (license 14)
	  Tested by: ranjtech ........

2010-12-09 22:18 +0000 [r297965]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 297960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297960 | twilson | 2010-12-09 16:10:31 -0600
	  (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
	  | 14 lines Ignore spurious REGISTER requests If a REGISTER
	  request with a Call-ID matching an existing transaction is
	  received it was possible that the REGISTER request would
	  overwrite the initreq of the private structure. This info is used
	  to generate messages for other responses in the transaction. This
	  patch ignores REGISTER requests that match non-REGISTER
	  transactions. (closes issue #18051) Reported by: eeman Tested by:
	  twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
	  ................

2010-12-09 21:32 +0000 [r297957]  David Vossel <dvossel@digium.com>

	* channels/chan_gtalk.c: Fixes issue with outbound google voice
	  calls not working. Thanks to az1234 and nevermind_quack for their
	  input in helping debug the issue. (closes issue #18412) Reported
	  by: nevermind_quack Patches: fix uploaded by dvossel (license
	  671)

2010-12-09 20:48 +0000 [r297952]  Terry Wilson <twilson@digium.com>

	* main/features.c: Don't crash after Set(CDR(userfield)=...) in
	  ast_bridge_call Instead of setting peer->cdr = NULL, set it to
	  not post. (closes issue #18415) Reported by: macbrody Patches:
	  patch-18415 uploaded by jsolares (license 1167) Tested by:
	  jsolares, twilson

2010-12-08 18:06 +0000 [r297909]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 297908 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010)
	  | 4 lines Use inheritance to get correct results for
	  SIPFROMDOMAIN. (from an internal Digium discussion) ........

2010-12-08 16:12 +0000 [r297905]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Display the capabilities requested when requesting
	  a fax session fails instead of displaying a hex value. Tweak the
	  way fax stats are calculated so that all fax attempts and
	  faliures are logged. Also make ensure faxes are either counted as
	  completed or falied and never both. FAX-210

2010-12-07 22:59 +0000 [r297825]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 297824 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600
	  (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
	  | 12 lines Revert code that changed SSRC for DTMF. Some previous
	  behavior was attempted to be restored, but mistakingly I did not
	  realize that the previous behavior was incorrect. This fixes DTMF
	  not being detected since DTMF shouldn't cause the SSRC to change.
	  (related to issue #17404) (closes issue #18189) (closes issue
	  #18352) Reported by: marcbou Tested by: cmbaker82 ........
	  ................

2010-12-07 22:51 +0000 [r297733-297821]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/org.asterisk.muted.plist (added), Makefile,
	  contrib/init.d/org.asterisk.asterisk.plist, utils/muted.c, /:
	  Merged revisions 297819 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600
	  (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
	  | 4 lines Use non-deprecated APIs for CoreAudio Review:
	  https://reviewboard.asterisk.org/r/1040/ ........
	  ................

	* apps/app_followme.c, /: Merged revisions 297713 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600
	  (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
	  | 8 lines Don't create a Local channel if the target extension
	  does not exist. (closes issue #18126) Reported by: junky Patches:
	  followme.diff uploaded by junky (license 177) (partially
	  restructured by me to avoid a possible memory leak) ........
	  ................

2010-12-06 22:06 +0000 [r297607]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 297605 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600
	  (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
	  | 12 lines Improve handling of REGISTER requests with multiple
	  contact headers. The changes here attempt to more strictly follow
	  RFC 3261 section 10.3. Basically the following will now cause a
	  400 Bad Response to be returned, if: - multiple Contact headers
	  are present with one set to expire all bindings ("*") - wildcard
	  parameter is specified for Contact without Expires header or
	  Expires header is not set to zero. ABE-2442 ABE-2443 ........
	  ................

2010-12-03 17:41 +0000 [r297535]  Sean Bright <sean@malleable.com>

	* channels/chan_console.c, /: Merged revisions 297534 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri,
	  03 Dec 2010) | 3 lines The CLI command should not contain
	  <placeholder>s, these are for descriptions. ........

2010-12-03 15:21 +0000 [r297486-297495]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Print a DEBUG message instead of a WARNING message
	  when the selected fax tech does not support reserving sessions.
	  Answer the channel before quering it for t.38 support. This is
	  necessary for the query to work properly over local channels.

	* include/asterisk/res_fax.h, res/res_fax.c: Add support for
	  reserving a fax session before answering the channel. Note: this
	  change breaks ABI compatibility. FAX-217

2010-12-02 20:09 +0000 [r297406]  Paul Belanger <pabelanger@digium.com>

	* Makefile, /: Merged revisions 297405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500
	  (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
	  2010) | 7 lines Resolve compile error under FreeBSD We now set
	  _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
	  ASTCFLAGS to override the setting. Review:
	  https://reviewboard.asterisk.org/r/1043/ ........
	  ................

2010-12-02 18:13 +0000 [r297312]  Terry Wilson <twilson@digium.com>

	* /, main/abstract_jb.c: Merged revisions 297311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297311 | twilson | 2010-12-02 12:07:39 -0600
	  (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
	  | 12 lines Initialize offset for adaptive jitter buffer When the
	  adaptive jitter buffer is enabled in sip.conf, the first frame
	  placed in the jitter buffer fails with something like:
	  jb_warning_output: Resyncing the jb. last_delay 0, this delay
	  -215886466, threshold 1000, new offset 215886466 This happens
	  because the offset is not initialized before calling jb_put().
	  This patch modifies jb_put_first_adaptive() to set the offset to
	  the frame's timestamp. Review:
	  https://reviewboard.asterisk.org/r/1041/ ........
	  ................

2010-12-02 13:20 +0000 [r297245]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 297229 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297229 | russell | 2010-12-02 07:16:47 -0600
	  (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
	  | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
	  This is in response to some questions on IRC. To the user, there
	  was nothing that made it obvious that this error had anything to
	  do with DAHDI not being loaded. ........ ................

2010-12-01 19:47 +0000 [r297157]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Changed some NOTICE and WARNING messages to DEBUG
	  messages.

2010-12-01 17:53 +0000 [r297075]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 297073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600
	  (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
	  | 23 lines Fix not stopping MOH when transfered local channel
	  queue member is answered. The problem here is only present when
	  local channels are used with the MOH passthru option as well as
	  no optimization (/nm). I will describe the slightly bizarre
	  scenario that was used to test, where phones B and C are queue
	  members: Phone A dials into a queue with two members using local
	  channels and the above options. Phone B answers. Phone A blind
	  transfers phone B into the same queue. Phone A hangs up. Phone C
	  answers, but phone B didn't stop playing MOH. In this scenario,
	  the unhold frame that should have gotten to phone B never arrived
	  due to the masquerade from the blind transfer. This is usually
	  fine since app_queue manages the starting and stopping of MOH.
	  However, with the passthrough option enabled when app_queue
	  attempts to stop MOH it tries to do so on the local channel
	  rather than the real channel. The easiest solution was to just
	  make sure to send an unhold frame during the transfer since it
	  wouldn't make sense to have MOH playing after a transfer anyway.
	  This only modifies SIP transfers, but the other transfers did not
	  seem to be a problem. If DTMF based transfers were a problem it
	  might be okay to add ast_moh_stop to finishup, but I didn't want
	  to have to add that unless required. ABE-2624 ........
	  ................

2010-12-01 17:01 +0000 [r296951-296992]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/frame.h, /: Merged revisions 296991 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600
	  (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010)
	  | 5 lines Clarify documentation on how we store codec preference
	  lists. (closes issue #18397) Reported by: birgita ........
	  ................

	* channels/chan_iax2.c, /: Merged revisions 296950 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30
	  Nov 2010) | 2 lines Missed initializations caused startup errors
	  on Mac OS X (and possibly others, too). ........

2010-12-01 00:28 +0000 [r296870]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 296869 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600
	  (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010)
	  | 4 lines Properly restore backup information file when hanging
	  up during message prepending. ABE-2654 ........ ................

2010-11-30 19:12 +0000 [r296787]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: DOC: Conference number can be omitted; if
	  omitted, all users in a meetme are listed.

2010-11-29 23:05 +0000 [r296673]  Paul Belanger <pabelanger@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 296671 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500
	  (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov
	  2010) | 5 lines Make sure nothing else is needed before
	  destroying the scheduler. (closes issue #18398) Reported by:
	  pabelanger ........ ................

2010-11-29 21:26 +0000 [r296628]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Complete some error handling in
	  transmit_publish() in chan_sip.c. This error handling block
	  caught my eye. It was missing a couple of things, but it should
	  be safe now. Thanks to mmichelson for the quick peer review on
	  IRC.

2010-11-29 20:46 +0000 [r296582]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
	  revision 296575 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon,
	  29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling
	  as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY
	  redirecting number and notification code, SETUP redirecting
	  number) is also sent in PTMP/TE mode. It should only apply in
	  PTMP/NT mode. The call setup proceeds but the network (Deutsche
	  Telekom) reacts with ugly ISDN STATUS messages. Also don't send
	  the redirecting number ie when PTP is also sending the
	  DivertingLegInformation2 facility. The redirecting number ie is
	  redundant and the network (Deutsche Telekom) complains about it.
	  Patches: abe_2651_v4.patch uploaded by rmudgett (license 664)
	  JIRA ABE-2651 JIRA SWP-2537 ..........

2010-11-29 07:28 +0000 [r296534]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 296533 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010)
	  | 13 lines I love standards. There are so many to choose from.
	  Except when there isn't one. Linux and *BSD disagree on the
	  elements within the ucred structure. Detect which one is in use
	  on the system. (closes issue #18384) Reported by: bjm Patches:
	  cred-diffs uploaded by bjm (license 473)
	  20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
	  (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
	  tilghman (license 14) Tested by: tilghman, bjm ........

2010-11-27 10:40 +0000 [r296429-296467]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_meetme.c: Merged revisions 296466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010)
	  | 5 lines 18 characters is too short for most date/times (20 is
	  the usual, but we add more in case of greater precision). (closes
	  issue #18369) Reported by: tnakonz ........

	* include/asterisk.h: Also don't build DEBUG_FD_LEAKS when
	  STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj

2010-11-26 21:37 +0000 [r296391]  Olle Johansson <oej@edvina.net>

	* main/say.c: Merged revisions 296351 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre,
	  26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
	  lines Fix bugs in saying numbers using the Swedish language
	  syntax (closes issue #18355) Reported by: oej Patch by: oej Much
	  help from Peter Lindahl. Testing by the ClearIT team during a
	  coffee break. Review: https://reviewboard.asterisk.org/r/1033/
	  ........ ................

2010-11-26 18:31 +0000 [r296352-296354]  Brad Watkins <Marquis42@gmail.com>

	* res/res_jabber.c: Fix XMPP PubSub-based distributed device state.
	  Initialize pubsubflags to 0 so res_jabber doesn't think there is
	  already an XMPP connection sending device state. Also clean up
	  CLI commands a bit. (closes issue #18272) Reported by: klaus3000
	  Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000, Marquis Review:
	  https://reviewboard.asterisk.org/r/1030/

	* channels/chan_sip.c: Fix reloading of peer when a user is
	  requested. Prevent peer reloading from causing multiple MWI
	  subscriptions to be created when using realtime. This had the
	  effect of sending one NOTIFY for every time a sip peer made a
	  call, in one case eventually overwhelming the phone and causing
	  it to reboot. (closes issue #18342) Reported by: nivek Patches:
	  issue0018342p1.patch uploaded by nivek (license 636) Tested by:
	  nivek Review: https://reviewboard.asterisk.org/r/1029/

2010-11-24 23:29 +0000 [r296230]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 296221 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296221 | russell | 2010-11-24 17:28:19 -0600
	  (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
	  | 6 lines Make Asterisk less crashy. Since we might not put a new
	  translation path on the channel, go ahead and set it to NULL
	  right after destroying the old one to ensure we don't try to free
	  an invalid translation path later on. ........ ................

2010-11-24 22:49 +0000 [r296167]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  /, channels/sig_analog.h: Merged revisions 296166 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600
	  (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010)
	  | 43 lines Oneway audio to SIP phone from FXS port after FXS port
	  gets a CallWaiting pip. The FXS connected phone has to have
	  CW/CID support to fail, as it will send back a DTMF 'A' or 'D'
	  when it's ready to receive CallerID. A normal phone with no CID
	  never fails. Also the SIP phone does not hear MOH when the CW
	  call is answered. The DTMF end frame is suppressed when the phone
	  acknowledges the CW signal for CID. The problem is the DTMF begin
	  frame needs to be suppressed as well. The DTMF begin frame is
	  causing SIP to start sending the DTMF RTP frames. Since the DTMF
	  end frame is suppressed, SIP will not stop sending those DTMF RTP
	  packets. * Suppress the DTMF begin and end frames when the
	  channel driver is looking for DTMF digits. * Fixed a couple
	  issues caused by not cleaning up the CID spill if you answer the
	  CW call while it is sending the CID spill. * Fixed not sending
	  CW/CID spill to the phone when the call is natively bridged.
	  (Fixed by not using native bridge if CW/CID is possible.) *
	  Suppress received audio when sending CW/CID spills. The other
	  parties involved do not need to hear the CW/CID spills and may be
	  confused if the CW call is for them. (closes issue #18129)
	  Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
	  uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
	  NOTE: * v1.4 does not have the main problem fixed by suppressing
	  the DTMF start frames. The other three items fixed are relevant.
	  * If you really must restore native bridging between analog
	  ports, you need to disable CW/CID either by configuring
	  chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
	  dialing the number to temporarily disable CW. ........
	  ................

2010-11-24 20:23 +0000 [r296002-296084]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 296083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296083 | russell | 2010-11-24 14:23:11 -0600
	  (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
	  | 12 lines Fix false reporting of an error by set_format(). In
	  the case that the native format was able to be changed to match
	  the new requested format, the code proceeded to attempt to build
	  a translation path, anyway. The result would be NULL, since no
	  translation path is necessary and resulted in this function
	  thinking an error has occurred. This case is now specifically
	  caught and no attempt to build a translation path is attempted.
	  Thanks to our automated tests and bamboo.asterisk.org for
	  catching this problem and making a whole lot of noise when things
	  started failing. :-) ........ ................

	* apps/app_dial.c, main/channel.c, /: Merged revisions 296001 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r296001 | russell | 2010-11-24 11:03:16 -0600
	  (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
	  | 38 lines Handle failures building translation paths more
	  effectively. The problem scenario occurred on a heavily loaded
	  system that was using the codec_dahdi module and exceeded the
	  hardware transcoding capacity. The failure mode at that point was
	  not good. The report came in to us as an Asterisk lock-up. The
	  "core show locks" shows a ton of threads locked up (but no
	  obvious deadlock). Upon deeper investigation, when the system is
	  in this state, the CPU was maxed out. The CPU was being consumed
	  by the Asterisk logger spewing messages on every audio frame for
	  calls set up after transcoder capacity was reached. The purpose
	  of this patch is to make Asterisk handle failures to create a
	  translation path in a more graceful manner. If we can't
	  translate, then the call just needs to be dropped, as it's not
	  going to work. These are the changes: 1) In set_format() of
	  channel.c (which is called by set_read_format() and
	  set_write_format()), it was ignoring if
	  ast_translator_build_path() failed and returned NULL. It now pays
	  attention to that case and returns a result reflecting failure.
	  With this change in place, the bridging code will immediately
	  detect a failure and end the bridge instead of proceeding to try
	  to bridge frames that can't be translated and making channel
	  drivers freak out by sending them frames in a format they weren't
	  expecting. 2) In ast_indicate_data() of channel.c, failure of
	  ast_playtones_start() was ignored. It is now reflected in the
	  return value of the function. This didn't turn out to have any
	  affect on the bug, but seemed like a good change to leave in. 3)
	  In app_dial(), when only sending a call to a single endpoint, it
	  will attempt to do some bridging of its own of early audio. It
	  uses make_compatible() when it's going to do this. However, it
	  ignored failure from make compatible. So, even with the fix from
	  #1, if there was early audio going through app_dial, there would
	  still be a period of invalid frames passing through. After
	  detecting failure here, Dial() exits. ABE-2658 ........
	  ................

2010-11-23 10:30 +0000 [r295949]  Olle Johansson <oej@edvina.net>

	* /, main/say.c: Merged revisions 295907 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis,
	  23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
	  lines Fix support of saynumber(1,n) in the Swedish language
	  (closes issue #18353) Reported by: oej Review:
	  https://reviewboard.asterisk.org/r/1031/ ........
	  ................

2010-11-22 20:03 +0000 [r295869]  Sean Bright <sean@malleable.com>

	* configs/chan_dahdi.conf.sample, /: Merged revisions 295868 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov
	  2010) | 2 lines Change some documentation to suggest
	  dahdi_monitor instead of ztmonitor. ........

2010-11-22 19:36 +0000 [r295866]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_macro.c, include/asterisk/channel.h,
	  include/asterisk/frame.h, main/channel.c, main/pbx.c, /: Merged
	  revisions 295843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600
	  (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
	  | 46 lines The channel redirect function (CLI or AMI) hangs up
	  the call instead of redirecting the call. To recreate the
	  problem: 1) Party A calls Party B 2) Invoke CLI "channel
	  redirect" command to redirect channel call leg associated with A.
	  3) All associated channels are hung up. Note that if the CLI
	  command were done on the channel call leg associated with B it
	  works. This regression was a result of the fix for issue #16946
	  (https://reviewboard.asterisk.org/r/740/). The regression affects
	  all features that use an async goto to execute the dialplan
	  because of an external event: Channel redirect, AMI redirect, SIP
	  REFER, and FAX detection. The struct ast_channel._softhangup code
	  is a mess. The variable is used for several purposes that do not
	  necessarily result in the call being hung up. I have added
	  doxygen comments to describe how the various _softhangup bits are
	  used. I have corrected all the places where the variable was
	  tested in a non-bit oriented manner. The primary fix is the new
	  AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
	  the soft hangup requests that do not normally result in a hangup
	  do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
	  Reported by: SantaFox (closes issue #18185) Reported by:
	  kwemheuer (closes issue #18211) Reported by: zahir_koradia
	  (closes issue #18230) Reported by: vmarrone (closes issue #18299)
	  Reported by: mbrevda (closes issue #18322) Reported by: nerbos
	  Review: https://reviewboard.asterisk.org/r/1013/ ........
	  ................

2010-11-20 03:11 +0000 [r295747]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: One way audio before answering call
	  waiting call on analog port. * Analog call waiting Caller ID
	  spills could get stuck resulting in one way audio until the
	  waiting call is answered. This only happens on the second (and
	  later) call waiting call if the active call is not the first
	  call. * The CLI/AMI "dahdi show channel" command could report the
	  wrong channel information. Must keep the struct analog_pvt.owner
	  and struct dahdi_pvt.owner pointer in sync.

2010-11-20 00:50 +0000 [r295711]  Russell Bryant <russell@digium.com>

	* main/event.c, include/asterisk/event.h, /: Merged revisions
	  295710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010)
	  | 29 lines Fix cache of device state changes for multiple
	  servers. This patch addresses a regression where device states
	  across multiple servers were not being processing completely
	  correctly. The code works to determine the overall state by
	  looking at the last known state of a device on each server.
	  However, there was a regression due to some invasive rewrites of
	  how the cache works that led to the cache only storing the last
	  device state change for a device, regardless of which server it
	  was on. The code is set up to cache device state change events by
	  ensuring that each event in the cache has a unique device name +
	  entity ID (server ID). The code that was responsible for
	  comparing raw information elements (which EID is) always returned
	  a match due to a memcmp() with a length of 0. There isn't much
	  code to fix the actual bug. This patch also introduces a new CLI
	  command that was very useful for debugging this problem. The
	  command allows you to dump the contents of the event cache.
	  (closes issue #18284) Reported by: klaus3000 Patches:
	  issue18284.rev1.txt uploaded by russell (license 2) Tested by:
	  russell, klaus3000 (closes issue #18280) Reported by: klaus3000
	  Review: https://reviewboard.asterisk.org/r/1012/ ........

2010-11-19 22:06 +0000 [r295673]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 295672 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r295672 | twilson | 2010-11-19 13:55:48 -0800
	  (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
	  | 8 lines Discard responses with more than one Via This is not a
	  perfect solution as headers that are joined via commas are not
	  detected. This is a parsing issue that to fix "correctly" would
	  necessitate a new SIP parser. Review:
	  https://reviewboard.asterisk.org/r/1019/ ........
	  ................

2010-11-19 21:40 +0000 [r295670]  Brett Bryant <bbryant@digium.com>

	* apps/app_queue.c: Patch for deadlock from ordering issue between
	  channel/queue locks in app_queue (set_queue_variables). (closes
	  issue #18031) Reported by: rain Review:
	  https://reviewboard.asterisk.org/r/1018/

2010-11-19 16:47 +0000 [r295516]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Bring sig_analog extraction more into
	  alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI
	  support. * Fixed initial value of struct analog_pvt.use_callerid.
	  It may get forced on depending upon other config options. * Call
	  analog_dnd() instead of manual inlined code. * Removed unused
	  struct analog_pvt.usedistinctiveringdetection. * Removed the
	  struct analog_pvt.unknown_alarm flag. It was really the struct
	  analog_pvt.inalarm flag. * Use ast_debug() instead of
	  ast_log(LOG_DEBUG). * Rename several function's index variable to
	  idx. * Some formatting tweaks.

2010-11-18 20:30 +0000 [r295477]  Leif Madsen <lmadsen@digium.com>

	* configs/sip_notify.conf.sample: 'sip notify clear-mwi' needs
	  terminating CRLF. (closes issue #18275) Reported by: klaus3000
	  Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license
	  65)

2010-11-18 18:02 +0000 [r295361-295441]  Paul Belanger <pabelanger@digium.com>

	* res/res_jabber.c, /, include/asterisk/jabber.h: Merged revisions
	  295440 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov
	  2010) | 4 lines Fix compiler warnings when using openssl-dev
	  1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........

	* contrib/scripts/install_prereq: Add RedHat specific dependencies

	* configs/res_curl.conf.sample: Uncomment settings under [global],
	  to surpress warning when loading Asterisk.

2010-11-16 23:02 +0000 [r295282]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, /: Merged revisions 295281 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600
	  (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16
	  Nov 2010) | 1 line Dead code elimination in
	  channel.c:ast_channel_bridge() variable who. ........
	  ................

2010-11-16 22:41 +0000 [r295164-295278]  Russell Bryant <russell@digium.com>

	* build_tools/prep_tarball: Check for pdftotext and give a useful
	  error if not found.

	* build_tools/prep_tarball: Remove intentional typo I had added
	  when testing the check. oops.

	* build_tools/prep_tarball: Check for wikiexport.py in PATH and
	  give a useful error message if not found.

2010-12-02  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.1 Released.

2010-11-16  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.1-rc1 Released.

2010-11-15 18:30 +0000 [r294989-295078]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_expr.c (added), /: Merged revisions 295062 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
	  (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
	  Nov 2010) | 2 lines Create test verifying results of expression
	  parser ........ ................

	* funcs/func_curl.c, /: Merged revisions 294988 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
	  | 8 lines It is possible to crash Asterisk by feeding the curl
	  engine invalid data. (closes issue #18161) Reported by: wdoekes
	  Patches: 20101029__issue18161.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman ........

2010-11-12 21:14 +0000 [r294905-294911]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 294910 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12
	  Nov 2010) | 4 lines Return correct error code if lock path fails.
	  The recent changes to open_mailbox actually caused it to be
	  fixed, but let's be consistent. Reported by alecdavis in
	  asterisk-dev. ........

	* apps/app_voicemail.c, /: Merged revisions 294904 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
	  (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
	  | 16 lines Fix regression causing abort in voicemail after
	  opening a mailbox with no mesgs. In order to be more safe, some
	  error handling code was changed to respect more error conditions
	  including the potential memory allocation failure for deleted and
	  heard message tracking introduced in 293004. However,
	  last_message_index returns -1 for zero messages (perhaps as
	  expected) and was triggering the stricter error checking. Because
	  last_message_index is only called directly in one place, just
	  return 0 from open_mailbox (for file based storage) when no
	  messages are detected unless a real error has occurred. (closes
	  issue #18240) Reported by: leobrown Patches:
	  bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
	  Tested by: pabelanger ........ ................

2010-11-12 02:45 +0000 [r294823]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, channels/sig_pri.h, /: Merged revisions
	  294822 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
	  (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
	  | 11 lines Asterisk is getting a "No D-channels available!"
	  warning message every 4 seconds. Asterisk is just whining too
	  much with this message: "No D-channels available! Using Primary
	  channel XXX as D-channel anyway!". Filtered the message so it
	  only comes out once if there is no D channel available without an
	  intervening D channel available period. (closes issue #17270)
	  Reported by: jmls ........ ................

2010-11-11 22:17 +0000 [r294740-294745]  Russell Bryant <russell@digium.com>

	* doc/CCSS_architecture.pdf (removed): Remove CCSS architecture
	  PDF. It has been moved to:
	  https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture

	* doc/digium-mib.txt (removed), doc/followme.txt (removed),
	  doc/building_queues.txt (removed), doc/timing.txt (removed),
	  doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
	  doc/video_console.txt (removed), doc/macroexclusive.txt
	  (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
	  (added), doc/callfiles.txt (removed), doc/externalivr.txt
	  (removed), doc/codec-64bit.txt (removed),
	  build_tools/prep_tarball, doc/video.txt (removed), doc/jingle.txt
	  (removed), doc/modules.txt (removed), doc/manager_1_1.txt
	  (removed), doc/PEERING (removed), doc/snmp.txt (removed),
	  doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt
	  (removed), doc/ldap.txt (removed), doc/sip-retransmit.txt
	  (removed), doc/distributed_devstate.txt (removed),
	  doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
	  doc/queue.txt (removed), doc/jabber.txt (removed),
	  doc/chan_sip-perf-testing.txt (removed), Makefile,
	  doc/asterisk-mib.txt (removed), doc/database_transactions.txt
	  (removed), doc/smdi.txt (removed), doc/janitor-projects.txt
	  (removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt
	  (removed), doc/osp.txt (removed), doc/speechrec.txt (removed),
	  doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt
	  (removed), doc/valgrind.txt (removed), doc/realtimetext.txt
	  (removed), doc/cli.txt (removed), doc/rtp-packetization.txt
	  (removed), doc/datastores.txt (removed), doc/CODING-GUIDELINES
	  (removed), doc/ss7.txt (removed), doc/backtrace.txt (removed),
	  doc/India-CID.txt (removed): Remove most of the contents of the
	  doc dir in favor of the wiki content. This merge does the
	  following things: * Removes most of the contents from the doc/
	  directory in favor of the wiki - http://wiki.asterisk.org/ *
	  Updates the build_tools/prep_tarball script to know how to export
	  the contents of the wiki in both PDF and plain text formats so
	  that the documentation is still included in Asterisk release
	  tarballs.

2010-11-11 21:58 +0000 [r294640-294734]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 294733 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
	  (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
	  | 18 lines Fix problem with qualify option packets for realtime
	  peers never stopping. The option packets not only never stopped,
	  but if a realtime peer was not in the peer list multiple options
	  dialogs could accumulate over time. This scenario has the
	  potential to progress to the point of saturating a link just from
	  options packets. The fix was to ensure that the poke scheduler
	  checks to see if a peer is in the peer list before continuing to
	  poke. The reason a peer must be in the peer list to be able to
	  properly manage an options dialog is because otherwise the call
	  pointer is lost when the peer is regenerated from the database,
	  which is how existing qualify dialogs are detected. (closes issue
	  #16382) (closes issue #17779) Reported by: lftsy Patches:
	  bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
	  zerohalo ........ ................

	* main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
	  revisions 294639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600
	  (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
	  | 47 lines Fix a deadlock in device state change processing.
	  Copied from some notes from the original author (Russell):
	  Deadlock scenario: Thread 1: device state change thread Holds -
	  rdlock on contexts Holds - hints lock Waiting on channels
	  container lock Thread 2: SIP monitor thread Holds the "iflock"
	  Holds a sip_pvt lock Holds channel container lock Waiting for a
	  channel lock Thread 3: A channel thread (chan_local in this case)
	  Holds 2 channel locks acquired within app_dial Holds a 3rd
	  channel lock it got inside of chan_local Holds a local_pvt lock
	  Waiting on a rdlock of the contexts lock A bunch of other threads
	  waiting on a wrlock of the contexts lock To address this
	  deadlock, some locking order rules must be put in place and
	  enforced. Existing relevant rules: 1) channel lock before a pvt
	  lock 2) contexts lock before hints lock 3) channels container
	  before a channel What's missing is some enforcement of the order
	  when you involve more than any two. To fix this problem, I put in
	  some code that ensures that (at least in the code paths involved
	  in this bug) the locks in (3) come before the locks in (2). To
	  change the operation of thread 1 to comply, I converted the
	  storage of hints to an astobj2 container. This allows processing
	  of hints without holding the hints container lock. So, in the
	  code path that led to thread 1's state, it no longer holds either
	  the contexts or hints lock while it attempts to lock the channels
	  container. (closes issue #18165) Reported by: antonio ABE-2583
	  ........ ................

2010-11-10 23:26 +0000 [r294569-294605]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_spool.c: Fixing the Mac OS X build (bamboo warning)

	* pbx/pbx_spool.c: Properly queue files with inotify(7). (closes
	  issue #18089) Reported by: abelbeck Patches:
	  20101021__issue18089.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman

2010-11-10 14:14 +0000 [r294501-294535]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt, res/ais/clm.c, res/ais/evt.c: Tweak a couple of CLI
	  commands back to their original form. The "module" in this case
	  is two parts, so there are two words before the verb of the CLI
	  command.

	* main/devicestate.c, /: Merged revisions 294500 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
	  | 7 lines Improve a debug message to be more readable and
	  consistent. (closes issue #18282) Reported by: klaus3000 Patches:
	  ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
	  ........

2010-11-09 22:46 +0000 [r294466]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Allow ast_do_masquerade() failure to be reported
	  again.

2010-11-09 20:33 +0000 [r294430]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 294429 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
	  | 8 lines Detect GMime properly on systems where gmime flags and
	  libs are configured with pkg-config. (closes issue #16155)
	  Reported by: jcollie Patches: 20100917__issue16155.diff.txt
	  uploaded by tilghman (license 14) Tested by: tilghman ........

2010-11-09 16:55 +0000 [r294349]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/channel.h, channels/sig_pri.c, main/channel.c,
	  channels/chan_misdn.c, channels/sig_analog.c: Analog lines do not
	  transfer CONNECTED LINE or execute the interception macros. Add
	  connected line update for sig_analog transfers and simplify the
	  corresponding sig_pri and chan_misdn transfer code. Note that if
	  you create a three-way call in sig_analog before transferring the
	  call, the distinction of the caller/callee interception macros
	  make little sense. The interception macro writer needs to be
	  prepared for either caller/callee macro to be executed. The
	  current implementation swaps which caller/callee interception
	  macro is executed after a three-way call is created. Review:
	  https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
	  SWP-2372

2010-11-08 22:32 +0000 [r294278-294313]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_timing_timerfd.c: Merged revisions 294312 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
	  Nov 2010) | 1 line add missing unlock not present in 294277
	  ........

	* include/asterisk/timing.h, main/timing.c, main/channel.c, /,
	  res/res_timing_timerfd.c: Merged revisions 294277 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08
	  Nov 2010) | 16 lines Fix playback failure when using IAX with the
	  timerfd module. To fix this issue the alert pipe will now be used
	  when the timerfd module is in use. There appeared to be a race
	  that was not solved by adding locking in the timerfd module, but
	  needed to be there anyway. The race was between the timer being
	  put in non-continuous mode in ast_read on the channel thread and
	  the IAX frame scheduler queuing a frame which would enable
	  continuous mode before the non-continuous mode event was read.
	  This race for now is simply avoided. (closes issue #18110)
	  Reported by: tpanton Tested by: tpanton I put tested by tpanton
	  because it was tested on his hardware. Thanks for the remote
	  access to debug this issue! ........

2010-11-08 20:56 +0000 [r294243]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 294242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
	  2010) | 8 lines Go off hold when we get an empty reinvite telling
	  us to. (closes issue 0014448) Reported by: frawd (closes issue
	  #17878) Reported by: frawd ........

2010-11-08 19:56 +0000 [r294207]  Terry Wilson <twilson@digium.com>

	* configs/calendar.conf.sample, res/res_calendar.c: Set a default
	  waittime, and make sure to convert it to milliseconds

2010-11-08 17:16 +0000 [r294125]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: valgrind reported references to freed
	  memory during a mISDN hangup collision. Bad things have been
	  happening in chan_misdn because the chan_misdn channel private
	  struct chan_list is not protected from reentrancy. Hangup
	  collisions have be causing read and write accesses to freed
	  memory. Converted chan_misdn struct chan_list to an ao2 object
	  for its reference counting feature. ********** Removed an
	  impediment to converting chan_list to an ao2 object. The use of
	  the other_ch member in chan_list is shaky at best. It is set if
	  the incoming and outgoing call legs are mISDN. The use of the
	  other_ch member goes against the Asterisk architecture and can
	  even cause problems. 1) It is used to disable echo cancellation.
	  This could be bad if the call is forked and the winning call leg
	  is not mISDN or the winning call leg is not the last mISDN
	  channel called by the fork. The other_ch would become a dangling
	  pointer. 2) It is used when the far end is alerting to hear the
	  far end's inband audio instead of Asterisk's generated ringback
	  tone. This is bad if the call is forked. You would only hear the
	  last forked mISDN channel and it may not be ringing yet. The
	  other_ch would become a dangling pointer if the call is later
	  transferred. ********** JIRA SWP-2423 JIRA ABE-2614

2010-11-05 22:03 +0000 [r294084]  Brett Bryant <bbryant@digium.com>

	* channels/chan_sip.c: Fixed deadlock avoidance issues while
	  locking channel when adding the Max-Forwards header to a request.
	  (closes issue #17949) (closes issue #18200) Reported by: bwg
	  Review: https://reviewboard.asterisk.org/r/997/

2010-11-05 16:05 +0000 [r294047-294049]  Terry Wilson <twilson@digium.com>

	* contrib/scripts/ast_tls_cert: Corret spelling and example

	* contrib/scripts/ast_tls_cert: Tell people to use the correct
	  common name for clients as well

2010-11-05 00:07 +0000 [r293970]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c, /: Merged revisions 293969 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
	  (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
	  | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
	  receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
	  commit 9034) added the capability for the wctc4xxp to return more
	  than a single packet of data in response to a read. However, when
	  decoding packets, codec_dahdi was still assuming that the default
	  number of samples was in each read. In other words, each packet
	  your provider sent you, regardless of size, would result in 20 ms
	  of decoded data (30 ms if decoding G723). If your provider was
	  sending 60 ms packets then codec_dahdi would end up stripping 40
	  ms of data from each transcoded frame resulting in "choppy"
	  audio. This would only affect systems where G729 packets are
	  arriving in sizes greater than 20ms or G723 packets arriving in
	  sizes greater than 30ms. DAHDI-744. ........ ................

2010-11-04 21:39 +0000 [r293924]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes ringback tone on sip semi-attended
	  transfer. ABE-2168

2010-11-04 13:27 +0000 [r293887]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c: Do not output port in IPaddress for AMI
	  sippeers. (closes issue #18248) Reported by: orn Patches:
	  ami_sippeers.patch uploaded by pabelanger (license 224) Tested
	  by: orn

2010-11-03 18:35 +0000 [r293807]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  293806 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
	  (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
	  | 20 lines Party A in an analog 3-way call would continue to hear
	  ringback after party C answers. All parties are analog FXS ports.
	  1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
	  bring C into 3-way call before C answers. (A and B hear ringback)
	  4) C answers 5) A continues to hear ringback during the 3-way
	  call. (All parties can hear each other.) * Fixed use of wrong
	  variable in dahdi_bridge() that stopped ringback on the wrong
	  subchannel. * Made several debug messages have more information.
	  A similar issue happens if B and C are SIP channels. B continues
	  to hear ringback. For some reason this only affects v1.8 and
	  trunk. * Don't start ringback on the real and 3-way subchannels
	  when creating the 3-way conference. Removing this code is benign
	  on v1.6.2 and earlier. ........ ................

2010-11-03 18:05 +0000 [r293803]  Terry Wilson <twilson@digium.com>

	* include/asterisk/rtp_engine.h, main/rtp_engine.c,
	  channels/chan_sip.c: Avoid valgrind warnings for
	  ast_rtp_instance_get_xxx_address The documentation for
	  ast_rtp_instance_get_(local/remote)_address stated that they
	  returned 0 for success and -1 on failure. Instead, they returned
	  0 if the address structure passed in was already equivalent to
	  the address instance local/remote address or 1 otherwise. 90% of
	  the calls to these functions completely ignored the return
	  address and passed in an uninitialized struct, which would make
	  valgrind complain even though the operation was technically safe.
	  This patch fixes the documentation and converts the
	  get_xxx_address functions to void since all they really do is
	  copy the address and cannot fail. Additionally two new functions
	  (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
	  for the 3 times where the return value was actually checked. The
	  get_and_cmp_local_address function is currently unused, but
	  exists for the sake of symmetry. The only functional change as a
	  result of this change is that we will not do an
	  ast_sockaddr_cmp() on (mostly uninitialized) addresses before
	  doing the ast_sockaddr_copy() in the get_*_address functions. So,
	  even though it is an API change, it shouldn't have a noticeable
	  change in behavior. Review:
	  https://reviewboard.asterisk.org/r/995/

2010-11-02 23:09 +0000 [r293724]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 293723 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
	  (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
	  | 8 lines Add enabled/disabled information for rtautoclear sip
	  show settings output. When setting to zero/"no", the numeric
	  default was shown making it not obvious the disabled setting was
	  respected. (closes issue #18123) Reported by: zerohalo ........
	  ................

2010-11-02 21:29 +0000 [r293648]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  293647 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
	  (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
	  | 6 lines Make warning message have more useful information in
	  it. Change "Unable to get index, and nullok is not asserted" to
	  "Unable to get index for '<channel-name>' on channel <number>
	  (<function>(), line <number>)". ........ ................

2010-11-02 20:45 +0000 [r293611]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/manager.c: If manager and tls are disabled, do not display
	  TCP/TLS Bindaddress.

2010-11-01 17:29 +0000 [r293530]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Analog 3-way call would not connect all
	  parties if one was using sig_pri. Also the "dahdi show channel"
	  would not show the correct 3-way call status. * Synchronized the
	  inthreeway flag between chan_dahdi and sig_analog. * Fixed a
	  my_set_linear_mode() sign error and made take an analog sub
	  channel enum.

2010-11-01 16:09 +0000 [r293496]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_iax2.c: Use ast_sockaddr_from_sin function not
	  memcpy This resolves some IAX2 registration issue report on the
	  asterisk-users mailing list. (closes issue #18202) Reported by:
	  pabelanger Patches: update_registry.patch.v2 uploaded by
	  pabelanger (license 224) Tested by: pabelanger, Nic Colledge
	  (mailing list) Review: https://reviewboard.asterisk.org/r/993

2010-11-01 14:58 +0000 [r293493]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Only offer codecs both sides support for
	  directmedia When using directmedia, Asterisk needs to limit the
	  codecs offered to just the ones that both sides recognize,
	  otherwise they may end up sending audio that the other side
	  doesn't understand. (closes issue #17403) Reported by: one47
	  Patches: sip_codecs_simplified4 uploaded by one47 (license 23)
	  Tested by: one47, falves11 Review:
	  https://reviewboard.asterisk.org/r/967/

2010-10-30 01:53 +0000 [r293341-293418]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  293417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
	  (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
	  Oct 2010) | 1 line Remove some more code that serves no purpose.
	  ........ ................

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  293340 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
	  (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
	  Oct 2010) | 1 line Remove some code that serves no purpose.
	  ........ ................

2010-10-29 21:48 +0000 [r293305]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Modify sip_setoption to not complain about
	  unknown options. This now behaves just like the other setoption
	  callbacks. For the curious the offending option for the reporter
	  was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
	  fix for chan_local in 286189. (closes issue #17985) Reported by:
	  globalnetinc

2010-10-28 20:00 +0000 [r293197]  Tilghman Lesher <tlesher@digium.com>

	* res/ael/ael.tab.h, main/ast_expr2.c, /, main/ast_expr2.h,
	  res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c: Merged
	  revisions 293195-293196 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
	  (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
	  Reported (though the reporter did not understand he was reporting
	  a bug) on the asterisk-users list:
	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
	  ........ ................ r293196 | tilghman | 2010-10-28
	  14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
	  293194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
	  Reported (though the reporter did not understand he was reporting
	  a bug) on the asterisk-users list:
	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
	  ........ ................

2010-10-28 16:11 +0000 [r293159]  Jeff Peeler <jpeeler@digium.com>

	* /, funcs/func_strings.c: Merged revisions 293158 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28
	  Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically
	  when you're using characters above \x7f or invalid character
	  escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
	  Patches: issue18060_func_strings_filter_infinite_loop.patch
	  uploaded by wdoekes (license 717) Tested by: wdoekes ........

2010-10-26 18:49 +0000 [r293119]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 293118 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
	  (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
	  | 29 lines Fix inprocess_container in voicemail to correctly
	  restrict max messages. The comparison function logic was off, so
	  the number of sessions for a given mailbox were not being
	  incremented properly. This problem caused the maximum number of
	  messages per folder to not be respected when simultaneously
	  leaving multiple voicemails just below the threshold. These
	  problems should be fixed by the above, but just in case: Fixed
	  resequence_mailbox to rely on the actual number of detected
	  number of files in a directory rather than just assuming only 10
	  messages more than the maximum had been left. Also if more
	  messages than the maximum are deleted they are actually removed
	  now. The second purpose of this commit should have been separated
	  out probably, but is related to the above. Again, if the number
	  of messages in a given voicemail folder exceeds the maximum set
	  limit make sure to allocate enough space for the deleted and
	  heard index tracking array. A few random fixes: There was a
	  forgotten decrement of the inprocess count in imap_store_file.
	  When using IMAP storage, do not look in the directory where file
	  based storage messages may still reside and influence the message
	  count. Ensure to use only the first format in sendmail. ABE-2516
	  ........ ................

2010-10-26 16:32 +0000 [r293046-293081]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: No need to define the struct if there are no
	  users.

	* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Allow the DAHDI driver to compile, even with a
	  sufficiently older version of libpri. Fixes our Bamboo builds.

2010-10-25 21:15 +0000 [r292906-292969]  Tilghman Lesher <tlesher@digium.com>

	* channels/sig_pri.c: Several more defines that need to be altered
	  for compiling against an older version of libpri

	* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Allow the DAHDI driver to compile, even with a
	  sufficiently older version of libpri. Fixes our Bamboo builds.

2010-10-25 19:07 +0000 [r292868]  David Vossel <dvossel@digium.com>

	* channels/chan_local.c, /: Merged revisions 292867 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
	  (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
	  | 27 lines This patch turns chan_local pvts into astobj2 objects.
	  chan_local does some dangerous things involving deadlock
	  avoidance. tech_pvt functions like hangup and queue_frame are
	  provided with a locked channel upon entry. Those functions are
	  completely safe as long as you don't attempt to give up that
	  channel lock, but that is impossible to guarantee due to the
	  required deadlock avoidance necessary to lock both the tech_pvt
	  and both channels involved. In the past, we have tried to account
	  for this by doing things like setting a "glare" flag that
	  indicates what function should destroy the pvt. This was used in
	  local_hangup and local_queue_frame to decided who should destroy
	  the pvt if they collided in separate threads. I have removed the
	  need to do this by converting all chan_local tech_pvts to
	  astobj2. This means we can ref a pvt before deadlock avoidance
	  and not have to worry about that pvt possibly getting destroyed
	  under us. It also cleans up where we destroy the tech_pvt. The
	  only unlink from the tech_pvt container occurs in local_hangup
	  now, which is where it should occur. Since there still may be
	  thread collisions on some functions like local_hangup after
	  deadlock avoidance, I have added some checks to detect those
	  collisions and exit appropriately. I think this patch is going to
	  solve quite a bit of weirdness we have had with local channels in
	  the past. ........ ................

2010-10-22 22:35 +0000 [r292794-292825]  Terry Wilson <twilson@digium.com>

	* contrib/scripts/ast_tls_cert: Don't create directories without at
	  least o+x Also, making files that you are going to modify
	  read-only is dumb.

	* contrib/scripts/ast_tls_cert: Make files readable only by the
	  owner

2010-10-22 21:28 +0000 [r292787]  Leif Madsen <lmadsen@digium.com>

	* configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif, /,
	  channels/chan_sip.c: Merged revisions 292786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
	  | 13 lines Update the LDIF file for LDAP. The LDIF file
	  asterisk.ldif was quite a bit out of date from the
	  asterisk.ldap-schema file, so I've now updated that to be in
	  sync. The asterisk.ldif file being out of sync was a problem on
	  my systems where I was doing an ldapadd to import the schema into
	  the LDAP database, and the existing file would cause problems and
	  ERROR messages when registering. Additional documention has been
	  added based on feedback in the issue I'm closing. (closes issue
	  #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
	  by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
	  rgenthner ........

2010-10-22 17:09 +0000 [r292741]  Mark Michelson <mmichelson@digium.com>

	* tests/test_event.c: Prevent multiple runs of event_sub_test from
	  producing false failure results. The array of test subscriptions
	  was declared "static," meaning that the data.count field would
	  retain its value between runs of the test. After the first test
	  run, this would result in false reports of test failures. I chose
	  to just remove the "static" keyword from the structure since it's
	  not a huge deal to construct this structure during each run of
	  the test. Another alternative would have been to zero out the
	  data.count fields of each test subscription instead.

2010-10-22 16:49 +0000 [r292740]  Terry Wilson <twilson@digium.com>

	* contrib/scripts/ast_tls_cert (added): Add TLS cert helper script
	  This script is useful for quickly generating self-signed CA,
	  server, and client certificates for use with Asterisk. It is
	  still recommended to obtain certificates from a recognized
	  Certificate Authority and to develop an understanding how SSL
	  certificates work. Real security is hard work. OPTIONS: -h Show
	  this message -m Type of cert "client" or "server". Defaults to
	  server. -f Config filename (openssl config file format) -c CA
	  cert filename (creates new CA cert/key as ca.crt/ca.key if not
	  passed) -k CA key filename -C Common name (cert field) For a
	  server cert, this should be the same address that clients attempt
	  to connect to. Usually this will be the Fully Qualified Domain
	  Name, but might be the IP of the server. For a CA or client cert,
	  it is merely informational. Make sure your certs have unique
	  common names. -O Org name (cert field) An informational string
	  (company name) -o Output filename base (defaults to asterisk) -d
	  Output directory (defaults to the current directory) Example: To
	  create a CA and a server (pbx.mycompany.com) cert with output in
	  /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
	  This will create a CA cert and key as well as asterisk.pem and
	  the the two files that it is made from: asterisk.crt and
	  asterisk.key. Copy asterisk.pem and ca.crt somewhere (like
	  /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and
	  tlscafile=/etc/ca.crt. Since this is a self-signed key, many
	  devices will require you to import the ca.crt file as a trusted
	  cert. To create a client cert using the CA cert created by the
	  example above: ast_tls_cert -m client -c /tmp/ca.crt -k
	  /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user
	  This will create client.crt/key/pem in /tmp. Use this if your
	  device supports a client certificate. Make sure that you have the
	  ca.crt file set up as a tlscafile in the necessary Asterisk
	  configs. Make backups of all .key files in case you need them
	  later.

2010-10-22 15:47 +0000 [r292704]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, main/channel.c, channels/chan_misdn.c:
	  Connected line is not updated when chan_dahdi/sig_pri or
	  chan_misdn transfers a call. When a call is transfered by ECT or
	  implicitly by disconnect in sig_pri or implicitly by disconnect
	  in chan_misdn, the connected line information is not exchanged.
	  The connected line interception macros also need to be executed
	  if defined. The CALLER interception macro is executed for the
	  held call. The CALLEE interception macro is executed for the
	  active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
	  abe_2589_c3bier.patch uploaded by rmudgett (license 664)
	  abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
	  https://reviewboard.asterisk.org/r/958/

2010-10-21 22:09 +0000 [r292667]  Tilghman Lesher <tlesher@digium.com>

	* channels/misdn/ie.c: Compile correctly on Linux
	  (asterisk/localtime.h depends upon asterisk/autoconfig.h loading
	  first).

2010-10-21 18:13 +0000 [r292628]  Paul Belanger <paul.belanger@polybeacon.com>

	* contrib/init.d/rc.suse.asterisk: Fix typo in SUSE init script.
	  Reported by: Dave Cotton on asterisk-users list.

2010-10-21 16:14 +0000 [r292595]  David Vossel <dvossel@digium.com>

	* main/manager.c: Fixes recursive lock problem in manager.c It is
	  possible for a AMI session to freeze because of invalid use of
	  recursive locks during the EVENT processing. This patch removes
	  the unnecessary locks. (closes issue #18167) Reported by: sustav
	  Patches: manager_locking_v1.diff uploaded by dvossel (license
	  671) Tested by: sustav

2010-10-21 13:12 +0000 [r292557]  Leif Madsen <lmadsen@digium.com>

	* configs/res_ldap.conf.sample, /: Merged revisions 292556 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010)
	  | 6 lines Change res_ldap.sample.conf to match the schema.
	  (closes issue #17376) Reported by: jcovert Patches:
	  res_ldap.conf.sample.patch uploaded by jcovert (license 551)
	  ........

2010-10-21 11:36 +0000 [r292523]  Russell Bryant <russell@digium.com>

	* res/res_config_ldap.c: Add var=value to log message on update
	  failure, and add newline. ... just for you, Leif.

2010-10-21 01:02 +0000 [r292489]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Send CONNECT_ACKNOWLEDGE for CIS calls too.
	  The originator of the Q.SIG call completion signaling link was
	  not changed to the active state when the CONNECT message came in.
	  The T309 processing would immediately kill the signaling link
	  because it was not in the active state.

2010-10-21 00:21 +0000 [r292413-292436]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_voicemail.c: Application not properly unregister in
	  voicemail (closes issue #18128) Reported by: junky Patches:
	  vm_unregister.diff uploaded by junky (license 177) Tested by:
	  pabelanger, lmadsen

	* apps/app_dial.c, /: Merged revisions 292412 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400
	  (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
	  2010) | 10 lines Record priv-recordintro as sln, not gsm This
	  removes the gsm->sln step when transcoding priv-recordintro.
	  (closes issue #18176) Reported by: pabelanger Patches:
	  chan_sip.diff uploaded by pabelanger (license 224) ........
	  ................

2010-10-20 00:40 +0000 [r292376]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Oops. This module uses the generic timer
	  and no longer uses DAHDI. This causes a problem with the Solaris
	  and other system builds that have gcc 4.1 (where optional_api is
	  non-optional).

2010-10-19 22:14 +0000 [r292343]  Paul Belanger <paul.belanger@polybeacon.com>

	* contrib/scripts/install_prereq: Add resample and imap_tk
	  dependencies.

2010-10-19 19:27 +0000 [r292309]  Terry Wilson <twilson@digium.com>

	* res/res_srtp.c, channels/chan_sip.c: Add sip show peer info about
	  crypto and remove dated comment This patch adds information about
	  the encryption setting to 'sip show peers' and removes an
	  out-of-date comment from res_srtp.c and instead directs users to
	  the proper documentation. (closes issue #18140) Reported by:
	  chodorenko

2010-10-21  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0 Released.

2010-10-18  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-rc5 Released.

2010-10-18 22:02 +0000 [r292230]  Leif Madsen <lmadsen@digium.com>

	* sounds/Makefile, /: Merged revisions 292229 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
	  | 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
	  ........

2010-10-18 21:55 +0000 [r292227]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
	  (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
	  | 11 lines Fix improper operator key acceptance and clean up temp
	  recording files. This is a fix for when pressing the operator key
	  after recording an unavailable, busy, name, or temporary message
	  in mailbox options. The operator key should not be accepted here,
	  but should be allowed during the message recording. If the
	  operator key is pressed during ensure the file is saved or
	  deleted as apporopriate. Also, ensure removal of temporary
	  recorded files after an early hang up or when message acceptance
	  confirmation times out. ABE-2518 ........ ................

2010-10-18 21:51 +0000 [r292225]  Leif Madsen <lmadsen@digium.com>

	* sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
	  (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
	  | 9 lines Add support for the new English (Australian Accent)
	  sound files. (closes issue #17426) Reported by: camsown Patches:
	  core-sounds-en_AU.txt uploaded by camsown (license 1050)
	  add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
	  by: camsown, lmadsen, jtodd, qwell ........ ................

2010-10-18 19:50 +0000 [r292188]  Russell Bryant <russell@digium.com>

	* main/netsock2.c: Resolve some compiler errors in
	  ast_sockaddr_is_any(). These errors came up once this function
	  was used from within netsock2.c. The errors were like the
	  following: netsock2.c:393: error: dereferencing pointer
	  ‘({anonymous})’ does break strict-aliasing rules The usage of a
	  union here avoids this problem.

2010-10-18 19:16 +0000 [r292155]  David Vossel <dvossel@digium.com>

	* main/netsock2.c: Fixes build error for systems not supporting
	  IPV6_TCLASS.

2010-10-18 17:15 +0000 [r292122]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
	  Reported by: menschentier

2010-10-18  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-rc4 Released

2010-10-18 16:02 +0000 [r292085]  David Vossel <dvossel@digium.com>

	* main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
	  or IPv4 address. (closes issue #18099) Reported by: jamesnet
	  Patches: issues_18099_v3.diff uploaded by dvossel (license 671

2010-10-18 15:32 +0000 [r292083]  Jeff Peeler <jpeeler@digium.com>

	* pbx/pbx_spool.c: Disable use of inotify for call file handling as
	  it is not working properly. (related to #18089)

2010-10-16 10:47 +0000 [r292050]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
	  revisions 292049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) |
	  15 lines Base directory for MOH should be ASTDATADIR If the
	  directive 'directory' is relative, make it relative to the
	  datadir, rather than to the varlibdir. In the sample
	  configuration it is relative ('moh'). This has no effect unless
	  you have actively set the datadir explicitly (at build time or at
	  run time). (closes issue #16906) Patches: moh_datadir uploaded by
	  tzafrir (license 46) Review:
	  https://reviewboard.asterisk.org/r/974/ ........

2010-10-15 21:40 +0000 [r292016]  Terry Wilson <twilson@digium.com>

	* res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
	  session This avoids unhappy crashing when we try to 'core stop
	  gracefully' and res_srtp tries to unload before chan_sip does.
	  Thanks, Russell! (closes issue #18085) Reported by: st

2010-10-15 20:12 +0000 [r291942]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes peer's host port information being
	  lost on sip reload. (closes issue #18135) Reported by: lmadsen
	  Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
	  Tested by: lmadsen

2010-10-15 19:50 +0000 [r291940]  Paul Belanger <paul.belanger@polybeacon.com>

	* configs/gtalk.conf.sample, /: Merged revisions 291939 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
	  (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
	  15 Oct 2010) | 2 lines Clean up formatting. ........
	  ................

2010-10-15 16:39 +0000 [r291905]  Terry Wilson <twilson@digium.com>

	* res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
	  | 7 lines Don't crash or deadlock on module unload We can't hold
	  the lock while pthread_join is called since aji_log_hook will
	  attempt to lock from the other therad. We reorder the
	  pthread_join and ast_aji_disconnect so that we don't do an
	  SSL_read() while SSL_shutdown is running, causing a crash.
	  ........

2010-10-14 22:09 +0000 [r291827-291829]  David Vossel <dvossel@digium.com>

	* main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
	  options are set. (closes issue #18099) Reported by: jamesnet
	  Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
	  Tested by: dvossel, jamesnet

	* channels/chan_gtalk.c: Safer xml parsing, treat all clients the
	  same, and better local candidate selection. The gtalk channel
	  driver was doing several unsafe operations in regards to how it
	  parsed incoming XML messages. I have cleaned that code up so it
	  should be much safer now. We now treat all clients types the
	  same. We have no reason to distinguish between GMAIL and GOOGLE
	  VOICE clients anymore because they all work the same way. I also
	  modified how the local ip is found. If no bindaddress is provided
	  in the config file, we attempt to determine the local ip we would
	  use to connect to google.com. If that fails, then we fall back to
	  the ast_find_ourip() function as a last resort. Using the new
	  method makes it much less likely that we would ever advertise a
	  local RTP candidate as a loopback address.

2010-10-14 18:45 +0000 [r291791]  Jeff Peeler <jpeeler@digium.com>

	* main/stdtime/localtime.c: Add missing ifdefs for test framework
	  and new locale code. (closes issue #18137) Reported by: ovi
	  Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
	  (license 717) 18137_localelist_warning.patch uploaded by wdoekes
	  (license 717) Tested by: ovi

2010-10-14 15:15 +0000 [r291758]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_gtalk.c, channels/chan_jingle.c,
	  include/asterisk/acl.h, channels/chan_sip.c,
	  channels/chan_h323.c, main/acl.c: Add the ability for
	  ast_find_ourip to return IPv4, IPv6 or both. While testing
	  chan_gtalk I noticed jabber was using my IPv6 address and not
	  IPv4. When using bindaddr=0.0.0.0 it is possible for
	  ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
	  family parameter gives you the ablility to choose. Since
	  jabber/gtalk/h323 do not support IPv6, we should only return IPv4
	  results. Review: https://reviewboard.asterisk.org/r/973/

2010-10-14 12:08 +0000 [r291725]  Russell Bryant <russell@digium.com>

	* doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/

2010-10-13 23:45 +0000 [r291656]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/sig_analog.h: Merged revisions 291655 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
	  (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
	  | 20 lines Deadlock between dahdi_exception() and
	  dahdi_indicate(). There is a deadlock between dahdi_exception()
	  and dahdi_indicate() for analog ports. The call-waiting and
	  three-way-calling feature can experience deadlock if these
	  features are trying to do something and an event from the bridged
	  channel happens at the same time. Deadlock avoidance code added
	  to obtain necessary channel locks before attemting an operation
	  with call-waiting and three-way-calling. (closes issue #16847)
	  Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
	  uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
	  uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
	  uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
	  Review: https://reviewboard.asterisk.org/r/971/ ........
	  ................

2010-10-13 23:01 +0000 [r291581]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /: Merged revisions 291580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
	  (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
	  | 21 lines Don't ignore frames that have been queued when
	  softhangup'd When an outgoing call is answered and hung up by the
	  far end *very* quickly, we may not read any frames and therefor
	  end up with a call that displays the wrong
	  disposition/DIALSTATUS. The reason is because ast_queue_hangup()
	  immediately sets the _softhangup flag on the channel and then
	  queues the HANGUP control frame, but __ast_read refuses to read
	  any frames if ast_check_hangup() indicates that a hangup request
	  has been made (which it will if _softhangup is set). So, we end
	  up losing control frames. This change makes __ast_read continue
	  to read frames even if a soft hangup has been requested. It
	  queues a hangup frame to make sure that __ast_read() will still
	  eventually return NULL. Much thanks to David Vossel for all of
	  the reviews, discussion, and help! (closes issue #16946) Reported
	  by: davidw Review: https://reviewboard.asterisk.org/r/740/
	  ........ ................

2010-10-13 22:46 +0000 [r291578]  David Vossel <dvossel@digium.com>

	* channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
	  makes the xml parsing safer.

2010-10-13 22:24 +0000 [r291575]  Terry Wilson <twilson@digium.com>

	* Makefile, static-http/mantest.html (added): Add a simple AMI
	  client web page This patch uses the XML docs to parse all of the
	  available AMI commands and allows you to enter the command name
	  and be presented with a form with the available fields. You can
	  then rapidly tab through the fields and submit the command and
	  view the response. It is much faster/easier than having to use
	  telnet for testing purposes.

2010-10-13 20:21 +0000 [r291469-291541]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
	  for the first FAX call. The chan_dahdi faxdetect option only
	  works for the first call. After that the option no longer works.
	  The struct dahdi_pvt.callprogress member is the encoded user
	  config setting for the callprogress and faxdetect config options.
	  Changing this value alters the configuration for all following
	  calls until the chan_dahdi.conf file is reloaded. * Fixed the
	  chan_dahdi ast_channel_setoption callback to not change the users
	  faxdetect config setting except for the current call. * Fixed the
	  chan_dahdi ast_channel_queryoption callback to read the active
	  DSP setting of the faxdetect option. * Made actually disable the
	  active faxdetect DSP setting for the current call on the analog
	  port. my_handle_dtmfup() is used for normal analog ports.
	  dahdi_handle_dtmfup() is the legacy code and is no longer used
	  unless in a radio mode. (closes issue #18116) Reported by:
	  seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
	  (license 664) Review: https://reviewboard.asterisk.org/r/972/

	* channels/chan_misdn.c: Merged revision 291504 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
	  13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
	  ast_channel. Must get the ast_channel lock before proceeding with
	  release_chan() and release_chan_early() to hold off ast_hangup()
	  from destroying the ast_channel. Missed this change for -r291468.
	  JIRA ABE-2598 JIRA SWP-2317 ..........

	* channels/chan_misdn.c: Merge revision 291468 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
	  13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
	  call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
	  --> RELEASE_COMPLETE * Add lock protection around channel list
	  for find/add/delete operations. * Protect misdn_hangup() from
	  release_chan() and vise versa using the release_lock. JIRA
	  ABE-2598 JIRA SWP-2317 ..........

2010-10-13 15:46 +0000 [r291394]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291393 | russell | 2010-10-13 10:29:21 -0500
	  (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
	  | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
	  a frame. This fixes a crash due to a hangup race condition.
	  ABE-2601 ........ ................

2010-10-12 17:20 +0000 [r291284]  Leif Madsen <lmadsen@digium.com>

	* configs/phoneprov.conf.sample, /: Merged revisions 291280 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
	  | 7 lines Add undocumented variables to phoneprov.conf.sample
	  (closes issue #18107) Reported by: lathama Patches:
	  phoneprov.conf.sample.diff uploaded by lathama (license 1028)
	  ........

2010-10-12 17:06 +0000 [r291265]  Tilghman Lesher <tlesher@digium.com>

	* /, main/acl.c: Merged revisions 291264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
	  (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
	  Oct 2010) | 2 lines Oops, incorrect range (although unallocated
	  at ARIN) ........ ................

2010-10-12 16:08 +0000 [r291230]  Leif Madsen <lmadsen@digium.com>

	* configs/manager.conf.sample, /: Merged revisions 291229 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
	  | 2 lines Add documention that mentions options are defined but
	  not used. (Issue #18101) ........

2010-10-12 15:58 +0000 [r291192-291227]  David Vossel <dvossel@digium.com>

	* main/manager.c: Fixes manager.c crash. This issue was caused by
	  improper use of the mansession lock and manession_session lock.
	  These two structures are confusing to begin with so I'm not
	  surprised this occurred. I fixed this by consistently making sure
	  we use each of these locks only to protect the data in the
	  corresponding structure. We had mismatched usage of these locks
	  which resulted in no mutual exclusivity occurring at all. (closes
	  issue #17994) Reported by: vrban Patches:
	  mansession_locking_fix.diff uploaded by dvossel (license 671)
	  Tested by: vrban

	* CHANGES: Update CHANGES to reflect new gtalk.conf options.

	* channels/chan_gtalk.c, include/asterisk/stun.h,
	  configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
	  enhancements and general code cleanup. This patch includes
	  several chan_gtalk enhancements. Two new gtalk.conf options have
	  been added, externip and stunadd. Setting externip allows us to
	  manually specify what the external IP address is outside of a NAT
	  environment. Setting the stunaddr option to a valid stun server
	  allows for that external ip to be retrieved via a STUN server
	  automatically. This external IP is then advertised during call
	  setup as a possible candidate. I have also attempted to clean up
	  chan_gtalk's code so it meets our coding guidelines. During this
	  cleanup I noticed several things that need to be done in the code
	  and made a TODO section at the top of the file.

2010-10-11 18:51 +0000 [r291075-291113]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_sip.c: Move declaration closer to where now used.

	* /, channels/chan_sip.c: Merged revisions 291110-291111 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
	  (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
	  Oct 2010) | 1 line Add missing unlock to an exception condition
	  in reload_config(). ........ ................ r291111 | rmudgett
	  | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
	  from handle_request_do() consistent. ................

	* main/cli.c, /: Merged revisions 291073 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
	  | 15 lines Fixed infinite loop in verbose/debug message output.
	  Setting the module/filename specific message level and then
	  changing it resulted in the linked list being looped on itself.
	  Traversing this linked list is an infinite loop if what you are
	  looking for is not in the list. Also plugged some CLI parsing
	  holes in the associated CLI command: * Removing a nonexistent
	  module from the list actually added it with a level of zero. *
	  Setting the non-module specific level to zero is now equivalent
	  to setting it to "off" as documented. ........

2010-10-09 23:25 +0000 [r291038]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
	  option to set calls to be logged in GMT/UTC.

2010-10-09 15:00 +0000 [r291005-291037]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.c: small correction for verbose
	  print h.323 packets

	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
	  options per user and peer. Added options for faststart/h.245
	  tunneling per user/peer, properly handle these and global
	  options, correction of handling fs/tunneling fields in signalling
	  responses (issue #17972) Reported by: salecha Patches:
	  fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
	  Tested by: may213, salecha

2010-10-08 20:44 +0000 [r290973]  David Vossel <dvossel@digium.com>

	* channels/chan_gtalk.c: Make outbound Google Voice calls. This
	  patch allows for outbound Google Voice calls to be dialed from
	  Asterisk using chan_gtalk. Below is an example dialstring. exten
	  -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In
	  this example, 'asterisk' is the jabber.conf profile configured to
	  connect to your gmail account. In order to receive Google Voice
	  calls make sure to enable 'allowguest=yes' in gtalk.conf.

2010-10-08 15:49 +0000 [r290937-290938]  Erin Spiceland <erin@thespicelands.com>

	* addons/res_config_mysql.c: Parentheses around assignment used as
	  truth value, introduced in r290937.

	* addons/res_config_mysql.c, addons/app_mysql.c,
	  configs/res_config_mysql.conf.sample: Add option to
	  res_config_mysql and app_mysql to specify a character set that
	  MySQL should use. (closes issue 17948) Reported by qmax.

2010-10-08 02:56 +0000 [r290864]  Jeff Peeler <jpeeler@digium.com>

	* main/asterisk.c, /: Merged revisions 290863 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
	  (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
	  | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
	  at control console. A recent change was made to avoid a race
	  condition on shutdown which only called the end functions from
	  the console thread. However, when pressing Ctrl-C the quit
	  handler is called from the signal handler thread. (closes issue
	  #17698) Reported by: jmls ........ ................

2010-10-07 22:38 +0000 [r290828-290829]  David Vossel <dvossel@digium.com>

	* channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
	  list. Philippe has made some notable contributions to the gtalk
	  channel driver. His name deserves to be listed amoung the authors
	  of that file. Thanks Philippe!

	* channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
	  There was a problem with how the candidates were being built on
	  an outbound call. This patch fixes that.

2010-10-07 20:58 +0000 [r290752]  Jason Parker <jparker@digium.com>

	* autoconf/ast_ext_lib.m4, /, configure,
	  include/asterisk/autoconfig.h.in: Merged revisions 290751 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
	  (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
	  9 lines Allow PRI to build properly when using --with-pri. Use
	  the directories found for the parent when using lib dependencies.
	  (closes issue #17314) Reported by: tzafrir Patches:
	  17314-withdeps.diff uploaded by qwell (license 4) ........
	  ................

2010-10-07  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-rc3 Released.

2010-10-07 11:00 +0000 [r290713]  Russell Bryant <russell@digium.com>

	* main/pbx.c, /: Merged revisions 290712 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
	  | 4 lines Don't crash when Set() is called without a value.
	  Review: https://reviewboard.asterisk.org/r/949/ ........

2010-10-06 21:22 +0000 [r290648-290674]  David Vossel <dvossel@digium.com>

	* channels/chan_gtalk.c: Fixes commented out code to use #if 0
	  instead. Thanks to rmudgett for catching this!

	* channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
	  properly. Outbound DTMF with gtalk needs to be done within the
	  RTP stream. I discovered this after investigating a packet
	  capture from the gmail client. Instead of performing jingle
	  signaling DTMF, the gtalk servers expect all DTMF to arrive on
	  the RTP stream using RFC2833 way of doing things. Chan_gtalk also
	  had an issue with negotiating RTP payload type 106 for the
	  telephony-event and then sending DTMF as payload 101. This has
	  been resolved by always negotiating 101 as the payload type like
	  we do everywhere else. With this patch, incoming google voice
	  calls forwarded to Asterisk via gtalk work.

2010-10-06 18:50 +0000 [r290614]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c: Merged revision 290613 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
	  06 Oct 2010) | 5 lines Eliminate a redundant test for
	  AST_CONTROL_REDIRECTING. Eliminate redundant test for
	  AST_CONTROL_REDIRECTING that prevents running the redirecting
	  interception macro if it is defined. ..........

2010-10-06 13:49 +0000 [r290576]  Tilghman Lesher <tlesher@digium.com>

	* /, main/file.c: Merged revisions 290575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
	  | 8 lines Allow streaming audio from a pipe. (closes issue
	  #18001) Reported by: jamicque Patches:
	  20100926__issue18001.diff.txt uploaded by tilghman (license 14)
	  Tested by: jamicque ........

2010-10-06 04:35 +0000 [r290542]  Terry Wilson <twilson@digium.com>

	* res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
	  is null It is possible for ast_rtp_stop() to be called which will
	  clear the remote address and cause the sendto to fail and spam
	  warnings. Don't send in this case.

2010-10-05 22:23 +0000 [r290479-290506]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
	  set debug peer' option.

	* include/asterisk/jingle.h, channels/chan_gtalk.c,
	  res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
	  work with gmail client This patch was written by Philippe Sultan
	  (phsultan). Thanks for keeping this up to date!

2010-10-05 20:23 +0000 [r290408]  Tilghman Lesher <tlesher@digium.com>

	* res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
	  (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
	  | 8 lines Fix a crash by ensuring that we don't alter memory
	  after it's freed. (closes issue #17387) Reported by: jmls
	  Patches: 20100726__issue17387.diff.txt uploaded by tilghman
	  (license 14) Tested by: jmls ........ ................

2010-10-05 20:09 +0000 [r290376-290378]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Resolves dnsmgr memory corruption in
	  chan_iax2. (closes issue #17902) Reported by: afried Patches:
	  issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
	  afried, russell, dvossel Review:
	  https://reviewboard.asterisk.org/r/965/

	* /, apps/app_directed_pickup.c: Merged revisions 290375 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
	  | 10 lines Fixes PickupChan() not working with full channel name.
	  (closes issue #18011) Reported by: schern Patches:
	  app_directed_pickup.c.2.patch uploaded by schern (license 995)
	  app_directed_pickup.c.trunk.patch uploaded by schern (license
	  995) Tested by: schern, dvossel ........

2010-10-05 14:15 +0000 [r290066-290289]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Restore run directory for OS X, as well
	  as standardizing some other paths to Mac OS X.

	* pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
	  pbx/ael/ael-test/ref.ael-vtest17, /,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
	  Merged revisions 290254 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
	  | 11 lines Change new pattern matcher to regard dashes the same
	  as the old pattern matcher -- as visual candy to be ignored. Also
	  change the AEL parser to not generate dashes within extensions,
	  as those dashes would be ignored. Update the AEL tests to match
	  this behavior. (closes issue #17366) Reported by: murf Patches:
	  20100727__issue17366.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman ........

	* /, configure, configure.ac: Merged revisions 290201 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
	  (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
	  Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
	  ................

	* /, configure, configure.ac: Merged revisions 290101 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
	  (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
	  Oct 2010) | 2 lines Automatically re-run configure test for
	  menuselect, when the relevant makeopts settings change. ........
	  ................

	* pbx/pbx_spool.c: Get notification only when file is closed, not
	  when created. (closes issue #17924) Reported by: mkeuter Patches:
	  asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
	  Tested by: abelbeck

2010-10-02 17:57 +0000 [r290026]  Kevin P. Fleming <kpfleming@digium.com>

	* contrib/scripts/get_mp3_source.sh: Allow users to pass additional
	  arguments to the Subversion command that obtains the MP-3 source
	  code. (reported on IRC by jmls)

2010-10-02 08:56 +0000 [r289951]  Olle Johansson <oej@edvina.net>

	* main/manager.c, /: Merged revisions 289950 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
	  02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
	  lines Add documentation for undocumented option to AMI action
	  originate ........ ................

2010-10-02 04:46 +0000 [r289875]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
	  (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
	  | 8 lines When forwarding a message, a prepend means that the
	  filesystem will always have a better copy. (closes issue #17803)
	  Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
	  uploaded by tilghman (license 14) Tested by: dpetersen ........
	  ................

2010-10-02 02:43 +0000 [r289840]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
	  main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
	  289798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
	  (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
	  | 15 lines Change RFC2833 DTMF event duration on end to report
	  actual elapsed time. The scenario here is with a non P2P early
	  media session. The reported time length of DTMF presses are
	  coming up short when sending to the remote side. Currently the
	  event duration is a running total that is incremented when
	  sending continuation packets. These continuation packets are only
	  triggered upon incoming media from the remote side, which means
	  that the running total probably is not going to end up matching
	  the actual length of time Asterisk received DTMF. This patch
	  changes the end event duration to be lengthened if it is detected
	  that the end event is going to come up short. Review:
	  https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
	  ................

2010-10-01 17:19 +0000 [r289718]  Paul Belanger <paul.belanger@polybeacon.com>

	* res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
	  289704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
	  (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
	  2010) | 6 lines Disable debugging by default and reformat .config
	  file. Review: https://reviewboard.asterisk.org/r/929/ ........
	  ................

2010-10-01 16:22 +0000 [r289701]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
	  (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
	  | 14 lines Ensure user portion of SIP URI matches dialplan when
	  using encoded characters. This commit takes a simliar approach to
	  288112 and checks the dialplan to determine the proper action for
	  an incoming contact header as to whether or not it should be
	  decoded or not. sip_new was blindly always decoding the
	  extension, which also caused the outgoing contact header to be
	  incorrect as well as failing to match the encoded extension in
	  the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
	  bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
	  wdoekes ........ ................

2010-10-01 09:42 +0000 [r289622]  Stefan Schmidt <sst@sil.at>

	* channels/chan_sip.c: don't iterate through all dialogs to find
	  and delete old subscribes On every incoming subscribe there is a
	  iteration through all dialogs to find old subscribes and delete
	  them. This is slow and not RFC conform. This was only needed in
	  1.2 cause a subscribe was not deleted when a dialog was
	  destroyed, after 1.4 a subscribe get removed when its dialog is
	  destroyed. (closes issue #17950) Reported by: schmidts Tested by:
	  schmidts Review: https://reviewboard.asterisk.org/r/901/

2010-09-30 20:23 +0000 [r289581]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Solaris fixes.

2010-09-30 19:53 +0000 [r289554]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
	  2010) | 4 lines Properly handle channel allocation failures duing
	  invites with replaces. ABE-2588 ........

2010-09-30 19:28 +0000 [r289549]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Merged revision 289547 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
	  30 Sep 2010) | 10 lines In chan_misdn, the
	  DivertingLegInformation2 DivertingNr is garbage when the number
	  is restricted. The same thing happens with
	  DivertingLegInformation1 DivertedTo number. The
	  misdn_PresentedNumberUnscreened_extract() extracted the
	  Unscreened PartyNumber field unconditionally. It now checks the
	  presented number unscreened type to see if the PartyNumber was
	  even present. JIRA ABE-2595 ..........

2010-09-30 17:50 +0000 [r289543]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/localtime.h, main/stdtime/localtime.c,
	  tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
	  Solaris compatibility fixes

2010-09-30 15:39 +0000 [r289426]  Russell Bryant <russell@digium.com>

	* apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289425 | russell | 2010-09-30 10:37:29 -0500
	  (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
	  | 8 lines Fix a crash in app_sms. Since the data being passed to
	  the generator callback is on the stack of the SMS() application,
	  we must ensure that the generator is stopped before the
	  application exits. ABE-2587 ........ ................

2010-09-29 21:12 +0000 [r289340]  Jason Parker <jparker@digium.com>

	* main/channel.c, /, main/features.c: Merged revisions 289339 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
	  (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
	  8 lines Allow a manager originate to succeed on forwarded
	  devices. The timeout to wait for an answer was being set to 0
	  when a device forwarded to another extension. We don't always
	  need the timeout set like this, so make it an optional parameter,
	  and don't use it in this case. ABE-2544 ........ ................

2010-09-29 20:27 +0000 [r289336]  Leif Madsen <lmadsen@digium.com>

	* configs/res_ldap.conf.sample, /: Merged revisions 289334 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
	  | 1 line Update sample documentation to note md5secret
	  requirements. ........

2010-09-29 20:20 +0000 [r289333]  Russell Bryant <russell@digium.com>

	* res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
	  Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP
	  if the value does not begin with {md5}. This fixes a problem that
	  lmadsen ran in to where md5secret was not working for him.
	  ........

2010-09-29 17:53 +0000 [r289268-289300]  Matthew Nicholson <mnicholson@digium.com>

	* configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
	  file

	* main/channel.c: Update the CDR record when
	  ast_channel_set_caller_event() is called (related to issue
	  #17569) Reported by: tbelder

2010-09-29 16:16 +0000 [r289253]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Make development error message indicate which
	  channel.

2010-09-29 15:04 +0000 [r289179]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, /: Merged revisions 289178 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
	  (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
	  2010) | 8 lines Set the caller id on CDRs when it is set on the
	  parent channel. (closes issue #17569) Reported by: tbelder
	  Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
	  tbelder ........ ................

2010-09-28 18:18 +0000 [r289104]  Tilghman Lesher <tlesher@digium.com>

	* makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
	  configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
	  configure.ac: Solaris compatibility fixes Review:
	  https://reviewboard.asterisk.org/r/942/

2010-09-28 18:18 +0000 [r289099]  Brett Bryant <bbryant@digium.com>

	* main/channel.c, /: Merged revisions 289095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
	  (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
	  | 14 lines Fixes an issue with the Newchannel AMI event during
	  the Masquerading process. Fixes an issue with the Newchannel AMI
	  event during the Masquerading process, where no Newchannel AMI
	  event was generated for the psuedo channel used during the
	  masquerading process. (closes issue #17987) Reported by:
	  RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
	  (license 1122) Tested by: RadicAlish Review:
	  https://reviewboard.asterisk.org/r/937/ ........ ................

2010-09-28 01:04 +0000 [r289054-289057]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
	  messages. Deadlock avoidance for the owner channel was not done
	  when processing incoming AOC-E messages.

	* channels/sig_pri.c: Revert stuff not ready for commit in
	  -r289054.

	* channels/sig_pri.c, channels/chan_sip.c: Break up long
	  ast_manager_event_multichan() event lines.

2010-09-27 18:37 +0000 [r288961]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Still build SIP, even if res_crypto cannot
	  be built (use, not depend). (closes issue #18062) Reported by: a
	  user on the mailing list

2010-09-27 13:03 +0000 [r288925-288927]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Fix some documentation typos and spelling errors.

	* res/res_agi.c: Fix a documentation spelling error.

2010-09-24 17:58 +0000 [r288821-288852]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Append Retry-After header on 500 error
	  response to Re-INVITE according to RFC3261 section 14.2. ABE-2301

	* channels/chan_sip.c: Inspect Require header on BYE transaction
	  according to RFC3261 section 8.2.2.3. ABE-2293

2010-09-24 16:02 +0000 [r288748]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c, /: Merged revisions 288747 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288747 | twilson | 2010-09-24 08:37:39 -0700
	  (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
	  | 5 lines Don't fail a masquerade if it is already being hung up
	  This avoids noise on some Local channel situations where we don't
	  use /n. Thanks to Alec Davis for the suggestion. ........
	  ................

2010-09-24 13:54 +0000 [r288606-288713]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
	  Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
	  #18041) Reported by: asgaroth ........

	* main/asterisk.exports.in: Export timersub for platforms which do
	  not have it

	* include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
	  include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
	  main/strcompat.c, configure.ac: Merged revisions 288637 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
	  (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
	  Sep 2010) | 2 lines Solaris compatibility fixes ........
	  ................

	* CHANGES: Add note about the checkhangup option of ${CHANNEL()}

2010-09-23  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-rc2 Released.

2010-09-23 18:05 +0000 [r288507-288572]  Terry Wilson <twilson@digium.com>

	* main/manager.c: Make AMI honor enabled=no (closes issue #18040)
	  Reported by: twilson Review:
	  https://reviewboard.asterisk.org/r/938/

	* channels/chan_local.c, /: Merged revisions 288500 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288500 | twilson | 2010-09-22 16:10:09 -0700
	  (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
	  | 8 lines Don't let a Local channel get bridged to itself If a
	  local channel gets bridged to itself, it becomes orphaned with no
	  devices left to actually tell it to hang up. This patch modifies
	  local_fixup() to detect this case and deny it. Review:
	  https://reviewboard.asterisk.org/r/934 ........ ................

2010-09-22  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-rc1 Released.

2010-09-22 17:49 +0000 [r288345-288418]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
	  (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
	  | 5 lines RFC3261 section 12.2 explicitly says out of order
	  requests are responded with a 500 Server Internal Error response.
	  ABE-2458 ........ ................

	* /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
	  (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
	  Sep 2010) | 2 lines During check_pendings, if the dialog is
	  terminated with a CANCEL, change the invitestate to INV_CANCEL
	  like in sip_hangup. ........ ................

2010-09-22 16:45 +0000 [r288341]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, /: Merged revisions 288340 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288340 | russell | 2010-09-22 11:44:13 -0500
	  (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
	  | 11 lines Fix a 100% CPU consumption problem when setting
	  console=yes in asterisk.conf. The handling of -c and console=yes
	  should be the same, but they were not. When you specify -c, it
	  sets both a flag for console module and for asterisk not to
	  fork() off into the background. The handling of console=yes only
	  set console mode, so you would end up with a background process()
	  trying to run the Asterisk console and freaking out since it
	  didn't have anything to read input from. Thanks to beagles for
	  reporting and helping debug the problem! ........
	  ................

2010-09-22 15:14 +0000 [r288268]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
	  Merged revisions 288267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
	  (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
	  | 9 lines Allow the encoding to be set, in case local charset
	  does not agree with database. (closes issue #16940) Reported by:
	  jamicque Patches: 20100827__issue16940.diff.txt uploaded by
	  tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
	  uploaded by tilghman (license 14) Tested by: jamicque ........
	  r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
	  | 5 lines Document addition of encoding parameter. (issue #16940)
	  Reported by: jamicque ........ ................

2010-09-22 00:06 +0000 [r288194]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
	  (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
	  | 26 lines In chan_iax2.c:schedule_delivery() calls
	  ast_bridged_channel() on an unlocked channel. Near the beginning
	  of schedule_delivery(), ast_bridged_channel() is called on
	  iaxs[fr->callno]->owner. However, the channel is not locked,
	  which can result in ast_bridged_channel() crashing should
	  owner->tech change to a technology that doesn't implement
	  bridged_channel. I also fixed the other calls to
	  ast_bridged_channel() in chan_iax2.c since the owner lock was not
	  held there either. Converted the existing channel deadlock
	  avoidance to use iax2_lock_owner(). Using the new function
	  simplified some awkward code. In the process of fixing the
	  locking on ast_bridged_channel(), I also found a memory leak in
	  socket_process() for v1.6.2 and v1.8. The local struct variable
	  ies.vars is not freed on early/abnormal function exits. (closes
	  issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
	  uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
	  uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
	  uploaded by rmudgett (license 664) Review:
	  https://reviewboard.asterisk.org/r/926/ ........ ................

2010-09-21 22:57 +0000 [r288159]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
	  (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
	  | 15 lines Try both the encoded and unencoded subscription URI
	  for a match in hints. When a phone sends an encoded URI for a
	  subscription, the URI is not matched with the actual hint that is
	  in decoded format. For example, if we have an extension with a
	  hint that is named: "#5601" or "*5601", the subscription will
	  work fine if the phone subscribes with an already decoded URI,
	  but when it's decoded like "%255601" or "%2A5601", Asterisk is
	  unable to match it with the correct hint. (closes issue #17785)
	  Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
	  uploaded by tilghman (license 14) Tested by: ramonpeek ........
	  ................

2010-09-21 22:26 +0000 [r288157]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
	  21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
	  issue #18019) Reported by: Netview Patches: issue_0018019.patch
	  uploaded by pabelanger (license 224) Tested by: Netview ........

2010-09-21 21:03 +0000 [r288079-288082]  Richard Mudgett <rmudgett@digium.com>

	* doc/tex/partymanip.tex: Add note in party manipulation chapter on
	  interception macros.

	* apps/app_queue.c, apps/app_dial.c: Simplify locking code for
	  REDIRECTING interception macro when forwarding a call. Simplified
	  the locking code by using a local copy of the redirecting party
	  information in app_dial.c:do_forward() and
	  app_queue.c:wait_for_answer() for launching the REDIRECTING
	  interception macro when a call is forwarded. Reduced the lock
	  time of the 'o->chan' and 'in' channels.

	* main/channel.c: Protect channel access in CONNECTED_LINE and
	  REDIRECTING interception macro launch code.

2010-09-21 19:48 +0000 [r288007]  Brett Bryant <bbryant@digium.com>

	* main/channel.c, /: Merged revisions 288006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
	  (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
	  | 8 lines Add a check to fix a rare segmentation fault you'd get
	  if ast_frdup couldn't allocate memory on the first frame being
	  queued in ast_queue_frame. (closes issue #17882) Reported by:
	  seanbright Tested by: seanbright ........ ................

2010-09-21 19:08 +0000 [r287935]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /: Merged revisions 287934 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
	  (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
	  Sep 2010) | 2 lines Less than zero is an error, not any non-zero
	  value. ........ ................

2010-09-21 19:02 +0000 [r287931]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Revert change in favor of a more targeted fix

2010-09-21 18:32 +0000 [r287929]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Send a "415 Unsupported Media Type" after
	  failure to process sdp due to unknown Content-Encoding header.
	  ABE-2258

2010-09-21 15:53 +0000 [r287897]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Cut-n-paste error in builtin_blindtransfer().

2010-09-21 15:43 +0000 [r287895]  Russell Bryant <russell@digium.com>

	* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
	  main/acl.c: Don't use ast_strdupa() from within the arguments to
	  a function. (closes issue #17902) Reported by: afried Patches:
	  issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
	  russell Review: https://reviewboard.asterisk.org/r/927/

2010-09-21 15:24 +0000 [r287893]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
	  prefix. (closes issue #17981) Reported by: avalentin Patches:
	  sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
	  (plus an additional fix by me) Tested by: avalentin

2010-09-21 13:41 +0000 [r287863]  Russell Bryant <russell@digium.com>

	* main/logger.c: Fix a regression in verbose logger processing.

2010-09-21 04:37 +0000 [r287833]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Don't generate connected line buffer twice for
	  comparison

2010-09-21 00:00 +0000 [r287760]  Brett Bryant <bbryant@digium.com>

	* /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
	  (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
	  | 16 lines Fix misvalidation of meetme pins in conjunction with
	  the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
	  user and admin pin setup for your conference, using the user pin
	  would gain you admin priviledges. Also, when no user pin was set,
	  an admin pin was, the 'a' MeetMe flag wasn't used, and the user
	  tried to enter a conference then they were still prompted for a
	  pin and forced to hit #. (closes issue #17908) Reported by: kuj
	  Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
	  kuj Review: [full review board URL with trailing slash] ........
	  ................

2010-09-20 23:51 +0000 [r287757]  Terry Wilson <twilson@digium.com>

	* main/channel.c: Avoid infinite loop with certain local channel
	  connected line updates Compare connected line data before sending
	  a connected line indication to avoid possible loops. Review:
	  https://reviewboard.asterisk.org/r/932/

2010-09-20 23:20 +0000 [r287701]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/channel.c, /: Merged revisions 287685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
	  2010) | 18 lines ast_channel_masquerade: Avoid recursive
	  masquerades. Check all 4 combinations of (original/clonechan) *
	  (masq/masqr). Initially original->masq and clonechan->masqr were
	  only checked. It's possible with multiple masq's planned - and
	  not yet executed, that the 'original' chan could already have
	  another masq'd into it - thus original->masqr would be set, that
	  masqr would lost. Likewise for the clonechan->masq. (closes issue
	  #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
	  based on bug16057.diff4.txt uploaded by alecdavis (license 585)
	  Tested by: ramonpeek, davidw, alecdavis ........

2010-09-20 23:14 +0000 [r287683]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
	  struct if the port is initially in alarm. Fixed initial inalarm
	  value for sig_analog ports. Along with -r261007, this gets the
	  inalarm flag in sync with chan_dahdi for sig_analog ports.
	  (closes issue #16983)

2010-09-20 22:21 +0000 [r287661]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/channel.c: ast_do_masquerade. Keep channels ao2_container
	  locked while unlink and linking channels. Previously, Masquerade
	  would unlock 'original' and 'clonechan' and allow another masq
	  thread to run. End result would be corrupted memory, and the
	  frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
	  Reported by: notthematrix Patches: Based on bug17801.diff1.txt
	  uploaded by alecdavis (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/928

2010-09-20 22:09 +0000 [r287645-287647]  David Vossel <dvossel@digium.com>

	* include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
	  (added), main/channel.c, main/framehook.c (added),
	  funcs/func_frame_trace.c (added): Addition of the FrameHook API
	  (AKA AwesomeHooks) So far all our tools for viewing and
	  manipulating media streams within Asterisk have been entirely
	  focused on audio. That made sense then, but is not scalable now.
	  The FrameHook API lets us tap into and manipulate _ANY_ type of
	  media or signaling passed on a channel present today or in the
	  future. This tool is a step in the direction of expanding
	  Asterisk's boundaries and will help generate some rather
	  interesting applications in the future. In addition to the
	  FrameHook API, a simple dialplan function exercising the api has
	  been included as well. This function is called FRAME_TRACE().
	  FRAME_TRACE() allows for the internal ast_frames read and written
	  to a channel to be output. Filters can be placed on this function
	  to debug only certain types of frames. This function could be
	  thought of as an internal way of doing ast_frame packet captures.
	  Review: https://reviewboard.asterisk.org/r/925/

	* channels/chan_sip.c: Fixes issue with registrations not working
	  properly with pedantic=yes. (closes issue #18017) Reported by:
	  schmidts Patches: issues_18017_v1.diff uploaded by dvossel
	  (license 671) Tested by: schmidts

2010-09-20 21:29 +0000 [r287643]  Jason Parker <jparker@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
	  2010) | 8 lines Don't crash when parking a non-bridged call.
	  (closes issue #17680) Reported by: jmhunter Patches:
	  chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
	  jmhunter, DEA ........

2010-09-20 21:19 +0000 [r287639]  Brett Bryant <bbryant@digium.com>

	* main/logger.c: Fixes an error with the logger that caused verbose
	  messages to be spammed to the screen if syslog was configured in
	  logger.conf (closes issue #17974) Reported by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/915/

2010-09-20 15:57 +0000 [r287559]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 287558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
	  (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
	  state changes Merged revisions 287555 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
	  2010) | 5 lines Use ast_dynamic_str when processing hint state
	  changes (related to issue #17928) Reported by: mdu113 ........
	  ................

2010-09-19 16:09 +0000 [r287471]  Olle Johansson <oej@edvina.net>

	* main/manager.c, /: Merged revisions 287470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
	  19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
	  lines Make sure we always free variables properly in manager
	  originate. (closes issue #17891) reported, solved and tested by
	  oej Review: https://reviewboard.asterisk.org/r/869/ ........
	  ................

2010-09-17 21:08 +0000 [r287388]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
	  (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
	  | 7 lines Blank columns should get set on reload, not ignored.
	  (closes issue #16893) Reported by: haakon Patches:
	  20100818__issue16893.diff.txt uploaded by tilghman (license 14)
	  ........ ................

2010-09-17 13:37 +0000 [r287309]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 287308 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
	  (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
	  2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
	  processing in ast_hint_state_changed(). (related to issue #17928)
	  Reported by: mdu113 ........ ................

2010-09-17 08:44 +0000 [r287269-287271]  Jan Kalab <pitlicek@gmail.com>

	* res/res_calendar_ews.c: Events are visible after they were
	  removed from EWS calendar Because we must merge calendar even
	  when it's empty. (closes issue #17786)

	* res/res_calendar_ews.c: Asterisk crashing because of double free
	  when EWS request fails The free is done later in code. I think
	  ast_free() should have built in checks for double free. (closes
	  issue #17782)

	* res/res_calendar_caldav.c, res/res_calendar_ews.c,
	  res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
	  Support for HTTP redirects in calendar's URL libneon does not
	  support HTTP redirects (3xx responses) by default. You must tell
	  it to follow them. Also, another little unsigned int fix. (closes
	  issue #17776) Review: https://reviewboard.asterisk.org/r/921/

2010-09-16 22:04 +0000 [r287195]  Jason Parker <jparker@digium.com>

	* contrib/init.d/rc.debian.asterisk: Don't fail when running the
	  Debian init script directly (as one would normally do). readlink
	  apparently returns 1 when the arg isn't a symlink, which caused
	  the script to exit. (closes issue #17910) Reported by: wurstsalat

2010-09-16 21:57 +0000 [r287193]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
	  the default for "autofill" and "shared_lastcall" to "yes" in
	  queues.conf. Review: https://reviewboard.asterisk.org/r/922/

2010-09-16 20:07 +0000 [r287116-287120]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 287119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
	  (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
	  2010) | 8 lines Don't limit hint processing in
	  ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
	  (closes issue #17928) Reported by: mdu113 Patches:
	  20100831__issue17928.diff.txt uploaded by tilghman (license 14)
	  Tested by: mdu113 ........ ................

	* main/cdr.c, /: Merged revisions 287115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
	  (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
	  2010) | 8 lines Don't stop printing cdr variables if we encounter
	  one with a blank name or value. (closes issue #17900) Reported
	  by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
	  mnicholson (license 96) Tested by: mnicholson ........
	  ................

2010-09-15 22:17 +0000 [r287056]  Terry Wilson <twilson@digium.com>

	* res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
	  Also make it more obvious when there is an issue en/decrypting.
	  (closes issue #17563) Reported by: Alexcr Patches:
	  res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
	  twilson

2010-09-15 20:58 +0000 [r287020]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: fix uninintialized variable

2010-09-15 20:53 +0000 [r287017]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
	  revision 287014 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
	  15 Sep 2010) | 58 lines The handling of call transfer signaling
	  for mISDN PTMP is not fully implemented. The handling of call
	  transfer signaling for mISDN PTMP is not fully implemented. The
	  signaling of number updates with ISDN/DSS1 ECT supplementary
	  services (ETS 300 369-1) comes along with a notification
	  indicator IE and redirection number IE for PTMP. The
	  implementation in the current Asterisk mISDN channel
	  unfortunately can handle these information elements only in a
	  NOTIFY message. These information elements are also signaled in a
	  FACILTY message with a RequestSubaddress facility, when the
	  subscriber is already in the active state (see 9.2.4 and 9.2.5 of
	  ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
	  handle the notification indicator IE and redirection number IE
	  with the RequestSubaddress facility. * Made
	  misdn_update_connected_line() send a NOTIFY message if Asterisk
	  originated the call and it is not connected yet. * Made
	  misdn_update_connected_line() send a FACILITY message if the call
	  is already connected. This patch requires the presence of the
	  associated mISDN patches to compile. I had to enhance mISDN to
	  allow the notification indicator IE and the redirection number IE
	  to be used with a FACILITY message. Earlier versions of the
	  Digium enhanced mISDN are no longer going to work. **********
	  abe_2526_misdn.patch * Made an incoming FACILITY message allow
	  the presence of the notification indicator IE and the redirection
	  number IE. ********** abe_2526_misdnuser_v3.patch * Added support
	  to send and receive a FACILITY message with the notification
	  indicator IE and the redirection number IE. * Added the ability
	  to send a NOTIFY message in PTMP/NT mode to all responding
	  subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
	  abe_2526_ast.patch uploaded by rmudgett (license 664)
	  abe_2526_misdn.patch uploaded by rmudgett (license 664)
	  abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
	  Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
	  ..........

2010-09-15 20:32 +0000 [r286931-287015]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
	  (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
	  | 7 lines Ensure mailbox is not filled to capacity before doing
	  message forwarding. Specifically, before prompting to record a
	  prepended message the capacity is checked first. If the mailbox
	  is full the extension will be reprompted. ABE-2517 ........
	  ................

	* CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
	  configs/features.conf.sample, channels/chan_mgcp.c,
	  include/asterisk/features.h, channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
	  parking extension for non-default parking lots. This is a new
	  feature that allows for parking to custom parking lots to be
	  accessed directly, rather than with channel variables or by
	  changing the default parking lot. The extension is set with the
	  parkext option just as the default parking lot is done. Also, the
	  manager action has been updated to optionally allow a specified
	  parking lot. (closes issue #14882) Reported by: vmikhnevych
	  Patches: patch_14882.txt uploaded by mnick (license 874) modified
	  by me Review: https://reviewboard.asterisk.org/r/884/

2010-09-15 18:29 +0000 [r286904-286905]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_analog.c: Simplify some code in sig_analog.

	* channels/sig_analog.c: Unable to originate calls using E&M over
	  T1. When originating a call from Unit Under Test to Reference
	  Unit using E&M RBS signaling mode, I get the following warning
	  message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
	  the sig_analog outgoing flag. It was never set when sig_analog
	  was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408

2010-09-15 13:05 +0000 [r286868]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Set tohost to the domain specified in the
	  configuration file instead of the IP address of the host we are
	  calling. This fixes a regression introduced in r274783. (closes
	  issue #17960) Reported by: adriavidal Patches:
	  sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
	  by: mich, mnicholson, adriavidal (closes issue #17676) Reported
	  by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
	  (license 96) Tested by: mnicholson

2010-09-14 21:57 +0000 [r286834]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Sets subscribed type for outgoing MWI
	  subscriptions so correct Event header is used.

2010-09-14 19:28 +0000 [r286682-286758]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
	  (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
	  2010) | 13 lines Don't clear the username from a realtime
	  database when a registration expires. Non-realtime chan_sip does
	  not clear the username from memory when a registration expiries
	  so realtime probably shouldn't either. (closes issue #17551)
	  Reported by: ricardolandim Patches:
	  reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
	  96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
	  (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
	  mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
	  uploaded by mnicholson (license 96) Tested by: ricardolandim,
	  mnicholson ........ ................

	* main/channel.c, /: Merged revisions 286681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
	  (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
	  2010) | 7 lines Only drop duplicate answer frames if the channel
	  is bridged. Back in r3710 ast_read() was modified to drop answer
	  frames on channels that were in the UP state. This modification
	  prevented bridges that were up before the answer from being
	  broken and reestablished by an ANSWER control frame. That change
	  also prevents pickup of channels called from the ast_dial
	  framework from working properly. The ast_dial framework expects
	  to see an ANSWER frame after dialing and the pickup code queues
	  one but ast_read() drops it. This new change only drops ANSWER
	  frames when the channel is bridged, allowing the answer queued by
	  the pickup code to properly pass through ast_read() on to the
	  ast_dial framework. ABE-2473 (related to issue #2342) ........
	  ................

2010-09-14 15:30 +0000 [r286647]  Richard Mudgett <rmudgett@digium.com>

	* doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
	  documented CONNECTED_LINE and REDIRECTING party manipulation
	  macro names.

2010-09-14 06:55 +0000 [r286617]  Jan Kalab <pitlicek@gmail.com>

	* res/res_calendar_ews.c: Merging events for Exchange web service
	  doesn't work as expected, resulting in only one event in calendar
	  The solution is to use "global" counter of events, since we do
	  new requests for every event and calendar sync after every
	  request. So now we do sync only after last request. (closes issue
	  #17877) Review: https://reviewboard.asterisk.org/r/916/

2010-09-14 05:07 +0000 [r286528-286588]  Tilghman Lesher <tlesher@digium.com>

	* contrib/realtime/mysql/voicemail_data.sql (added), /,
	  contrib/realtime/mysql/voicemail_messages.sql (added): Merged
	  revisions 286587 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
	  | 2 lines Add documentation on missing backend tables for
	  Voicemail ........

	* /, main/features.c: Merged revisions 286557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
	  | 2 lines C precedence got me ........

	* /, main/features.c: Merged revisions 286527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
	  | 2 lines Refactor conversion to ast_poll() to fix callparking
	  regression. ........

2010-09-13 19:40 +0000 [r286457]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
	  5 lines Remove "Internal IP" from sip show settings, as it's not
	  at all useful to display. (closes issue #17840) Reported by: oej
	  ........

2010-09-13 15:52 +0000 [r286426]  Richard Mudgett <rmudgett@digium.com>

	* configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
	  reflect new libpri T309 default value.

2010-09-11 17:09 +0000 [r286270]  Olle Johansson <oej@edvina.net>

	* /, main/file.c: Merged revisions 286268 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
	  11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
	  lines Handle error response when we can't make file compatible
	  Review: https://reviewboard.asterisk.org/r/911/ ........
	  ................

2010-09-10 22:04 +0000 [r286189]  Terry Wilson <twilson@digium.com>

	* include/asterisk/channel.h, include/asterisk/pbx.h,
	  include/asterisk/frame.h, channels/chan_local.c,
	  funcs/func_channel.c: Merged revisions 286115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
	  (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
	  | 16 lines Inherit CHANNEL() writes to both sides of a Local
	  channel Having Local (/n) channels as queue members and setting
	  the language in the extension with Set(CHANNEL(language)=fr) sets
	  the language on the Local/...,2 channel. Hold time report
	  playbacks happen on the Local/...,1 channel and therefor do not
	  play in the specified language. This patch modifies
	  func_channel_write to call the setoption callback and pass the
	  CHANNEL() write info to the callback. chan_local uses this
	  information to look up the other side of the channel and apply
	  the same changes to it. (closes issue #17673) Reported by:
	  Guggemand Review: https://reviewboard.asterisk.org/r/903/
	  ........ ................

2010-09-10 21:11 +0000 [r286120]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
	  (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
	  2010) | 4 lines Load iax.conf before registering any
	  functions/applications/actions. Review:
	  https://reviewboard.asterisk.org/r/914/ ........ ................

2010-09-10 20:55 +0000 [r286118]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
	  (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
	  | 11 lines An outgoing call may not get hung up if a pre-connect
	  incoming ISDN call is disconnected. If the ISDN link a
	  pre-connect incoming call is using fails or is reset, the
	  outgoing leg may not hang up or be delayed in hanging up.
	  (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
	  PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
	  PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
	  incoming call leg hangs up before connecting for any reason. It
	  makes no sense to send a BUSY or CONGESTION control frame to the
	  outgoing call leg under these circumstances. ........
	  ................

2010-09-10 20:31 +0000 [r286112]  Russell Bryant <russell@digium.com>

	* main/db.c: Rate limit calls to fsync() to 1 per second after
	  astdb updates. Astdb was determined to be one of the most
	  significant bottlenecks in SIP registration processing. This
	  patch improved the speed of an astdb load test by 50000% (yes,
	  Fifty-Thousand Percent). On this particular load test setup, this
	  doubled the number of SIP registrations the server could handle.
	  Review: https://reviewboard.asterisk.org/r/825/

2010-09-10 18:31 +0000 [r286025]  Tilghman Lesher <tlesher@digium.com>

	* /: Merged revisions 286024 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
	  (Fri, 10 Sep 2010) | 9 lines Merged revisions 286023 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10
	  Sep 2010) | 2 lines Missing newline ........ ................

2010-09-10 13:13 +0000 [r285992]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt, CHANGES: Added missing documentation for
	  ExternalIVR feature added in January 2010

2010-09-10 05:32 +0000 [r285931-285962]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/select.h, /: Merged revisions 285961 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
	  | 6 lines Another fix for Mac OS X. While trying to fix this the
	  "right" way, I wandered into dependency hell. Two hours later, I
	  backed out, and just removed the offending code. ast_inline_api
	  only goes one level deep and then it breaks. Ouch. ........

	* tests/test_poll.c, include/asterisk/select.h, /, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
	  285930 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500
	  (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
	  | 7 lines Fix Mac OS X build. This also fixes a rather grievous
	  calculation error for the offset of ast_fdset, which was masked
	  on Linux and FreeBSD, because these platforms check the first 256
	  FDs regardless of the bitmask setting (due to backwards
	  compatibility). ........ ................

2010-09-09 22:52 +0000 [r285819]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, codecs/gsm/Makefile: Merged revisions 285818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400
	  (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
	  2010) | 8 lines GCC 4.2.x optimizations result in improper
	  behavior of GSM codec (closes issue #17688) Reported by:
	  pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
	  pprindeville (license 347) Tested by: mkeuter, pprindeville
	  ........ ................

2010-09-09 20:11 +0000 [r285745]  Jason Parker <jparker@digium.com>

	* main/channel.c, /: Merged revisions 285744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285744 | qwell | 2010-09-09 15:09:23 -0500
	  (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
	  9 lines Transmit silence when reading DTMF in ast_readstring.
	  Otherwise, you could get issues with DTMF timeouts causing
	  hangups. (closes issue #17370) Reported by: makoto Patches:
	  channel-readstring-silence-generator.patch uploaded by makoto
	  (license 38) ........ ................

2010-09-09 18:51 +0000 [r285640-285711]  Brett Bryant <bbryant@digium.com>

	* main/pbx.c, /: Merged revisions 285710 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010)
	  | 8 lines Fixes an issue with dialplan pattern matching where the
	  specificity for pattern ranges and pattern special characters was
	  inconsistent. (closes issue #16903) Reported by: Nick_Lewis
	  Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
	  657) Tested by: Nick_Lewis ........

	* res/res_musiconhold.c, /: Merged revisions 285639 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400
	  (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010)
	  | 7 lines Fixes an issue with MOH where it doesn't recover
	  cleanly when it can't play a file and would just stop, instead of
	  continuing to find the next playable file in the MOH class.
	  (closes issue #17807) Reported by: kshumard Review:
	  https://reviewboard.asterisk.org/r/910/ ........ ................

2010-09-08 22:14 +0000 [r285564-285568]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 285567 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500
	  (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08
	  Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the
	  end of the function on a transmit failure. ........
	  ................

	* /, channels/chan_sip.c: Merged revisions 285563 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010)
	  | 54 lines Fixes interoperability problems with session timer
	  behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require"
	  header. This is not to our benefit and RFC 4028 section 7.1 even
	  warns against it. It is possible for one endpoint to perform
	  session-timer refreshes while the other endpoint does not support
	  them. If in this case the end point performing the refreshing
	  puts "timer" in the Require field during a refresh, the dialog
	  will likely get terminated by the other end. 2. Change the
	  behavior of 'session-timer=accept' in sip.conf (which is the
	  default behavior of Asterisk with no session timer configuration
	  specified) to only run session-timers as result of an incoming
	  INVITE request if the INVITE contains an "Session-Expires"
	  header... Asterisk is currently treating having the "timer"
	  option in the "Supported" header as a request for session timers
	  by the UAC. I do not agree with this. Session timers should only
	  be negotiated in "accept" mode when the incoming INVITE supplies
	  a "Session-Expires" header, otherwise RFC 4028 says we should
	  treat a request containing no "Session-Expires" header as a
	  session with no expiration. Below I have outlined some situations
	  and what Asterisk's behavior is. The table reflects the behavior
	  changes implemented by this patch. SITUATIONS: -Asterisk as UAS
	  1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
	  "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
	  "Session-Expires". 200 Ok Response HAS "Session-Expires" header
	  4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
	  "Session-Expires" header 5. Outgoing INVITE: HAS
	  "Session-Expires". Active - Asterisk will have an active refresh
	  timer regardless if the other endpoint does. Inactive - Asterisk
	  does not have an active refresh timer regardless if the other
	  endpoint does. XXXXXXX - Not possible for mode.
	  ______________________________________ |SITUATIONS |
	  'session-timer' MODES | |___________|________________________| |
	  | originate | accept | |-----------|------------|-----------| |1.
	  | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
	  Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
	  -------------------------------------- (closes issue #17005)
	  Reported by: alexrecarey ........

2010-09-08 20:58 +0000 [r285533]  Brett Bryant <bbryant@digium.com>

	* /, apps/app_meetme.c: Merged revisions 285532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010)
	  | 8 lines Fixes a bug with MeetMe where after announcing the
	  amount of time left in a conference, if music on hold was
	  playing, it doesn't restart. (closes issue #17408) Reported by:
	  sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
	  sysreq (license 1009) Tested by: sysreq ........

2010-09-08 20:43 +0000 [r285527-285530]  Jason Parker <jparker@digium.com>

	* res/res_musiconhold.c, /, include/asterisk/astobj2.h: Merged
	  revisions 285529 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) |
	  1 line Follow coding guidelines in moh rescan fix. Also fix the
	  documentation that got me in trouble. ........

	* res/res_musiconhold.c, /: Merged revisions 285526 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep
	  2010) | 8 lines Fixes issue where moh files were no longer
	  rescanned during a reload. (closes issue #16744) Reported by: pj
	  Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
	  by: qwell ........

2010-09-08 07:14 +0000 [r285484]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_channel.c: Documentation only

2010-09-07 22:22 +0000 [r285455]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Don't automatically add domains for wildcard
	  bindaddrs. (closes issue #17832) Reported by: oej Patches:
	  17832-wildcard.diff uploaded by qwell (license 4) Tested by:
	  qwell

2010-09-07 21:20 +0000 [r285373-285386]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_spool.c: Don't notify on attribute changes, and change
	  how the queuing mechanism works. Fixes call spools in 1.8.
	  (closes issue #17337) Reported by: loloski Patches:
	  20100827__issue17337.diff.txt uploaded by tilghman (license 14)
	  (closes issue #17924) Reported by: mkeuter Tested by: mkeuter

	* funcs/func_channel.c: Add CHANNEL(checkhangup) to check whether a
	  channel is in the process of being hanged up. (closes issue
	  #17652) Reported by: kobaz Patches: func_channel.patch uploaded
	  by kobaz (license 834)

2010-09-07 21:08 +0000 [r285371]  Richard Mudgett <rmudgett@digium.com>

	* main/features.c: Fix cut-n-paste error.

2010-09-07 20:58 +0000 [r285369]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Add note to 'sip show settings' regarding
	  dual-stack support, and a :: bindaddress. (closes issue #17831)
	  Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by
	  qwell (license 4)

2010-09-07 20:56 +0000 [r285268-285367]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_config.c, /: Merged revisions 285366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500
	  (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
	  | 9 lines Catch invalid extensions at the parser, instead of
	  making the core deal with them. (closes issue #17794) Reported
	  by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
	  by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
	  uploaded by tilghman (license 14) Tested by: PavelL ........
	  ................

	* include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: Fix
	  build on FreeBSD 8.0, take 2.

	* main/poll.c, /: Merged revisions 285267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500
	  (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
	  | 4 lines Use poll, if indicated to do so, in the ast_poll2
	  implementation. This fixes the unit tests on FreeBSD 8.0.
	  ........ ................

2010-09-07 17:54 +0000 [r285197]  Brett Bryant <bbryant@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 285196 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400
	  (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010)
	  | 10 lines Fixes voicemail.conf issues where mailboxes with
	  passwords that don't precede a comma would throw unnecessary
	  error messages. (closes issue #15726) Reported by: 298 Patches:
	  M15726.diff uploaded by junky (license 177) Tested by: junky
	  Review: [full review board URL with trailing slash] ........
	  ................

2010-09-07 17:47 +0000 [r285195]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Merged revisions 285193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  ........ Merged revisions 285192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........
	  r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010)
	  | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does
	  not update the caller id of the channel if a new connected number
	  or ECT-INFORM (w/ new peer number on call transfer) is received.
	  JIRA ABE-2502 JIRA SWP-2058 ........ ........

2010-09-06 20:10 +0000 [r285161-285162]  Russell Bryant <russell@digium.com>

	* configure: regenerate configure script.

	* include/asterisk/autoconfig.h.in, configure.ac: Fix libsrtp -fPIC
	  check for when non-standard prefix is used. Thanks to loompek in
	  #asterisk for reporting the issue and testing this patch.

2010-09-06 06:56 +0000 [r285090]  Tilghman Lesher <tlesher@digium.com>

	* BSDmakefile (added), makeopts.in, /: Merged revisions 285089 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500
	  (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06
	  Sep 2010) | 2 lines Silly convenience script for BSD platforms.
	  ........ ................

2010-09-04 18:08 +0000 [r285057]  Russell Bryant <russell@digium.com>

	* include/asterisk/cli.h: Add a C++ compatible version of
	  AST_CLI_DEFINE().

2010-09-03 23:19 +0000 [r285017]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Call correct lock function as transferer is
	  a sip_pvt not a channel Both functions are #defined to ao2_lock,
	  but still...

2010-09-03 22:21 +0000 [r285006]  David Vossel <dvossel@digium.com>

	* configs/sip.conf.sample, channels/sip/include/sip.h,
	  channels/chan_sip.c: Disables auth_options_request option by
	  default. The auth_options_request option was created to do
	  authentication on OPTIONS request just like INVITES are done.
	  Since it has been noted that some endpoints use OPTIONS requests
	  as a way of qualifying a peer and that a 401 authentication
	  response could result in interoperability issues, this option has
	  been disabled by default.

2010-09-03 18:19 +0000 [r284967]  Brett Bryant <bbryant@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 284958 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03
	  Sep 2010) | 8 lines This is a patch provided for issue #17935 to
	  add the ActionID to the IAXregistry AMI response. (closes issue
	  #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by
	  alexkuklin (license 1115) Tested by: alexkuklin ........

2010-09-03 18:03 +0000 [r284950-284952]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: During OPTIONS authentication, the authpeer
	  does not need to be returned for any reason.

	* configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
	  channels/chan_sip.c: authenticate OPTIONS requests just like we
	  would an INVITE OPTIONS requests should be treated the same as an
	  INVITE This includes authentication. This patch adds the ability
	  for incoming out of dialog OPTION requests to be authenticated
	  before providing a response indicating whether an extension is
	  available or not. The authentication routine works the exact same
	  way as it does for incoming INVITEs. This means that if a peer
	  has 'insecure=invite' in their peer definition, the same will be
	  true for the processing of the OPTIONS request. Review:
	  https://reviewboard.asterisk.org/r/881/

2010-09-03 16:28 +0000 [r284921]  Terry Wilson <twilson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 284897 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284897 | twilson | 2010-09-03 11:20:45 -0500
	  (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
	  | 5 lines Properly detect when a sound file doesn't exist
	  ast_fileexists returns -1 for error and 0 for a non-existant
	  file. The existing code treated missing files as though they
	  existed. ........ ................

2010-09-03 13:07 +0000 [r284849-284852]  Jan Kalab <pitlicek@gmail.com>

	* res/res_calendar_ews.c: Calendar categories and priorities:
	  strdupa() fix

	* res/res_calendar_ews.c: Fix for calendar categories and
	  priorities according to ISO C90

	* res/res_calendar_caldav.c, include/asterisk/calendar.h,
	  res/res_calendar_ews.c, res/res_calendar.c,
	  res/res_calendar_icalendar.c: Support for calendar events
	  priorities and categories Review 880

2010-09-02 21:04 +0000 [r284781]  Brett Bryant <bbryant@digium.com>

	* main/manager.c, /: Merged revisions 284778 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284778 | bbryant | 2010-09-02 16:54:33 -0400
	  (Thu, 02 Sep 2010) | 14 lines Merged revisions 284777 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
	  | 7 lines Fixes a bug in manager.c where the default
	  configuration values weren't reset when the manager configuration
	  was reloaded. (closes issue #17917) Reported by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/883/ ........ ................

2010-09-02 21:02 +0000 [r284779-284780]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Simplified pri_dchannel() poll timeout
	  duration code.

	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
	  Made output libpri event names if pri debugging is enabled when
	  sig_pri processes them. * Simplified CLI "pri debug xx span xx"
	  command code and removed redundant debugging enabled messages. *
	  Made CLI "pri debug xx span xx" command only close the debugging
	  log file if it was opened.

2010-09-02 16:56 +0000 [r284705]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 284704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500
	  (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
	  | 7 lines Removed relatedpeer code from sip_autodestruct Handling
	  of the relatedpeer structure associated with a sip_pvt should be
	  done during the final sip_destruction function, not in
	  sip_autodestruct. ........ ................

2010-09-02 16:43 +0000 [r284701]  Jason Parker <jparker@digium.com>

	* formats/format_wav.c: Add slin16 support for format_wav (new
	  wav16 file extension) (closes issue #15029) Reported by: andrew
	  Patches: wav16.patch uploaded by andrew (license 240) Tested by:
	  qwell, andrew

2010-09-02 16:34 +0000 [r284698]  Richard Mudgett <rmudgett@digium.com>

	* doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added),
	  doc/tex/asterisk.tex: Added documentation for CONNECTEDLINE and
	  REDIRECTING functions. (closes issue #17808) Reported by: jtodd
	  Review: https://reviewboard.asterisk.org/r/875/

2010-09-02 16:27 +0000 [r284597-284696]  Tilghman Lesher <tlesher@digium.com>

	* addons/ooh323c/src/oochannels.c: Fixing build

	* channels/chan_usbradio.c, /: Merged revisions 284665 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02
	  Sep 2010) | 2 lines Fixing build. ........

	* apps/app_queue.c, /: Merged revisions 284631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010)
	  | 7 lines Don't reset queue stats on a module reload. (closes
	  issue #17535) Reported by: raarts Patches:
	  20100819__issue17535.diff.txt uploaded by tilghman (license 14)
	  ........

	* channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c,
	  apps/app_followme.c, main/loader.c, apps/app_speech_utils.c,
	  pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c,
	  include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
	  apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c,
	  apps/app_adsiprog.c, channels/chan_sip.c, channels/chan_agent.c:
	  When optional_api is non-optional, force dependent modules to be
	  loaded. (closes issue #17707) Reported by: ira Patches:
	  20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman Review:
	  https://reviewboard.asterisk.org/r/876/

	* include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c,
	  main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h
	  (added), channels/chan_phone.c, channels/chan_misdn.c, configure,
	  main/features.c, include/asterisk/poll-compat.h,
	  tests/test_poll.c (added), addons/ooh323c/src/oochannels.c,
	  main/asterisk.c, addons/ooh323c/src/ooSocket.h, main/stun.c,
	  res/res_ais.c, /, include/asterisk/autoconfig.h.in, configure.ac,
	  channels/console_video.c: Merged revisions 284593,284595 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500
	  (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010)
	  | 11 lines Ensure that all areas that previously used select(2)
	  now use poll(2), with implementations that need poll(2)
	  implemented with select(2) safe against 1024-bit overflows. This
	  is a followup to the fix for the pthread timer in 1.6.2 and
	  beyond, fixing a potential crash bug in all supported releases.
	  (closes issue #17678) Reported by: russell Branch:
	  https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
	  Review: https://reviewboard.asterisk.org/r/824/ ........
	  ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500
	  (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after
	  last commit ................

2010-09-01 21:47 +0000 [r284561]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: During request to dialog matching, verify
	  init_ruri is present before comparing. During request to dialog
	  matching, we attempt a best effort routine for fork detection
	  which requires several elements to be in place. The dialog's
	  initial request uri is one of those elements. Since it is best
	  effort, if the init_ruri is not present for some reason we can
	  not proceed with that routine.

2010-09-01  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-beta5 released.

2010-09-01 18:44 +0000 [r284477]  Terry Wilson <twilson@digium.com>

	* res/res_srtp.c, res/res_rtp_asterisk.c,
	  include/asterisk/res_srtp.h, main/rtp_engine.c,
	  channels/chan_sip.c: Fix SRTP for changing SSRC and multiple
	  a=crypto SDP lines Adding code to Asterisk that changed the SSRC
	  during bridges and masquerades broke SRTP functionality. Also
	  broken was handling the situation where an incoming INVITE had
	  more than one crypto offer. This patch caches the SRTP policies
	  the we use so that we can change the ssrc and inform libsrtp of
	  the new streams. It also uses the first acceptable a=crypto line
	  from the incoming INVITE. (closes issue #17563) Reported by:
	  Alexcr Patches: srtp.diff uploaded by twilson (license 396)
	  Tested by: twilson Review:
	  https://reviewboard.asterisk.org/r/878/

2010-09-01 18:16 +0000 [r284415-284473]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, /: Merged revisions 284472 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01
	  Sep 2010) | 5 lines Don't warn on floats and timestamps (closes
	  issue #17082) Reported by: coolmig ........

	* /, channels/chan_sip.c: Merged revisions 284399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
	  (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
	  | 7 lines Don't send a devstate change on poke_noanswer if the
	  state did not change. (closes issue #17741) Reported by: schmidts
	  Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
	  ........ ................

2010-08-31 19:00 +0000 [r284318]  Leif Madsen <lmadsen@digium.com>

	* configs/say.conf.sample, /: Merged revisions 284317 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
	  (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010)
	  | 7 lines Update say.conf.sample to match the rules in say.c
	  (closes issue #17835) Reported by: RoadKill Patches:
	  say.conf.sample.patch.rules uploaded by RoadKill (license 933)
	  Tested by: RoadKill ........ ................

2010-08-30 22:28 +0000 [r284281]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_festival.c: Merged revisions 284280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
	  | 11 lines Fix 3 coding errors: 1) After we close FD, we should
	  not be trying to write to it. 2) Call _exit(0), not exit(0), to
	  avoid running shutdown routines in a child. 3) Use endian, not
	  processor, detection to ensure bytes are written in the correct
	  order. (closes issue #15706) Reported by: modelnine Patches:
	  asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine
	  (license 865) Tested by: gmartinez ........

2010-08-29 07:05 +0000 [r284096-284158]  Tilghman Lesher <tlesher@digium.com>

	* configs/res_curl.conf.sample (added): Missed adding this file

	* sounds: Also ignore the checksums

	* configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c
	  (removed), cel/cel_odbc.c (added),
	  configs/cel_adaptive_odbc.conf.sample (removed): Rename CEL
	  adaptive driver to plain driver, since there isn't another ODBC
	  driver (and the other CEL drivers have adaptive capabilities,
	  anyway).

2010-08-28 21:29 +0000 [r284065]  Russell Bryant <russell@digium.com>

	* main/manager.c: Be more flexible with whitespace on AMI action
	  headers. Previously, this code required exactly one space to be
	  after the ':' in headers for an AMI action. This now makes
	  whitespace optional, and allows whitespace that is there to vary
	  in amount. (closes issue #17862) Reported by: cmoye Patches:
	  manager.c.patch_trunk uploaded by cmoye (license 858)
	  manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
	  cmoye

2010-08-27 22:37 +0000 [r284032]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 284002 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
	  (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
	  | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
	  (closes issue #17758) Reported by: ibc Patches:
	  multiple_accept_headers_1.4.diff uploaded by dvossel (license
	  671) ........ ................

2010-08-27 21:33 +0000 [r283951]  Russell Bryant <russell@digium.com>

	* pbx/pbx_realtime.c: Print exten@context:priority in verbose
	  messages from pbx_realtime.

2010-08-27 20:31 +0000 [r283882]  Jason Parker <jparker@digium.com>

	* main/config.c, addons/res_config_mysql.c, res/res_config_odbc.c,
	  /: Merged revisions 283881 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r283881 | qwell | 2010-08-27 15:30:27 -0500
	  (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
	  8 lines Fix issue with decoding ^-escaped characters in realtime.
	  (closes issue #17790) Reported by: denzs Patches:
	  17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
	  denzs ........ ................

2010-08-26 23:47 +0000 [r283770]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Convert MOH to use generic timers. (closes
	  issue #17726) Reported by: lmadsen Patches:
	  20100825__issue17726__2.diff.txt uploaded by tilghman (license
	  14) Tested by: tilghman

2010-08-26 15:26 +0000 [r283692]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 283691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
	  (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
	  | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
	  before invite receives a response. If an ast_channel with a SIP
	  tech pvt hangs up before the sip dialog gets a response to its
	  outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
	  not rfc compliant and results in confusion at the other endpoint.
	  sip_pretend_ack will ack and remove all the packets in the
	  retransmit queue. This means that the INVITE will stop
	  retransmitting, and that any response to that INVITE that comes
	  after the pretend_ack occurs will be ignored. Instead of faking
	  any sort of acknowledgement for an outgoing INVITE during an
	  internal hangup, we should let the protocol stack process the
	  INVITE transaction and terminate the dialog properly. This is
	  achieved by setting the PENDING_BYE flag. When this flag is used,
	  once the dialog proceeds to an escapable state the transaction
	  will either be canceled with a SIP_CANCEL or completed followed
	  immediately by a BYE. Attempting to do this any other way is
	  incorrect. If the endpoint is not responding to the INVITE
	  request, the INVITE must continue to be retransmitted until it
	  times out which will result in the dialog being destroyed.
	  ........ ................

2010-08-26 13:26 +0000 [r283627-283659]  Russell Bryant <russell@digium.com>

	* res/res_odbc.c: Slight improvement to a debug message.

	* keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed),
	  Makefile: Remove public keys that are no longer useful.

	* configs/manager.conf.sample: Move httptimeout out from in between
	  port and bindaddr.

2010-08-25 22:57 +0000 [r283595]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 283594 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
	  | 7 lines Add to and from tags to NOTIFY dialog-info xml body so
	  pickup can occur. When pedantic mode is used, the dialog-info xml
	  generated during a ringing event must contain the to and from tag
	  values. Otherwise if a pickup occurs using INVITE with replaces,
	  Astrisk will not be able to locate the subscription. ........

2010-08-25 16:12 +0000 [r283561]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Initialize connect timeout on each time through
	  the loop. (closes issue #17911) Reported by: wurstsalat

2010-08-25 15:54 +0000 [r283559]  David Vossel <dvossel@digium.com>

	* channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
	  revisions 283558 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
	  | 10 lines Asterisk will not advertise session timers are
	  supported when 'session-timers=refuse' is used. Asterisk now
	  dynamically builds the "Supported" header depending on what is
	  enabled/disabled in sip.conf. Session timers used to always be
	  advertised as being supported even when they were disabled in the
	  configuration. This caused problems with some end points. (issue
	  #17005) ........

2010-08-25 14:55 +0000 [r283527]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to
	  ast_debug(...)

2010-08-24 20:34 +0000 [r283493]  David Vossel <dvossel@digium.com>

	* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
	  Changes the default behavior for sip.conf's pedantic option from
	  "no" to "yes".

2010-08-24 18:56 +0000 [r283457]  Leif Madsen <lmadsen@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_sip.c: Fix issue where TOS
	  is no longer set on RTP packets. Fix issue where the tos is no
	  longer being set on RTP packets through res_rtp_asterisk. (closes
	  issue #17890) Reported by: elguero Patches: qos_18.diff uploaded
	  by elguero (license 37) Review:
	  https://reviewboard.asterisk.org/r/868

2010-08-24 16:11 +0000 [r283382]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 283381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
	  (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
	  | 11 lines This fix makes sure the ast_channel hangs up correctly
	  when the dialog's PENDING_BYE flag is set. When the pending bye
	  flag is used, it is possible that the dialog will terminate and
	  leave the sip_pvt->owner channel up. This is because we never
	  hangup the ast_channel after sending the SIP_BYE request. When we
	  receive the response for the SIP_BYE we set need_destroy which we
	  would expect to destroy the dialog on the next do_monitor loop,
	  but this is not the case. The dialog will only be destroyed once
	  the owner is hungup even with the need_destroy flag set. This
	  patch sets the softhangup flag on the ast_channel when a SIP_BYE
	  request is sent as a result of the pending bye flag. ........
	  ................

2010-08-24 12:49 +0000 [r283350]  Russell Bryant <russell@digium.com>

	* funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle.

2010-08-23 21:33 +0000 [r283319]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c,
	  /: Merged revisions 283318 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010)
	  | 2 lines CDR drivers depend upon res_odbc, not directly on the
	  ODBC libraries ........

2010-08-23  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-beta4 Released.

2010-08-23 13:35 +0000 [r283177-283241]  Russell Bryant <russell@digium.com>

	* configs/cel.conf.sample: Add sample configuration for cel_radius.

	* main/cel.c, include/asterisk/cel.h: Make the AST_CEL_AMA enum
	  match up with the AST_CDR_ ama flag values. Really, having 2
	  enums for this is silly and error prone, demonstrated by the
	  crash that I hit because there was an assumption in the code that
	  the values in each matched up. However, this is a quick fix to
	  get them to match up so it will work.

	* main/cel.c: Don't blow up on an invalid AMA flag.

	* configs/cel_custom.conf.sample: Tack on ${eventextra} to the
	  sample cel_custom.conf.

	* configs/cel_custom.conf.sample: Cut down on excessive quotation.

2010-08-23 12:06 +0000 [r283175]  Tilghman Lesher <tlesher@digium.com>

	* res/res_stun_monitor.c: Don't fail to start if the config file is
	  missing.

2010-08-23 11:58 +0000 [r283173]  Russell Bryant <russell@digium.com>

	* configs/cel_custom.conf.sample: Expand cel_custom.conf.sample.
	  Include the usage of CSV_QUOTE() to ensure data has valid CSV
	  formatting. Also list the special CEL variables that are
	  available for use in the mapping.

2010-08-20 16:51 +0000 [r283050-283125]  Richard Mudgett <rmudgett@digium.com>

	* /: Recorded merge of revisions 283124 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r283124 | rmudgett | 2010-08-20 11:48:10 -0500
	  (Fri, 20 Aug 2010) | 16 lines Merged revisions 283123 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
	  (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
	  https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
	  | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
	  line Reference correct struct member for unlikely event
	  PRI_EVENT_CONFIG_ERR. .......... ................
	  ................

	* channels/sig_pri.c, /: Merged revisions 283049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
	  (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010)
	  | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a
	  protocol error The PRI layer in chan_dadhi will check if a
	  PROGRESS message has already been sent, and not allow sending
	  another (although that is technically allowed by the Q931 spec),
	  however it does not protect against sending an ALERTING and then
	  sending a PROGRESS message, which is a violation of the
	  specification. Most switches don't seem to care too deeply about
	  this, but some do, and will disconnect the call when receiving
	  this invalid sequence. Protocol specification reference:
	  T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
	  protocol control (network side) point-point (sheet 3 of 8)"
	  (closes issue #17874) Reported by: nic_bellamy Patches:
	  asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
	  nic bellamy (license 299)
	  asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299)
	  asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299) ........ ................

2010-08-20 12:45 +0000 [r282979-283013]  Russell Bryant <russell@digium.com>

	* configs/cel_adaptive_odbc.conf.sample: Fix a typo in a column
	  name.

	* apps/app_celgenuserevent.c: Add an argument missing from the
	  CELGenUserEvent documentation.

2010-08-19 21:07 +0000 [r282891-282895]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 282894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
	  (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
	  | 11 lines tos_sip option was not being set correctly When
	  tos_sip is used, the tos of the sip socket is only set correctly
	  if the socket binding changes on a reload. If the binding stays
	  the same but the TOS changes, the new tos value would not take
	  into effect. This patch fixes that. (closes issue #17712)
	  Reported by: nickb ........ ................

	* /, channels/chan_sip.c: Merged revisions 282890 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
	  | 5 lines fixes sip peer memory leaks in the peer_by_ip table
	  (issue #17798) ........

2010-08-19 20:01 +0000 [r282860]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 282859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
	  (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
	  2010) | 16 lines Regression with T.38 negotiation Prior to
	  1.4.26.3 T.38 negotiation worked properly, in the case of the
	  reporter. (issue #16852) Reported by: cfc (closes issue #16705)
	  Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
	  by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
	  samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
	  ................

2010-08-19 14:44 +0000 [r282826]  Tilghman Lesher <tlesher@digium.com>

	* main/netsock2.c: Only output debugging if the debug level is on.

2010-08-19 02:18 +0000 [r282740]  Terry Wilson <twilson@digium.com>

	* configs/sip.conf.sample, /: Merged revisions 282730 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r282730 | twilson | 2010-08-18 21:14:28 -0500
	  (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
	  Aug 2010) | 2 lines Add some documentation about codec
	  negotiation to sip.conf ........ ................

2010-08-18 15:28 +0000 [r282671-282672]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h: Use the correct type for aoce_delayhangup bit
	  field.

	* channels/chan_dahdi.c: Use the correct operator when calculating
	  the PRI span devstate.

2010-08-18 13:10 +0000 [r282639]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Properly handle 200 and unknown responses
	  conatined in NOTIFY requests received in response to REFER
	  requests. This patch fixes the way asterisk handles NOTIFY
	  requests received in response to REFER requests. These changes to
	  NOTIFY handler were first introduced in r217482. This new change
	  properly handles the 200 response by queueing an
	  AST_TRANSFER_SUCCESS control frame and also prevents that control
	  frame from being queued when provisional and unknown responses
	  are received. (issue #17486) Reported by: davidw Tested by:
	  mnicholson (issue #12713) Reported by: davidw Review:
	  https://reviewboard.asterisk.org/r/860/

2010-08-18 12:30 +0000 [r282638]  Russell Bryant <russell@digium.com>

	* channels/chan_multicast_rtp.c: Split _all_ arguments before
	  parsing them. This fixes multicast RTP paging using linksys mode.

2010-08-18 07:49 +0000 [r282608]  Tilghman Lesher <tlesher@digium.com>

	* channels/sig_pri.c, /: Merged revisions 282607 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
	  | 9 lines Don't warn on callerid when completely text, instead of
	  numeric with localdialplan prefixes. (closes issue #16770)
	  Reported by: jamicque Patches: 20100413__issue16770.diff.txt
	  uploaded by tilghman (license 14) 20100811__issue16770.diff.txt
	  uploaded by tilghman (license 14) Tested by: jamicque ........

2010-08-17 21:36 +0000 [r282543-282577]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 282576 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
	  | 9 lines fixes no default transport for temp peer creation in
	  chan_sip (closes issue #17829) Reported by: falves11 Patches:
	  issue_17829.rev1.txt uploaded by russell (license 2)
	  issue_17829.diff uploaded by dvossel (license 671) Tested by:
	  falves11 ........

	* channels/chan_iax2.c: ACCEPT message should respond with the new
	  FORMAT2 ie (closes issue #17804) Reported by: tpanton

	* include/asterisk/unaligned.h: fixes truncated uint64_t value in
	  put_unaligned_uint64_t() function (issue #17804)

2010-08-16 18:01 +0000 [r282470]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/asterisk.tex, doc/tex/sounds.tex (added), /: Merged
	  revisions 282469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
	  | 7 lines Add information about creating sounds files using the
	  sounds tools publically available so that others can create their
	  own sounds prompts using the same tools we use to generate sounds
	  releases. This allows people creating their own prompts to sound
	  consistent with the prompts available from the open source
	  project. SWP-595 ........

2010-08-16 17:53 +0000 [r282468]  Terry Wilson <twilson@digium.com>

	* main/channel.c, /: Merged revisions 282467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r282467 | twilson | 2010-08-16 12:32:01 -0500
	  (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
	  | 16 lines Send a SRCCHANGE indication when we masquerade
	  Masquerading a channel means that the src of the audio is
	  potentially changing, so send a SRCCHANGE so that RTP-based media
	  streams can get a new SSRC generated to reflect the change.
	  Original patch by addix (along with lots of testing--thanks!).
	  (closes issue #17007) Reported by: addix Patches:
	  1001-reset-SSRC-original-channel.diff uploaded by addix (license
	  1006) srcchange.diff uploaded by twilson (license 396) Tested by:
	  addix, twilson Review: https://reviewboard.asterisk.org/r/862/
	  ........ ................

2010-08-14 04:53 +0000 [r282366]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c, include/asterisk/sched.h: Fix our FRACKing
	  issue with chan_iax2 a different way. Review:
	  https://reviewboard.asterisk.org/r/861/

2010-08-13 23:53 +0000 [r282334]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: PRI CCSS may use a stale dial string for
	  the recall dial string. If an outgoing call negotiates a
	  different B channel than initially requested, the saved original
	  dial string was not transferred to the new B channel. CCSS uses
	  that dial string to generate the recall dial string.

2010-08-13 22:23 +0000 [r282236-282302]  David Vossel <dvossel@digium.com>

	* UPGRADE.txt, configs/sip.conf.sample, CHANGES,
	  channels/chan_sip.c: remove current STUN support from chan_sip.c
	  This patch removes the current broken/useless stun support from
	  chan_sip. (closes issue #17622) Reported by: philipp2 Review:
	  https://reviewboard.asterisk.org/r/855/

	* CHANGES: res_stun_monitor and corresponding options CHANGES
	  documentation

	* configs/res_stun_monitor.conf.sample (added),
	  configs/sip.conf.sample, channels/chan_iax2.c,
	  configs/iax.conf.sample, channels/chan_sip.c,
	  include/asterisk/event_defs.h, res/res_stun_monitor.c (added):
	  res_stun_monitor for monitoring network changes behind a NAT
	  device Review: https://reviewboard.asterisk.org/r/854

	* /, channels/chan_sip.c: Merged revisions 282235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
	  | 16 lines only do magic pickup when notifycid is enabled A new
	  way of doing BLF pickup was introduced into 1.6.2. This feature
	  adds a call-id value into the XML of a SIP_NOTIFY message sent to
	  alert a subscriber that a device is ringing. This option should
	  only be enabled when the new 'notifycid' option is set... but
	  this was not the case. Instead the call-id value was included for
	  every RINGING Notify message, which caused a regression for
	  people who used other methods for call pickup. (closes issue
	  #17633) Reported by: urosh Patches: chan_sip.txt uploaded by
	  urosh (license ) blf_cid_issue.diff uploaded by dvossel (license
	  671) Tested by: dvossel, urosh, okrief, alecdavis ........

2010-08-13 16:02 +0000 [r282200-282201]  Terry Wilson <twilson@digium.com>

	* configure.ac: Whitespace fix :-/

	* configure, configure.ac: Detect when libsrtp cannot be linked in
	  a shared library The libsrtp build system currently does not
	  produce a shared library or a static library compiled with -fPIC,
	  so on 64-bit systems it is possible that we will get a compile
	  error if libsrtp is installed and res_srtp is selected in
	  menuselect. This patch attempts to detect this situation and
	  provide the user with instructions to work around the problem.

2010-08-12 22:51 +0000 [r282131]  Jason Parker <jparker@digium.com>

	* pbx/pbx_config.c, /: Merged revisions 282130 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r282130 | qwell | 2010-08-12 17:50:54 -0500
	  (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug
	  2010) | 1 line Register CLI commands before parsing config, in
	  case there is a config error. ........ ................

2010-08-12 22:06 +0000 [r282098]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/ccss.h, main/ccss.c: Separate call completion
	  config parameter allocation and default initialization. If you
	  ever have a need to reset the call completion config parameters
	  to defaults, now you can. And no Virginia, C++ idioms do not
	  always work in C.

2010-08-12 20:41 +0000 [r282066]  Russell Bryant <russell@digium.com>

	* CHANGES, main/cli.c: Add a "core reload" CLI command. Review:
	  https://reviewboard.asterisk.org/r/859/

2010-08-12 20:15 +0000 [r282047]  David Vossel <dvossel@digium.com>

	* CHANGES, include/asterisk/translate.h, main/cli.c,
	  main/translate.c: improved translation paths for wideband codecs
	  The problem I'm addressing is that Asterisk's current method of
	  building the least cost translation paths between codecs does not
	  take into account sample rate. For instance, it was possible for
	  siren14 (a 32khz codec), to contain the a translation path to
	  siren7 (a 16khz audio codec) that goes through slin at 8khz. In
	  this case Asterisk takes a 32khz codec, down samples it to 8khz
	  and then up samples it to 16khz which is terrible regardless if
	  it is computationally less expensive. This patch now builds
	  translation paths that give priority to maintaining the best
	  possible sample rate before taking into consideration
	  computational cost. This patch also adds cli commands to expose
	  what translation paths are actually being used. Changes: 1.
	  Translation paths will never contain a step that changes the
	  sample rate unless absolutely necessary. 2. When choosing the
	  best codec to make two channels compatible. Shared codecs with
	  the highest sample rate are given priority. 3. A new cli command
	  to show all translation paths available for a specific codec
	  'core show translation paths [codec name]' has been added. 4.
	  'core show translation' which displays the translation matrix now
	  includes the new higher bit audio codecs in the table. 5. 'core
	  show channel [channel name]' now displays the translation paths
	  if translation is used. (closes issue #16841) Reported by:
	  dvossel Review: https://reviewboard.asterisk.org/r/842/

2010-08-12 18:03 +0000 [r281982-282015]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Put back pointer value output for ast_debug(), such
	  that it is only removed for verbose output.

	* main/pbx.c: Remove debugging output from verbose messages.
	  Pointer values to internal objects is not terribly useful to
	  users in the verbose messages about adding extensions and
	  contexts.

2010-08-12 03:03 +0000 [r281913]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 281912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
	  (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
	  | 20 lines Ensure SSRC is changed when media source is changed to
	  resolve audio delay. This change causes the SSRC to change right
	  before the channels are bridged, which is what used to happen. It
	  seems that fixes were made to attempt limiting SSRC changes,
	  targeted mainly at sending DTMF. DTMF is not affecting the SSRC
	  with this change. There are two other control frames sent in
	  ast_channel_bridge that probably should also be changed to
	  AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
	  up to the discretion of resolving issue #17007. For reference -
	  old review implementing new control frame SRCCHANGE:
	  https://reviewboard.asterisk.org/r/540 (closes issue #17404)
	  Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
	  (license 325) Tested by: sdolloff ........ ................

2010-08-11 21:12 +0000 [r281875]  Leif Madsen <lmadsen@digium.com>

	* configs/say.conf.sample, /: Merged revisions 281873 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
	  (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010)
	  | 6 lines Add Danish support to say.conf.sample (closes issue
	  #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk
	  uploaded by RoadKill (license 933) ........ ................

2010-08-11 21:11 +0000 [r281874]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: handle all possible responses to REFER
	  requests (closes issue #17486) Reported by: davidw Patches:
	  Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
	  Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/

2010-08-11 20:30 +0000 [r281870]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_analog.c, channels/sig_analog.h: Fix a call to
	  analog_set_pulsedial() not setting 0 or 1 only. * Also a couple
	  minor tweaks.

2010-08-11 17:54 +0000 [r281764]  Leif Madsen <lmadsen@digium.com>

	* configs/say.conf.sample, /: Merged revisions 281763 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
	  (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010)
	  | 6 lines Allow say.conf to handle large numbers ending with
	  multiple zeros. (closes issue #17833) Reported by: RoadKill
	  Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
	  (license 933) ........ ................

2010-08-11 17:27 +0000 [r281760]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Avoid a deadlock in
	  add_header_max_forwards(). Related to r276951

2010-08-11 15:18 +0000 [r281723]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_readexten.c: Merged revisions 281722 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11
	  Aug 2010) | 7 lines Only set status TIMEOUT, if we have no
	  digits. (closes issue #15188) Reported by: jcovert Patches:
	  app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
	  551) ........

2010-08-11 13:30 +0000 [r281687]  <simon.perreault@viagenie.ca>

	* include/asterisk/netsock2.h, configs/sip.conf.sample,
	  channels/sip/config_parser.c, main/netsock2.c: Fix parsing of
	  IPv6 address literals in outboundproxy (closes issue #17757)
	  Reported by: oej Patches: 17757.diff uploaded by sperreault
	  (license 252) sip.conf.diff uploaded by sperreault (license 252)
	  Tested by: oej

2010-08-10 21:47 +0000 [r281568-281650]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
	  Change the default value for alwaysauthreject in sip.conf to
	  "yes". (closes issue #17756) Reported by: oej

	* main/sched.c, /: Merged revisions 281574 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
	  | 9 lines Don't move the time threshold for running scheduled
	  events on every iteration. Instead, only calculate the time
	  threshold each time ast_sched_runq() is called. (closes issue
	  #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
	  schmidts (license 1077) ........

	* apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281567 | russell | 2010-08-10 12:47:13 -0500
	  (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
	  | 8 lines Reset visible indication after answer. (closes issue
	  #17641) Reported by: klaus3000 Patches:
	  ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
	  klaus3000 (license 65) Tested by: schmidts ........
	  ................

2010-08-10  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-beta3 Released.

2010-08-10 17:48 +0000 [r281529-281568]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281567 | russell | 2010-08-10 12:47:13 -0500
	  (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
	  | 8 lines Reset visible indication after answer. (closes issue
	  #17641) Reported by: klaus3000 Patches:
	  ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
	  klaus3000 (license 65) Tested by: schmidts ........
	  ................

	* channels/chan_sip.c: Ensure that the proper external address is
	  used for the RTP destination. (closes issue #17044) Reported by:
	  ebroad Tested by: ebroad Review:
	  https://reviewboard.asterisk.org/r/566/

	* main/cli.c: Resolve a problem with channel name tab completion.
	  Hitting tab without typing any part of a channel name resulted in
	  no results. This now results in getting a full list of active
	  channels, just as it did in previous versions of Asterisk.
	  Review: https://reviewboard.asterisk.org/r/818/

2010-08-10 07:26 +0000 [r281497]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Fixed the issue caused by EXTEN including
	  user parameters.

2010-08-09 23:04 +0000 [r281466]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c: Add some more stuff to copy from 281429.

2010-08-09 20:47 +0000 [r281432]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
	  | 13 lines fixes SIP peers memory leak We zeroed out the peer's
	  addr before it was removed from the peers_by_ip container. This
	  made it impossible to be removed from the container as the addr
	  is the key used by the container to find the peer. (closes issue
	  #17774) Reported by: kkm Patches:
	  017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
	  017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
	  ........

2010-08-09 20:43 +0000 [r281429]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c, /: Merged revisions 281391 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
	  (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010)
	  | 13 lines Prevent loss of Caller ID information set on local
	  channel after masquerade. Caller ID set on the channel before a
	  masquerade occurs when using a local channel would cause the
	  information to be lost. The problem was that the information was
	  set on a channel destined to be hung up. The somewhat confusing
	  fix is to detect if any Caller ID has been set on the channel and
	  if so preswap the Caller ID data so that basically the masquerade
	  puts the data back. (closes issue #17138) Reported by: kobaz
	  Review: https://reviewboard.asterisk.org/r/847/ ........
	  ................

2010-08-09 14:49 +0000 [r281358]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Validate minrate, maxrate, and modem settings
	  before attempting a fax session. FAX-224

2010-08-09 14:31 +0000 [r281356]  <simon.perreault@viagenie.ca>

	* configs/sip.conf.sample: Added comment about IPv4-mapped IPv6
	  addresses and the output of netstat.

2010-08-09 12:51 +0000 [r281294-281325]  Russell Bryant <russell@digium.com>

	* configs/cdr.conf.sample: Add a couple of default values to the
	  documentation of cdr.conf.

	* configs/cdr.conf.sample: Reorder some options in cdr.conf.sample.
	  Put all of the options that affect the contents of CDRs together,
	  instead of having the batch mode options in the middle of them.

2010-08-06 18:57 +0000 [r281085]  Tilghman Lesher <tlesher@digium.com>

	* main/utils.c: Fix alignment of stringfields on the SPARC
	  architecture (closes issue #17789) Reported by: Ian Mason
	  Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman
	  (license 14) Tested by: Ian_Mason

2010-08-05 13:16 +0000 [r281052]  Russell Bryant <russell@digium.com>

	* main/cdr.c, /: Merged revisions 281051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
	  | 9 lines Cleanup default option value handling for cdr.conf
	  [general]. The default values would differ depending on whether
	  or not cdr.conf exists. That is no longer the case. Apply a
	  default value to the unanswered option. Define all default values
	  as named constants. ........

2010-08-05 07:46 +0000 [r280984]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
	  (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
	  | 8 lines Change context lock back to a mutex, because
	  functionality depends upon the lock being recursive. (closes
	  issue #17643) Reported by: zerohalo Patches:
	  20100726__issue17643.diff.txt uploaded by tilghman (license 14)
	  Tested by: zerohalo ........ ................

2010-08-04 15:11 +0000 [r280909]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Initialize FAXOPT() status variables in sendfax
	  and receivefax instead of when the details structure is created.

2010-08-04 14:04 +0000 [r280809-280879]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c: Check cur value before attempting a deref.
	  (closes issue #17775) Reported by: svinson Patches:
	  20100804__issue17775.diff.txt uploaded by tilghman (license 14)
	  Tested by: svinson (closes issue #17743) Reported by: tgruenberg
	  Patches: 20100804__issue17775.diff.txt uploaded by tilghman
	  (license 14) Tested by: tgruenberg

	* CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns
	  a 1-based index into a list of a specified item. Matches up with
	  FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth
	  Patches: svn-279754.diff uploaded by gareth (license 208) Tested
	  by: gareth, tilghman Review:
	  https://reviewboard.asterisk.org/r/810/

2010-08-03 19:54 +0000 [r280777-280778]  <simon.perreault@viagenie.ca>

	* channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
	  issue #17663) Reported by: oej Patches: diff uploaded by
	  sperreault (license 252) diff2 uploaded by sperreault (license
	  252) get_domain.diff uploaded by sperreault (license 252)

	* configs/sip.conf.sample: Better documentation related to IPv6.
	  (closes issue #17737) Reported by: oej Patches: doc.diff uploaded
	  by sperreault (license 252) Tested by: mmichelson

2010-08-03 18:48 +0000 [r280742]  Russell Bryant <russell@digium.com>

	* addons/Makefile, addons/mp3 (removed),
	  contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder
	  source code and replace it with a small shell script. Review:
	  https://reviewboard.asterisk.org/r/836/

2010-08-03 18:42 +0000 [r280624-280740]  Tilghman Lesher <tlesher@digium.com>

	* doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added):
	  Merged revisions 280739 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010)
	  | 2 lines Document -B and -W flags and regenerate manpage from
	  sgml ........

	* apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
	  Aug 2010) | 2 lines Allow the pipe, but also allow the comma
	  ........

	* main/Makefile: Make this a little more deterministic... we want
	  the latest value, not just a 1 somewhere.

	* main/Makefile: Apparently, the values in makeopts are sometimes
	  1:1 and sometimes 1. Compensate for this.

2010-07-29 21:07 +0000 [r280557]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Fix regression introduced in r1664. Give the fax
	  stack time to shutdown and populate the FAXOPT output variables.
	  FAX-222

2010-07-29 20:43 +0000 [r280552]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
	  | 11 lines fixes wrong SRV query for TLS connection (closes issue
	  #17612) Reported by: marcelloceschia Patches:
	  chan-sip_srvQuery.patch uploaded by marcelloceschia (license
	  1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
	  chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
	  (license 1079) Tested by: marcelloceschia, st, pabelanger
	  ........

2010-07-29 20:35 +0000 [r280549]  Russell Bryant <russell@digium.com>

	* configs/ccss.conf.sample: Add header to ccss.conf to appease oej.
	  (closes issue #17755) Reported by: oej

2010-07-29 19:47 +0000 [r280519]  Sean Bright <sean@malleable.com>

	* channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa
	  -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
	  strdupa_oops.diff uploaded by malcolmd (license 924)

2010-07-29 19:13 +0000 [r280450]  David Vossel <dvossel@digium.com>

	* main/channel.c, /: Merged revisions 280449 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
	  (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
	  | 12 lines fixes issue with translator frame not getting freed A
	  translator frame even if it local storage so the translation path
	  can be freed. This issue prevented g729 licenses from being freed
	  up. (closes issue #17630) Reported by: manvirr Patches:
	  encoder_fix.diff uploaded by dvossel (license 671) Tested by:
	  manvirr, dvossel ........ ................

2010-07-29 18:37 +0000 [r280414-280446]  Paul Belanger <paul.belanger@polybeacon.com>

	* tests/test_utils.c: Remove res_crypto dependency.

	* tests/test_utils.c: crypto_loaded_test depends on res_crypto,
	  else test will fail.

2010-07-29 16:25 +0000 [r280391]  Russell Bryant <russell@digium.com>

	* main/rtp_engine.c: Don't blow up if get_codec() was not provided
	  in the RTP glue.

2010-07-29 16:07 +0000 [r280346]  Jean Galarneau <jgalarneau@digium.com>

	* /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r280345 | jeang | 2010-07-29 11:01:35 -0500
	  (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
	  2 lines Fix a dsp structure leak occuring when a local channel is
	  put into a meetme conference, then masquaraded away. ABE-2422
	  ........ ................

2010-07-29 15:57 +0000 [r280307-280343]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format
	  string. related to r280302

	* main/channel.c, channels/chan_local.c, /: Merged revisions 280306
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
	  2010) | 2 lines Implement support for ast_channel_queryoption on
	  local channels. Currently only AST_OPTION_T38_STATE is supported.
	  ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........
	  Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
	  through generic bridges. This change appears to have been
	  unintentionally left out of rev 203699.

2010-07-29 00:45 +0000 [r280302]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_usbradio.c: Use PRId64 with format_t

2010-07-28 20:49 +0000 [r280269]  Jeff Peeler <jpeeler@digium.com>

	* channels/sip/reqresp_parser.c: Give test category missing leading
	  slash

2010-07-28 20:12 +0000 [r280235]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
	  Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
	  called_nai and calling_nai config options. ........

2010-07-28 20:03 +0000 [r280233]  Jason Parker <jparker@digium.com>

	* sounds/Makefile, /: Merged revisions 280231 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
	  6 lines Work around some silly behavior on BSD. A non-zero exit
	  from a subshell should make the build fail. (closes issue #17621)
	  ........

2010-07-28 19:34 +0000 [r280225]  Terry Wilson <twilson@digium.com>

	* res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned
	  on w/o filtering

2010-07-28 18:24 +0000 [r280195]  Jason Parker <jparker@digium.com>

	* sounds/Makefile, /: Merged revisions 280193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
	  9 lines Remove unnecessary subshells. Attempt to make
	  checksumming work. Also improves readability. (issue #17621)
	  Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
	  ........

2010-07-28 16:52 +0000 [r280161]  Sean Bright <sean@malleable.com>

	* apps/app_queue.c, /: Merged revisions 280160 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
	  2010) | 8 lines Plug a reference leak in app_queue when adding
	  members dynamically. (closes issue #17738) Reported by:
	  bobwienholt Patches: issue17738.patch uploaded by bobwienholt
	  (license 950) Tested by: bobwienholt, seanbright ........

2010-07-28 13:52 +0000 [r280090]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
	  (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
	  Jul 2010) | 1 line Update help text to be less confusing.
	  ........ ................

2010-07-28 13:01 +0000 [r280058]  Russell Bryant <russell@digium.com>

	* res/res_crypto.c: s/init keys/keys init/

2010-07-28 01:37 +0000 [r280023]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_usbradio.c: Resolve compiler warning about
	  formatting (closes issue #17732) Reported by: pabelanger

2010-07-27 22:30 +0000 [r280019-280020]  Sean Bright <sean@malleable.com>

	* main/editline/el.h, main/term.c, main/cli.c,
	  main/editline/parse.c, main/editline/tokenizer.c,
	  main/editline/config.sub, main/editline/parse.h,
	  main/editline/tokenizer.h, configure, main/editline/histedit.h,
	  main/editline/sig.c, main/editline/PLATFORMS,
	  main/editline/sig.h, main/editline/key.c, main/editline/editrc.5,
	  main/editline/np/fgetln.c, main/editline/key.h,
	  main/editline/TEST/test.c, main/Makefile,
	  main/editline/configure, main/editline/Makefile.in, configure.ac,
	  main/editline/configure.in, main/editline/readline/readline.h,
	  main/editline/README, main/editline/editline.3,
	  main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c,
	  main/asterisk.c, main/editline/install-sh, main/editline/term.c,
	  main/editline/config.guess, main/editline/read.c,
	  main/editline/term.h, main/editline/map.c,
	  main/editline/np/strlcpy.c, main/editline (added),
	  main/editline/config.h.in, main/editline/read.h,
	  main/editline/tty.c, main/editline/np/unvis.c,
	  main/editline/prompt.c, main/editline/map.h, main/editline/tty.h,
	  main/editline/chared.c, main/editline/prompt.h,
	  main/editline/np/strlcat.c, main/editline/chared.h,
	  main/editline/np, main/editline/TEST, main/editline/refresh.c,
	  main/editline/history.c, main/editline/readline,
	  include/asterisk/term.h, main/editline/refresh.h,
	  main/editline/search.c, main/editline/hist.c,
	  main/editline/search.h, main/editline/hist.h,
	  main/editline/np/vis.c, build_tools/menuselect-deps.in, main,
	  main/editline/readline.c, main/editline/np/vis.h,
	  main/editline/INSTALL, makeopts.in, main/editline/CHANGES,
	  main/editline/common.c, main/xmldoc.c, main/editline/makelist.in,
	  include/asterisk/autoconfig.h.in, main/editline/el.c: Revert
	  r280019 for now - This was poorly executed.

	* include/asterisk/term.h, makeopts.in, main/asterisk.c,
	  main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
	  main: Add ability to use system libedit and update bundled
	  libedit. The version of libedit that is bundled with asterisk is
	  old and has some bugs. This patch updates the bundled version of
	  libedit within asterisk, and also updates asterisk to use the
	  system libedit instead if one is available (and pkg-config is
	  available). This review integrates several patches from other
	  users specifically kkm and tzafrir. (closes issue #15929)
	  Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff
	  uploaded by kkm (license 888) (issue #16858) Reported by:
	  jw-asterisk (closes issue #17039) Reported by: tzafrir Patches:
	  0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir
	  (license 46) Review: https://reviewboard.asterisk.org/r/807/

2010-07-27 21:16 +0000 [r279953]  Russell Bryant <russell@digium.com>

	* res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr,
	  formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
	  main/db1-ast/btree, configure, main/editline, codecs/g722, main,
	  main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael,
	  channels, main/stdtime, main/editline/np, codecs, utils,
	  main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add
	  --enable-coverage option to configure script. This option enables
	  the proper compiler flags for tracking code coverage, which is
	  useful along side automated testing.

2010-07-27 20:57 +0000 [r279949]  David Vossel <dvossel@digium.com>

	* main/audiohook.c, main/channel.c, /,
	  include/asterisk/audiohook.h: Merged revisions 279946 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
	  (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
	  | 19 lines remove empty audiohook write list on channel If a
	  channel has an audiohook write list created on it, that list
	  stays on the channel until the channel is destroyed. There is no
	  reason to keep that list on the channel if it becomes empty. If
	  it is empty that just means we are doing needless translating for
	  every ast_read and ast_write. This patch removes the audiohook
	  list from the channel once it is detected to be empty on either a
	  read or write. If a audiohook is added back to the channel after
	  this list is destroyed, the list just gets recreated as if it
	  never existed to begin with. (closes issue #17630) Reported by:
	  manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
	  ................

2010-07-27 19:50 +0000 [r279916]  Russell Bryant <russell@digium.com>

	* channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF
	  detection on outgoing ISDN calls. This is a regression from the
	  sig_pri split from chan_dahdi. When a call is first initiated,
	  the inband DTMF detector is not enabled if it's an outgoing ISDN
	  call. However, it needs to be turned on once the media path
	  starts up. This handling was put back in the open_media()
	  callback of chan_dahdi. In sig_pri, open_media() calls were added
	  to a few places where it was needed, including handling of
	  PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING.
	  Thanks to rmudgett for helping me with the patch!

2010-07-27 18:54 +0000 [r279887]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The
	  code was written in a way that did a bad job of parsing the port
	  out of a URI. Specifically, it would do badly when dealing with
	  an IPv6 address. In this particular scenario, there was no value
	  from parsing the port out, so I just removed that logic. And
	  while I was messing around in the function, I changed some
	  variable names to be more descriptive. (closes issue #17661)
	  Reported by: oej Patches: 17661.diff uploaded by mmichelson
	  (license 60)

2010-07-27 16:40 +0000 [r279850]  Jason Parker <jparker@digium.com>

	* sounds/Makefile, /: Merged revisions 279849 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) |
	  1 line Simply sounds/Makefile some more. ........

2010-07-27 16:09 +0000 [r279817]  David Vossel <dvossel@digium.com>

	* main/netsock2.c, channels/chan_sip.c: fix sip transaction match
	  with authentication, fix confusing log message when using
	  getaddrinfo

2010-07-27 16:06 +0000 [r279815]  Russell Bryant <russell@digium.com>

	* channels/chan_dahdi.c: Support "channels" in addition to
	  "channel" in chan_dahdi.conf. Review:
	  https://reviewboard.asterisk.org/r/804

2010-07-27 15:15 +0000 [r279785]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
	  2010) | 14 lines Fix bad behavior of dynamic_exclude_static
	  option in sip.conf. We were attempting to create a contactdeny
	  rule based on the peer's IP address before the peer's IP address
	  had been set. By moving the processing further down in the
	  function, we can ensure stuff works as we expect for it to.
	  (closes issue #17717) Reported by: mmichelson Patches:
	  17717.patch uploaded by mmichelson (license 60) Tested by:
	  DennisD ........

2010-07-27 02:57 +0000 [r279726-279755]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_dahdi.c: If dringXcontext is null, fallback to
	  default context value. (closes issue #17693) Reported by:
	  iasgoscouk Patches: issue17693.patch uploaded by pabelanger
	  (license 224) Tested by: iasgoscouk Review:
	  https://reviewboard.asterisk.org/r/803/

	* main/http.c: Use ast_sockaddr_setnull() when http is not enabled.
	  Otherwise, ast_tcptls_server_start() will still start http.
	  (closes issue #17708) Reported by: pabelanger Patches: http.patch
	  uploaded by pabelanger (license 224)

2010-07-26  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-beta2 Released.

2010-07-26 23:29 +0000 [r279689]  Paul Belanger <paul.belanger@polybeacon.com>

	* UPGRADE.txt, CHANGES: Updated documentation for FAX logger level.

2010-07-26 23:03 +0000 [r279658]  Jason Parker <jparker@digium.com>

	* sounds/Makefile (added), /, sounds/Makefile.380 (removed),
	  configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
	  (removed), configure.ac: Merged revisions 279657 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul
	  2010) | 5 lines Really fix sounds Makefile (and make it
	  readableish). There was a rather large syntax error that should
	  have caused ALL versions of GNU make to fail. I don't know how it
	  worked. ........

2010-07-26 21:53 +0000 [r279636]  Russell Bryant <russell@digium.com>

	* main/channel.c: Ignore a control subclass of -1 in
	  ast_waitfordigit_full().

2010-07-26 21:20 +0000 [r279599-279619]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 279609 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
	  Jul 2010) | 2 lines Dunno why this worked on my machine, but it
	  works better this way. ........

	* res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26
	  Jul 2010) | 13 lines Apply all patches in:
	  https://issues.asterisk.org/view.php?id=13573 (closes issue
	  #13573) Reported by: navkumar Patches:
	  res_config_ldap-category.diff uploaded by navkumar (license 580)
	  res_config_ldap.patch uploaded by bencer (license 961)
	  res_config_ldap uploaded by bencer (license 961) Tested by:
	  suretec ........

	* /: Reverting property remove

2010-07-26 20:58 +0000 [r279598]  Gavin Henry <ghenry@suretecsystems.com>

	* /: Merged revisions 279597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/1.6.2
	  -----------------------------------------------------------------------
	  r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
	  13 lines Apply all patches in:
	  https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
	  0013573) Reported by: navkumar Patches:
	  res_config_ldap-category.diff uploaded by navkumar (license 580)
	  res_config_ldap.patch uploaded by bencer (license 961)
	  res_config_ldap uploaded by bencer (license 961) Tested by:
	  suretec
	  ------------------------------------------------------------------------

2010-07-26 19:59 +0000 [r279568]  David Vossel <dvossel@digium.com>

	* channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
	  channels/sip/reqresp_parser.c: transaction matching using top
	  most Via header This patch modifies the way chan_sip.c does
	  transaction to dialog matching. Asterisk now stores information
	  in the top most Via header of the initial incoming request and
	  compares that against other Requests that have the same call-id.
	  This results in Asterisk being able to detect a forked call in
	  which it has received multiple legs of the fork. I completely
	  stripped out the previous matching code and made the comparisons
	  a little more explicit and easier to understand. My comments in
	  the code should offer all the details involving this patch. This
	  patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
	  find multiple dialogs with the same call-id. Since the callback
	  function was returning (CMP_MATCH | CMP_STOP) only the first item
	  found was being returned. I fixed this by making a new callback
	  function for finding multiple dialogs that only returns
	  (CMP_MATCH) on a match allowing for multiple items to be
	  returned. Review: https://reviewboard.asterisk.org/r/776/

2010-07-26 19:51 +0000 [r279566]  Paul Belanger <paul.belanger@polybeacon.com>

	* UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add
	  documentation for FAX logger level. (closes issue #17715)
	  Reported by: vrban Patches: 17715.patch uploaded by pabelanger
	  (license 224) Tested by: vrban

2010-07-26 19:18 +0000 [r279562]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile (removed), /, sounds/Makefile.380 (added),
	  configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
	  (added), configure.ac: Merged revisions 279561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
	  r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010)
	  | 2 lines Use a special Makefile for noobs who still have GNU
	  Make 3.80. ........

2010-07-26 16:04 +0000 [r279504]  Mark Michelson <mmichelson@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  channels/sip/reqresp_parser.c: Allow for systems without locale
	  support to be usable. A recent change to SIP URI comparison code
	  added a locale-specific string comparison to the mix, and certain
	  systems do not support such functions. This fix allows for those
	  systems to still use Asterisk 1.8 (closes issue #17697) Reported
	  by: pprindeville Patches: asterisk-trunk-bugid17697.patch
	  uploaded by pprindeville (license 347) Tested by: mmichelson

2010-07-26 15:43 +0000 [r279502]  Sean Bright <sean@malleable.com>

	* autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon,
	  26 Jul 2010) | 5 lines Expand the correct value within
	  AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
	  ........

2010-07-26 03:27 +0000 [r279472]  Tilghman Lesher <tlesher@digium.com>

	* formats/format_sln16.c, formats/format_wav_gsm.c,
	  formats/format_siren7.c, formats/format_ilbc.c,
	  formats/format_vox.c, formats/format_pcm.c,
	  formats/format_h263.c, formats/format_g723.c,
	  formats/format_h264.c, formats/format_g726.c,
	  formats/format_jpeg.c, formats/format_siren14.c,
	  formats/format_gsm.c, formats/format_g719.c,
	  formats/format_g729.c, formats/format_sln.c,
	  formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need
	  to load before apps, because some apps call
	  ast_format_str_reduce() at load time.

2010-07-25 21:26 +0000 [r279442]  Paul Belanger <paul.belanger@polybeacon.com>

	* tests/test_func_file.c: Add trailing backslash to silence warning
	  message.

2010-07-25 18:21 +0000 [r279390-279410]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes
	  issue #17304) Reported by: jnemeth Patches:
	  20100507__issue17304.diff.txt uploaded by tilghman (license 14)
	  Tested by: jnemeth

	* main/logger.c: Don't assume qlog is open. (closes issue #17704)
	  Reported by: vrban Patches: issue17704.patch uploaded by
	  pabelanger (license 224) Tested by: vrban

2010-07-24 23:58 +0000 [r279348]  Bradley Latus <brad.latus@gmail.com>

	* doc/asterisk.8: Minor update to man page

2010-07-24 20:47 +0000 [r279273-279314]  Paul Belanger <paul.belanger@polybeacon.com>

	* Makefile: Remove duplicate -c flag when using $(INSTALL) (closes
	  issue #17695) Reported by: pabelanger Patches: Makefile.diff
	  uploaded by pabelanger (license 224)

	* include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then
	  return. (closes issue #17677) Reported by: outcast Patches:
	  issue0017677.patch uploaded by pabelanger (license 224) Tested
	  by: elguero

	* main/manager.c: Default sin_family to AF_INET for TCP / TLS
	  Bindaddress. Otherwise, 'manager show settings' will generate
	  errors if manager is not enabled.

2010-07-23 22:20 +0000 [r279227]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
	  ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
	  (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
	  | 7 lines SIP promiscuous redirect could fail to dial the
	  redirect. The ast_channel was created with one variable to
	  ast_request() but the call to ast_call() that initiates the
	  outgoing call was using a different variable. The two variables
	  are not equivalent if the call_forward string included a channel
	  technology specifier. e.g., SIP/200 ........ ................

2010-07-12  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.8.0-beta1 Released.

2010-07-23 18:56 +0000 [r279113]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)

2010-07-23 18:23 +0000 [r279056-279094]  Russell Bryant <russell@digium.com>

	* /: fix up properties on 1.8 branch

	* / (added): Create a branch for Asterisk 1.8.

	  ___      _            _     _      _   ___
	 / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
	| |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
	|  _  \__ \ ||  __/ |  | \__ \   <  | || (_) |
	|_| |_|___/\__\___|_|  |_|___/_|\_\ |_(_)___/

2010-07-23 17:05 +0000 [r278982-278985]  Tilghman Lesher <tlesher@digium.com>

	* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
	  revisions 278984 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
	  | 5 lines Establish a maximum version for openh323 (i.e. not
	  opal), because chan_h323 will fail to load, even if it links.
	  (issue #17679) Reported by: am ........

	* /, main/asterisk.c: Merged revisions 278981 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
	  | 8 lines Avoid race with consolethread on shutdown (on parallel
	  processors). (closes issue #17080) Reported by: sybasesql
	  Patches: 20100721__issue17080.diff.txt uploaded by tilghman
	  (license 14) Tested by: sybasesql ........

2010-07-23 16:33 +0000 [r278980]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c,
	  channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
	  This initially was created to work around the issue of using a
	  string comparison instead of a binary comparison for IP
	  addresses. It evolved a bit when test cases were created and it
	  was discovered that comparison of URI parameters was not working
	  exactly as it should. sip_uri_cmp() and its helpers have been
	  moved to reqresp_parser.c and a new test has been added. (closes
	  issue #17662) Reported by: oej Review:
	  https://reviewboard.asterisk.org/r/792

2010-07-23 16:19 +0000 [r278957]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, res/res_config_odbc.c,
	  configs/extconfig.conf.sample, CHANGES, main/config.c,
	  res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
	  failover branch

2010-07-23 16:07 +0000 [r278947]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* doc/asterisk.8: Some left-over hyphen-minus fixes in the man page

2010-07-23 15:57 +0000 [r278944-278945]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)

	* channels/chan_sip.c: Disable SIP support by default for Asterisk
	  1.8.

2010-07-23 15:52 +0000 [r278943]  Mark Michelson <mmichelson@digium.com>

	* addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
	  sure didn't!

2010-07-23 15:41 +0000 [r278942]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Rename sig_pri_pri to sig_pri_span. More descriptive of concept.

2010-07-23 15:16 +0000 [r278908]  Mark Michelson <mmichelson@digium.com>

	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
	  channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
	  streams. Review: https://reviewboard.asterisk.org/r/795

2010-07-23 13:37 +0000 [r278875]  Olle Johansson <oej@edvina.net>

	* res/res_config_ldap.c: Minor corrections to the LDAP realtime
	  driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
	  Mark for a quick review!

2010-07-23 13:26 +0000 [r278873]  Paul Belanger <paul.belanger@polybeacon.com>

	* Makefile, agi/Makefile, sounds/Makefile: Portability updates for
	  Makefiles. When possible, use $(INSTALL). This allows us to use
	  the functionality within install for setting directory / file
	  permissions, a requirement for unprivileged installation. Also
	  move any directory we plan to create within the installdirs
	  macro. Plus various other formatting issues. (issue #17436)
	  Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
	  pabelanger (license 224) Tested by: pabelanger Review:
	  https://reviewboard.asterisk.org/r/654/

2010-07-23 11:01 +0000 [r278809-278841]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl
	  start polarityswitch when finally on hook. (issue #17318)

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  channels/sig_analog.c, channels/sig_analog.h: Support FXS module
	  Polarity Reversal on remote party Answer and Hangup FXS lines
	  normally connect to a telephone. However, when FXS lines are
	  routed to an external PBX or Key System to act as "external" or
	  "CO" lines, it is extremely difficult, if not impossible for the
	  external PBX to know when the call has been disconnected without
	  receiving a polarity reversal on the line. Now using
	  answeronpolarityswitch and hanguponpolarityswitch keywords that
	  previously were used only for FXO ports, now applies like
	  functionality for an FXS port, but from the connected equipment's
	  point of view. (closes issue #17318) Reported by: armeniki
	  Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/797/

2010-07-22 21:16 +0000 [r278777]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: DNID not cleared when channel hang up
	  (Affects PRI and SS7) The "dahdi show channels" CLI command still
	  reports the DNID of the previous call even if the call is already
	  hang up. The "dahdi show channels" command of older releases
	  clear the DNID once the channel is hang up. Regression from the
	  sig_analog/sig_pri extraction from chan_dahdi. (closes issue
	  #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded
	  by rmudgett (license 664) Tested by: rmudgett

2010-07-22 19:45 +0000 [r278708]  Jeff Peeler <jpeeler@digium.com>

	* main/xmldoc.c: Add method for finding XML doc files for systems
	  that don't support GLOB_BRACE. In particular, Solaris and perhaps
	  others do not support the above mentioned GNU extension. In this
	  case the paths are simply expanded without the braces and the
	  calls to glob are made separately. Note: I could not explain
	  memory allocation failures that were being reported from within
	  libxml itself when making calls to glob without using
	  GLOB_NOCHECK. This is the only reason why that flag is being
	  used. (closes issue #15402) Reported by: snuffy Patches:
	  bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by
	  me

2010-07-22 14:58 +0000 [r278620]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 278618 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
	  2010) | 13 lines Allow PLC to function properly when channels use
	  SLIN for audio. If a channel involved in a bridge was using SLIN
	  audio, then translation paths were not guaranteed to be set up
	  properly since in all likelihood the number of translation steps
	  was only 1. This patch enforces the transcode_via_slin behavior
	  if transcode_via_slin or generic_plc is enabled and one of the
	  formats to make compatible is SLIN. AST-352 ........

2010-07-22 14:56 +0000 [r278619]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: update sip subscription debug message to a
	  warning message If the Expire header of a SUBSCRIBE is less that
	  our expiremin, a log warning will be displayed.

2010-07-22 05:29 +0000 [r278579]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/doxyref.h: Add the full current set of CDR
	  drivers

2010-07-21 19:16 +0000 [r278539]  David Vossel <dvossel@digium.com>

	* tests/test_func_file.c: make func_file unit test's category
	  consistent with other tests

2010-07-21 19:11 +0000 [r278538]  Terry Wilson <twilson@digium.com>

	* channels/iax2-parser.h, include/asterisk/crypto.h,
	  main/aescrypt.c (removed), include/asterisk/aes_internal.h
	  (removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c
	  (removed), main/aesopt.h (removed), include/asterisk/aes.h
	  (removed), main/aeskey.c (removed), pbx/pbx_dundi.c,
	  channels/chan_iax2.c, res/res_crypto.exports.in,
	  pbx/dundi-parser.h: Remove built-in AES code and use optional_api
	  instead Review: https://reviewboard.asterisk.org/r/793/

2010-07-21 18:52 +0000 [r278536]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: send "423 Interval too small" Response to
	  Subscribe with Expires less that min allowed [RFC3265]3.1.6.1....
	  The notifier MAY also check that the duration in the "Expires"
	  header is not too small. If and only if the expiration interval
	  is greater than zero AND smaller than one hour AND less than a
	  notifier- configured minimum, the notifier MAY return a "423
	  Interval too small" error which contains a "Min-Expires" header
	  field. The "Min- Expires" header field is described in SIP [1].

2010-07-21 17:44 +0000 [r278501]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Fix invalid test
	  for rxisoffhook in FXO channels This fixes some cases of no
	  outgoing calls on FXO before an incoming call. Remove an
	  unnecessary testing of an "off-hook" bit from DAHDI for FXO
	  (KS/GS) channels.In some cases the bit would not be initialized
	  properly before the first inbound call and thus prevent an
	  outgoing call. If those tests are actually required by anybody,
	  they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c
	  . (closes issue #14577) Reported by: jkroon Patches:
	  asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by
	  frawd (license 610) Tested by: frawd Review:
	  https://reviewboard.asterisk.org/r/699/

2010-07-21 16:15 +0000 [r278465]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c: Use poll() instead of select() in
	  res_timing_pthread to avoid stack corruption. This code did not
	  properly check FD_SETSIZE to ensure that it did not try to
	  select() on fds that were too large. Switching to poll() removes
	  the limitation on the maximum fd value. (closes issue #15915)
	  Reported by: keiron (closes issue #17187) Reported by: Eddie
	  Edwards (closes issue #16494) Reported by: Hubguru (closes issue
	  #15731) Reported by: flop (closes issue #12917) Reported by:
	  falves11 (closes issue #14920) Reported by: vrban (closes issue
	  #17199) Reported by: aleksey2000 (closes issue #15406) Reported
	  by: kowalma (closes issue #17438) Reported by: dcabot (closes
	  issue #17325) Reported by: glwgoes (closes issue #17118) Reported
	  by: erikje possibly other issues, too ...

2010-07-21 15:56 +0000 [r278463]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: Ensure realtime conferences are treated the
	  same as static conferences when trying to find an empty one.
	  Also, parse the useropts properly, when retrieving from realtime,
	  and add them to the existing flags. (closes issue #17502)
	  Reported by: kenji Patches: 20100720__issue17502.diff.txt
	  uploaded by tilghman (license 14) Tested by: kenji

2010-07-21 15:54 +0000 [r278426-278462]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax_spandsp.c: Properly show the current page being
	  transfered for 'fax show session'

	* channels/chan_sip.c: Properly set the port number for UDPTL media
	  sessions.

	* res/res_fax.c: Don't print failure status when the remote end
	  hangs up, it may not be an actual failure.

2010-07-21 13:02 +0000 [r278425]  Russell Bryant <russell@digium.com>

	* main/features.c, UPGRADE.txt, configs/features.conf.sample:
	  Update documentation for 'comebacktoorigin' in featuers.conf. The
	  documentation for this option did not match the code. Fix that
	  along with some minor cleanups to the code along the way.
	  Document a slight change in behavior (to something that was
	  previously undocumented) in UPGRADE.txt.

2010-07-21 06:45 +0000 [r278393]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Change order so that it more closely
	  matches the related SIP command. (closes issue #17648) Reported
	  by: GMLudo Review: https://reviewboard.asterisk.org/r/789/

2010-07-21 03:53 +0000 [r278361]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: include stat.h for everybody, needed for
	  device2chan

2010-07-20 23:23 +0000 [r278275-278307]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, main/logger.c, CHANGES,
	  contrib/realtime/mysql/queue_log.sql (added),
	  configs/logger.conf.sample: Separate queue_log arguments into
	  separate fields, and allow the text file to be used, even when
	  realtime is used. (closes issue #17082) Reported by: coolmig
	  Patches: 20100720__issue17082.diff.txt uploaded by tilghman
	  (license 14) Tested by: coolmig

	* /, apps/app_voicemail.c: Merged revisions 278261 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20
	  Jul 2010) | 7 lines Delete IMAP messages in reverse order, to
	  ensure reordering after each expunge does not cause deletion of
	  the wrong message. (closes issue #16350) Reported by: noahisaac
	  Patches: 20100623__issue16350.diff.txt uploaded by tilghman
	  (license 14) ........

2010-07-20 22:38 +0000 [r278274]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Reference correct struct member for unlikely
	  event PRI_EVENT_CONFIG_ERR.

2010-07-20 22:26 +0000 [r278272]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c, /, main/features.c,
	  include/asterisk/channel.h: Merged revisions 278167 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20
	  Jul 2010) | 4 lines Do not queue up DTMF frames while a call is
	  on hold. (Fixes ABE-2110) ........

2010-07-20 21:41 +0000 [r278234]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk
	  sends a 4xx error and the other side sends a CANCEl before
	  receiving the 4xx and responding with the ACK, Asterisk will
	  process the CANCEL and send a 487 Request Terminated as a new
	  final response to the INVITE. Since we are issuing a new final
	  response to the INVITE, the old one must be pretend_acked else it
	  will keep retransmitting.

2010-07-20 21:01 +0000 [r278168]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: This commit contains several changes to the way
	  output channel variables are handled. FAX output channel
	  variables will now match the values reported by FAXOPT() and
	  should be set in all failure and success cases. This commit also
	  contains a few modifications to the way FAXOPT() variables are
	  populated in a few spots and fixes for some reference count leaks
	  of the session details structure in some failure cases. Also
	  found and fixed more cases where FAXOPT(status) may not have
	  gotten set. FAX-214 FAX-203

2010-07-20 19:35 +0000 [r278132]  Tilghman Lesher <tlesher@digium.com>

	* cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
	  res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
	  res/res_calendar_caldav.c, formats/format_sln16.c,
	  formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c,
	  main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c,
	  res/res_smdi.c, channels/chan_skinny.c,
	  include/asterisk/module.h, formats/format_pcm.c,
	  channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c,
	  cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c,
	  formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c,
	  res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c,
	  channels/chan_bridge.c, channels/chan_agent.c,
	  formats/format_ogg_vorbis.c, res/res_monitor.c,
	  res/res_calendar_ews.c, res/res_config_curl.c,
	  channels/chan_misdn.c, funcs/func_curl.c,
	  res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c,
	  res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c,
	  cel/cel_radius.c, channels/chan_multicast_rtp.c,
	  apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c,
	  channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
	  res/res_jabber.c, res/res_config_sqlite.c,
	  formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c,
	  res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c,
	  cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c,
	  apps/app_confbridge.c, formats/format_h264.c,
	  res/res_config_ldap.c, addons/chan_mobile.c,
	  formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c,
	  res/res_rtp_asterisk.c, res/res_config_pgsql.c,
	  res/res_calendar_icalendar.c, channels/chan_sip.c,
	  cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c,
	  res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c,
	  channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c,
	  res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c,
	  res/res_timing_pthread.c, channels/chan_h323.c,
	  cel/cel_sqlite3_custom.c, formats/format_g723.c,
	  funcs/func_devstate.c, formats/format_g729.c,
	  addons/res_config_mysql.c: Add load priority order, such that
	  preload becomes unnecessary in most cases

2010-07-20 18:11 +0000 [r278051-278096]  Russell Bryant <russell@digium.com>

	* contrib/scripts/install_prereq: Add a package to install_prereq.

	* channels/chan_local.c: Only call ast_channel_cc_params_init() if
	  allocating a channel succeeds.

2010-07-20 16:50 +0000 [r278024]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 278023 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
	  | 7 lines Off-by-one error (closes issue #16506) Reported by:
	  nik600 Patches: 20100629__issue16506.diff.txt uploaded by
	  tilghman (license 14) ........

2010-07-19 21:07 +0000 [r277945]  Jean Galarneau <jgalarneau@digium.com>

	* /, main/features.c: Merged revisions 277906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
	  7 lines Avoid trying to pickup a parked extension before the park
	  operation is completed. A crash could occur if the extension is
	  picked up while the parking extension is being announced. Testing
	  pu->notquiteyet while searching for a parked extension resolves
	  this crash. (ABE-2418) ........

2010-07-19 17:16 +0000 [r277872-277873]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample,
	  channels/sip/include/sip.h: Fix port setting of external address
	  in SIP. There are two changes here: 1. Since the externip setting
	  can now have a port attached to it, calling it "externip" is
	  misleading. The option is now documented and parsed as
	  "externaddr." This also extends to the "matchexterniplocally"
	  setting. It is now documented and parsed as
	  "matchexternaddrlocally." The old names for the options may still
	  be used, but they are no longer used in the sip.conf.sample file.
	  2. If no port is set for the externaddr, and UDP is the transport
	  to be used, then we will set the port of the externaddr to that
	  of the udpbindaddr. This was how things worked prior to the IPv6
	  merge, so this is a regression fix. (closes issue #17665)
	  Reported by: mmichelson Patches: 17665.diff#2 uploaded by
	  pprindeville (license 347) Tested by: pprindeville

	* tests/test_acl.c: Remove the fe80:1234::1234 test case from
	  test_acl.c The ACL test was failing on Mac OS X because it would
	  convert the above invalid link-local address into fe80::1234
	  while reporting no error from getaddrinfo(). Linux does not do
	  this.

2010-07-19 14:39 +0000 [r277837]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Fix regression with distinctive ring
	  detection. The issue here is that passing an array to a function
	  prohibits the ARRAY_LEN macro from returning the real size. To
	  avoid this the size is now defined and use of ARRAY_LEN is
	  avoided. (closes issue #15718) Reported by: alecdavis Patches:
	  bug15718.patch uploaded by jpeeler (license 325)

2010-07-19 14:17 +0000 [r277814]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/acl.h, main/netsock2.c, main/manager.c,
	  channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c,
	  main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample,
	  channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be
	  configured to match IPv6 networks. This is only relevant for ACLs
	  in chan_sip for now since other channel drivers do not support
	  IPv6 addressing. However, once those channel drivers are
	  outfitted to support IPv6 addressing, the ACLs will already be
	  ready for IPv6 support. https://reviewboard.asterisk.org/r/791

2010-07-17 17:42 +0000 [r277773-277775]  Tilghman Lesher <tlesher@digium.com>

	* /, autoconf/ast_func_fork.m4, configure,
	  include/asterisk/autoconfig.h.in: Merged revisions 277738 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
	  | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
	  working fork()... it's only uclinux that does not. (closes issue
	  #17616) Reported by: pprindeville ........

	* res/res_config_pgsql.c, res/res_config_odbc.c, /,
	  include/asterisk/config.h, main/config.c,
	  addons/res_config_mysql.c: Merged revisions 277568 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16
	  Jul 2010) | 8 lines Since we split values at the semicolon, we
	  should store values with a semicolon as an encoded value. (closes
	  issue #17369) Reported by: gkservice Patches:
	  20100625__issue17369.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman ........

2010-07-17 13:10 +0000 [r277703]  Russell Bryant <russell@digium.com>

	* Makefile, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, makeopts.in: Allow xmllint to be used for XML docs
	  validation. xmllint seems to be more commonly available since it
	  comes with libxml2.

2010-07-17 00:03 +0000 [r277667]  Bradley Latus <brad.latus@gmail.com>

	* res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes
	  issues #17667) Reported by: snuffy

2010-07-16 23:23 +0000 [r277657]  Tim Ringenbach <tim.ringenbach@gmail.com>

	* main/features.c: Merged revisions 277625 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
	  2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
	  attended transfer. ast_bridge_call() clears
	  AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
	  ast_bridge_call() is called for a second bridge on the same
	  channel, and it clears that flag, which still needs to get set
	  for when the original ast_bridge_call() gets control back and
	  checks it. Review: https://reviewboard.asterisk.org/r/741
	  ........

2010-07-16 21:24 +0000 [r277530]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 277497 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
	  2010) | 4 lines Default to no udptl error correction so that
	  error correction will be disabled in the event that the remote
	  end indicates that they do not support the error correction mode
	  we requested. FAX-128 ........

2010-07-16 21:16 +0000 [r277488]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_queue.c: Fix reporting estimated queue hold time. Just
	  say the number of seconds (after minutes) rather than doing some
	  incorrect calculation with respect to minutes. (closes issue
	  #17498) Reported by: corruptor Patches: holdesecs_bug.diff
	  uploaded by corruptor (license 253)

2010-07-16 20:35 +0000 [r277484]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/sched.h, main/sched.c: Finally, a method that
	  really fixes the assertions in chan_iax2.c related to cancelling
	  lagid. No, replacing usleep(1) with sched_yield() did not have an
	  effect.

2010-07-16 20:27 +0000 [r277467]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 277419 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16
	  Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when
	  reloading dahdi module During a reload, the priexclusive and
	  outsignalling parameters are not read in from the config file as
	  intended. Unfortunately, they get set to defaults as a result.
	  This patch makes sure that they do not get set to defaults during
	  a reload. (closes issue #17441) Reported by: mtryfoss Patches:
	  issue17441_v1.4.patch uploaded by rmudgett (license 664)
	  issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
	  issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
	  by: rmudgett ........

2010-07-16 20:25 +0000 [r277452]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
	  (added): Add documentation for MOH realtime fields

2010-07-16 19:32 +0000 [r277409]  Matthew Nicholson <mnicholson@digium.com>

	* tests/test_devicestate.c: updated devicestate test for device
	  state changes

2010-07-16 19:22 +0000 [r277366]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_queue.c: Add missing handling for ringing state for use
	  with queue empty options. (closes issue #17471) Reported by:
	  jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056)

2010-07-16 18:31 +0000 [r277331]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 277327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
	  2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
	  extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
	  Reported by: francesco_r Patches: pbx.c.patch uploaded by
	  viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
	  ........

2010-07-16 18:14 +0000 [r277263]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 277261 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
	  | 5 lines If variable gotten is not set, will segfault on
	  Solaris. (closes issue #17636) Reported by: bklang ........

2010-07-16 18:05 +0000 [r277250-277262]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Print f->subclass.integer instead of f->subclass.
	  (fix build breakage introduced in r277250)

	* main/channel.c, /: Merged revisions 277247 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
	  2010) | 4 lines For pass through DTMF tones, measure the actual
	  duration between the begin and end packets on the wire. If it is
	  detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
	  emulation. AST-362 ........

2010-07-16 17:13 +0000 [r277183]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, apps/app_amd.c: Merged revisions 277182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
	  2010) | 8 lines Total analysis time error with SIP and silence
	  suppression When using app_amd with SIP providers that have
	  silence suppression on, the iTotalTime count increases
	  exponentially. (closes issue #17656) Reported by: juls ........

2010-07-16 16:25 +0000 [r277175]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/reqresp_parser.c: Fix up some weird indentation
	  problems in reqresp_parser.c

2010-07-16 15:20 +0000 [r277143]  Sean Bright <sean@malleable.com>

	* main/translate.c: Avoid crashing when installing a duplicate
	  translation path with a lower cost. (closes issue #17092)
	  Reported by: moy Patches: translate.rev254273.patch uploaded by
	  moy (license 222) Tested by: moy

2010-07-16 13:40 +0000 [r277103]  Eliel C. Sardanons <eliels@gmail.com>

	* CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file.

2010-07-16 13:32 +0000 [r276950-277102]  Olle Johansson <oej@edvina.net>

	* main/dnsmgr.c, main/srv.c: Formatting changes

	* channels/chan_sip.c: Formatting fixes

	* configs/sip.conf.sample: Clarify syntax changes

	* CREDITS: Adding a few more to the list of CREDITS

	* channels/chan_sip.c: Formatting changes (guideline corrections)
	  Found a unused bag of curly brackets under my table. I always
	  wondered where they had gone. They where indeed needed in
	  chan_sip.c

	* CREDITS: Adding a few more credits

	* channels/chan_sip.c, doc/tex/channelvariables.tex,
	  configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add
	  ability to configure the Max-Forwards header in the dialplan, as
	  well as in sip.conf configuration for the channel and for
	  devices. The Max-Forwards header is used to prevent loops in a
	  SIP network. Each intermediary, like SIP proxys and SBCs,
	  decrement this counter and detects when it reaches zero, at which
	  point the SIP request is nicely killed in a SIP-friendly way.
	  Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel
	  for the review and good advice.

	* CHANGES, apps/app_queue.c: Add a dialplan function to check if a
	  queue exists: QUEUE_EXISTS Review:
	  https://reviewboard.asterisk.org/r/777/

2010-07-16 06:04 +0000 [r276910-276911]  Tilghman Lesher <tlesher@digium.com>

	* res/res_jabber.c: And yet one more

	* res/res_jabber.c: "Item may be used uninitialized in this
	  function."

2010-07-16 05:42 +0000 [r276909]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix reversed logic of if statement. Found
	  based on message from Philip Prindeville on the Asterisk
	  Developers mailing list.

2010-07-16 05:38 +0000 [r276830-276908]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Detect the --dynamic-list flag a bit
	  better

	* configure, main/Makefile, configure.ac, makeopts.in: Fix build on
	  FreeBSD

	* tests/test_utils.c: Fix trunk build for Mac OS X 10.6

	* contrib/realtime/mysql/iaxfriends.sql,
	  contrib/realtime/mysql/meetme.sql,
	  contrib/realtime/postgresql/realtime.sql,
	  contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain
	  the maximum IPv6 address. Also, update meetme to the full list of
	  supported fields.

	* configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within
	  m4_ifval, so it does not get prematurely expanded. (closes issue
	  #17654) Reported by: pprindeville Patches: issue17654.diff
	  uploaded by qwell (license 4) Tested by: qwell, pprindeville

2010-07-15 20:21 +0000 [r276788]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Correct not setting the bindport before
	  attempting to open the socket. Related to changes from 276571, I
	  was accidentally testing with a port set in my configuration
	  causing me to miss this. Also moved the TCP handling as well to
	  occur before build_peer is called.

2010-07-15 19:46 +0000 [r276731-276769]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, configure.ac: Define LLONG_MAX on
	  systems that do not have it. (closes issue #17644) Reported by:
	  pprindeville

	* configure, main/Makefile, autoconf/ast_gcc_attribute.m4,
	  configure.ac, makeopts.in: Fix linking asterisk on CentOS 5,
	  which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review:
	  https://reviewboard.asterisk.org/r/790/

2010-07-15 13:51 +0000 [r276653]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 276652 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
	  | 2 lines In a perfect world, the frame source would never be
	  NULL. In the meantime, don't crash when it is. ........

2010-07-15 12:21 +0000 [r276616]  Russell Bryant <russell@digium.com>

	* contrib/scripts/install_prereq: Add lua5.1 to the handy dandy
	  list of packages.

2010-07-14 22:58 +0000 [r276571]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Fix MWI notification transmission problems
	  over SIP. MWI updates were not being sent if no messages were
	  found in the event cache. This was corrected since a phone may
	  need to clear its MWI status configured previously from another
	  mailbox. Upon module or sip reload, MWI updates could not be sent
	  due to the sipsock socket not being set early enough in
	  reload_config. The code handling the descriptor assignment and
	  such has simply been moved before the call to build_peer. Issuing
	  a sip reload cleared the IP address of the peer, but skipped
	  checking the database for registration information. The database
	  is now checked both for sip reload and actually reloading the
	  module. If a transmission occurs before the do_monitor thread has
	  started, do not attempt to send a signal to it. (closes issue
	  #17398) Reported by: ip-rob

2010-07-14 22:32 +0000 [r276570]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
	  main/acl.c: Fix errors where incorrect address information was
	  printed. ast_sockaddr_stringiy_fmt (which is call by all
	  ast_sockaddr_stringify* functions) uses thread-local storage for
	  storing the string that it creates. In cases where
	  ast_sockaddr_stringify_fmt was being called twice within the same
	  statement, the result of one call would be overwritten by the
	  result of the other call. This usually was happening in
	  printf-like statements and was resulting in the same stringified
	  addressed being printed twice instead of two separate addresses.
	  I have fixed this by using ast_strdupa on the result of stringify
	  functions if they are used twice within the same statement. As
	  far as I could tell, there were no instances where a pointer to
	  the result of such a call were saved anywhere, so this is the
	  only situation I could see where this error could occur.

2010-07-14 21:29 +0000 [r276531]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_h323.c: Make compile again.

2010-07-14 21:11 +0000 [r276490-276493]  Tilghman Lesher <tlesher@digium.com>

	* main/loader.c: Oops, merge reverted this fix.

	* include/asterisk/adsi.h, include/asterisk/agi.h,
	  include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile,
	  tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c
	  (removed), res/res_adsi.c, res/res_crypto.c,
	  res/res_crypto.exports.in (added), res/res_adsi.exports.in,
	  main/loader.c, include/asterisk/optional_api.h: Remove the old
	  stub files, preferring the optional_api method. (closes issue
	  #17475) Reported by: tilghman Review:
	  https://reviewboard.asterisk.org/r/695/

2010-07-14 20:15 +0000 [r276441]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c: Don't try to call an embedded module's
	  backup_globals() function until after confirming it exists.

2010-07-14 19:51 +0000 [r276439]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: handle special case were "200 Ok" to pending
	  INVITE never receives ACK Unlike most responses, the 200 Ok to a
	  pending INVITE Request is acknowledged by an ACK Request. If the
	  ACK Request for this Response is not received the previous
	  behavior was to immediately destroy the dialog and hangup the
	  channel. Now in an effort to be more RFC compliant, instead of
	  immediately destroying the dialog during this special case,
	  termination is done with a BYE Request as the dialog is
	  technically confirmed when the 200 Ok is sent even if the ACK is
	  never received. The behavior of immediately hanging up the
	  channel remains. This only affects how dialog termination
	  proceeds for this one special case. RFC 3261 section 13.3.1.4 "If
	  the server retransmits the 2xx response for 64*T1 seconds without
	  receiving an ACK, the dialog is confirmed, but the session SHOULD
	  be terminated. This is accomplished with a BYE, as described in
	  Section 15."

2010-07-14 16:58 +0000 [r276393]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_vpb.cc, channels/chan_sip.c,
	  include/asterisk/channel.h, channels/sig_pri.c,
	  channels/chan_iax2.c, main/cel.c, channels/chan_oss.c,
	  main/channel.c, main/cdr.c, channels/chan_jingle.c,
	  channels/chan_usbradio.c, channels/chan_dahdi.c,
	  channels/chan_phone.c, channels/sig_analog.c,
	  channels/chan_misdn.c, channels/chan_skinny.c,
	  channels/chan_h323.c, res/snmp/agent.c, apps/app_amd.c,
	  funcs/func_callerid.c, channels/sig_ss7.c, channels/chan_mgcp.c:
	  Expand the caller ANI field to an ast_party_id Expand the ani
	  field in ast_party_caller and ast_party_connected_line to an
	  ast_party_id. This is an extension to the ast_callerid
	  restructuring patch in review:
	  https://reviewboard.asterisk.org/r/702/ Review:
	  https://reviewboard.asterisk.org/r/744/

2010-07-14 16:40 +0000 [r276392]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: collapse debug code in retrans_pkt into
	  separate lines I've been working in this function a bunch lately,
	  and these huge debug strings are getting annoying.

2010-07-14 16:39 +0000 [r276391]  Richard Mudgett <rmudgett@digium.com>

	* res/snmp/agent.c: Make compile again.

2010-07-14 16:36 +0000 [r276389]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Do not skip sending MWI for a peer if an
	  address is defined. Really just a merge mistake from IPv6

2010-07-14 16:09 +0000 [r276349]  Tim Ringenbach <tim.ringenbach@gmail.com>

	* cel/cel_pgsql.c, doc/tex/celdriver.tex, doc/tex/cdrdriver.tex:
	  Fix documentation for pgsql cel and cdr, and slightly improve
	  pgsql_cel. Change the documented pgsql schema to use "timestamp"
	  instead of "time", as the latter is only a time without a date.
	  Added some missing columns for cel's pgsql schema, and corrected
	  spelling on some others. Updated cel's uniqueid size to be the
	  same as the cdr. Added id column to cel's pgsql schema and
	  updated code to allow unknown columns to get their default value
	  instead of forcing 0 or empty string. Added microseconds to the
	  timestamp cel logs to pgsql. Review:
	  https://reviewboard.asterisk.org/r/734

2010-07-14 15:48 +0000 [r276347]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_local.c, addons/chan_ooh323.c,
	  apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
	  channels/sig_analog.h, apps/app_amd.c, channels/sig_ss7.c,
	  apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_fax.c,
	  channels/chan_agent.c, apps/app_disa.c,
	  include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c,
	  funcs/func_redirecting.c (removed), channels/chan_misdn.c,
	  apps/app_macro.c, apps/app_zapateller.c, apps/app_voicemail.c,
	  channels/chan_unistim.c, tests/test_substitution.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
	  apps/app_readexten.c, channels/chan_gtalk.c, apps/app_followme.c,
	  include/asterisk/callerid.h, main/cdr.c, main/channel.c,
	  channels/chan_phone.c, main/dial.c, apps/app_setcallerid.c,
	  apps/app_osplookup.c, main/manager.c, apps/app_minivm.c,
	  res/res_agi.c, main/app.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  apps/app_parkandannounce.c, apps/app_while.c,
	  funcs/func_dialplan.c, channels/chan_sip.c, UPGRADE.txt,
	  channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
	  channels/chan_oss.c, channels/chan_usbradio.c,
	  channels/chan_jingle.c, funcs/func_blacklist.c,
	  apps/app_directed_pickup.c, main/file.c,
	  funcs/func_connectedline.c (removed), channels/chan_h323.c,
	  main/callerid.c, res/snmp/agent.c, apps/app_sms.c,
	  apps/app_stack.c, funcs/func_callerid.c: ast_callerid
	  restructuring The purpose of this patch is to eliminate struct
	  ast_callerid since it has turned into a miscellaneous collection
	  of various party information. Eliminate struct ast_callerid and
	  replace it with the following struct organization: struct
	  ast_party_name { char *str; int char_set; int presentation;
	  unsigned char valid; }; struct ast_party_number { char *str; int
	  plan; int presentation; unsigned char valid; }; struct
	  ast_party_subaddress { char *str; int type; unsigned char
	  odd_even_indicator; unsigned char valid; }; struct ast_party_id {
	  struct ast_party_name name; struct ast_party_number number;
	  struct ast_party_subaddress subaddress; char *tag; }; struct
	  ast_party_dialed { struct { char *str; int plan; } number; struct
	  ast_party_subaddress subaddress; int transit_network_select; };
	  struct ast_party_caller { struct ast_party_id id; char *ani; int
	  ani2; }; The new organization adds some new information as well.
	  * The party name and number now have their own presentation value
	  that can be manipulated independently. ISDN supplies the
	  presentation value for the name and number at different times
	  with the possibility that they could be different. * The party
	  name and number now have a valid flag. Before this change the
	  name or number string could be empty if the presentation were
	  restricted. Most channel drivers assume that the name or number
	  is then simply not available instead of indicating that the name
	  or number was restricted. * The party name now has a character
	  set value. SIP and Q.SIG have the ability to indicate what
	  character set a name string is using so it could be presented
	  properly. * The dialed party now has a numbering plan value that
	  could be useful to have available. The various channel drivers
	  will need to be updated to support the new core features as
	  needed. They have simply been converted to supply current
	  functionality at this time. The following items of note were
	  either corrected or enhanced: * The CONNECTEDLINE() and
	  REDIRECTING() dialplan functions were consolidated into
	  func_callerid.c to share party id handling code. * CALLERPRES()
	  is now deprecated because the name and number have their own
	  presentation values. * Fixed app_alarmreceiver.c
	  write_metadata(). The workstring[] could contain garbage. It also
	  can only contain the caller id number so using
	  ast_callerid_parse() on it is silly. There was also a typo in the
	  CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse()
	  on the channel's caller id number string. ast_callerid_parse()
	  alters the given buffer which in this case is the channel's
	  caller id number string. Then using ast_shrink_phone_number()
	  could alter it even more. * Fixed caller ID name and number
	  memory leak in chan_usbradio.c. * Fixed uninitialized char arrays
	  cid_num[] and cid_name[] in sig_analog.c. * Protected access to a
	  caller channel with lock in chan_sip.c. * Clarified intent of
	  code in app_meetme.c sla_ring_station() and dial_trunk(). Also
	  made save all caller ID data instead of just the name and number
	  strings. * Simplified cdr.c set_one_cid(). It hand coded the
	  ast_callerid_merge() function. * Corrected some weirdness with
	  app_privacy.c's use of caller presentation. Review:
	  https://reviewboard.asterisk.org/r/702/

2010-07-14 11:51 +0000 [r276268]  Leif Madsen <lmadsen@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 276267 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010)
	  | 1 line Update documentation for voicemail.conf externpass
	  option. ........

2010-07-13 22:18 +0000 [r276219]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: RFC
	  compliant retransmission timeout Retransmission of packets should
	  not be based on how many packets were sent, but instead on a
	  timeout period. Depending on whether or not the packet is for a
	  INVITE or NON-INVITE transaction, the number of packets sent
	  during the retransmission timeout period will be different, so
	  timing out based on the number of packets sent is not accurate.
	  This patch fixes this by removing the retransmit limit and only
	  stopping retransmission after a timeout period is reached. By
	  default this timeout period is 64*(Timer T1) for both INVITE and
	  non-INVITE transactions. For more information on sip timer values
	  refer to RFC3261 Appendix A. Review:
	  https://reviewboard.asterisk.org/r/749/

2010-07-13 21:42 +0000 [r276206]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/dialog.h, channels/chan_sip.c: Revert early
	  destruction of RTP sessions Some code improperly assumes that the
	  sessions are still there, so revert the change until I can find
	  all of them and fix them.

2010-07-13 19:15 +0000 [r276124-276127]  Russell Bryant <russell@digium.com>

	* /: Recorded merge of revisions 276126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010)
	  | 2 lines Only reset a CDR that exists. ........

	* /, main/features.c: Merged revisions 276123 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
	  | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
	  instead of peer_cdr in the last commit). ........

2010-07-13 19:05 +0000 [r276114-276122]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Oops, XML documentation fix.

	* funcs/func_env.c: It really cannot fail in the places below, but
	  the stupid compiler doesn't know that.

	* funcs/func_env.c: Weird compiler error on Bamboo.

	* funcs/func_env.c, CHANGES, tests/test_func_file.c (added): FILE()
	  now supports line-mode and writing (altering) files. (closes
	  issue #16461) Reported by: skyman Patches:
	  20100622__issue16461.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman Review:
	  https://reviewboard.asterisk.org/r/737/

2010-07-13 17:37 +0000 [r276074]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_meetme.c: Merged revisions 275773 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
	  | 12 lines Make user removals and traversals thread safe in
	  meetme. Race conditions present in meetme involving the user list
	  where a lack of locking has the potential for a user to be
	  removed during a traversal or as in the case of the reporter
	  after checking if the list is empty could cause a crash. Fixing
	  this was done by convering the userlist to an ao2 container.
	  (closes issue #17390) Reported by: Vince Review:
	  https://reviewboard.asterisk.org/r/746/ ........

2010-07-13 17:11 +0000 [r275998]  Terry Wilson <twilson@digium.com>

	* channels/sip/include/dialog.h, channels/chan_sip.c: Destroy RTP
	  fds when we schedule final dialog destruction Since we are only
	  keeping the dialog around for retransmissions at this point and
	  there is no possibility that we are still handling RTP, go ahead
	  and destroy the RTP sessions. Keeping them alive for 32 past when
	  they are used is unnecessary and can lead to problems with having
	  too many open file descriptors, etc.

2010-07-13 16:53 +0000 [r275995]  Russell Bryant <russell@digium.com>

	* /, main/features.c: Merged revisions 275994 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
	  | 14 lines Access peer->cdr directly instead of through a saved
	  off reference. At this point in the code, it is possible that
	  peer_cdr may be invalid. Specifically, in the blind transfer
	  code, CDRs are swapped between channels. So, peer_cdr is no
	  longer == peer->cdr. The scenario that exposed a crash in this
	  code was a blind transfer that hit the system call limit, causing
	  the transferee channel to get destroyed after the transfer
	  attempt failed. Even if it succeeds and this code doesn't crash,
	  this code was still trying to reset a CDR on a channel that was
	  now owned by a different thread, which is a BadThing(tm).
	  (ABE-2417) ........

2010-07-13 14:48 +0000 [r275910]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/realtime_pgsql.sql (removed),
	  contrib/scripts/iax-friends.sql (removed), /,
	  contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
	  (removed), contrib/realtime (added), contrib/realtime/postgresql,
	  contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
	  contrib/realtime/oracle, contrib/scripts/sip-friends.sql
	  (removed), contrib/realtime/mysql/sipfriends.sql,
	  contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
	  (removed), contrib/realtime/mysql/meetme.sql,
	  contrib/realtime/sqlserver: Merged revisions 275909 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
	  Jul 2010) | 2 lines Move SQL scripts into their own
	  database-specific directories. ........

2010-07-13 11:41 +0000 [r275863]  Russell Bryant <russell@digium.com>

	* configs/voicemail.conf.sample,
	  contrib/scripts/voicemailpwcheck.py (added): Add example script
	  for use with the externpasscheck voicemail.conf option. (closes
	  issue #17628) Reported by: lmadsen Tested by: russell, lmadsen
	  Review: https://reviewboard.asterisk.org/r/774/

2010-07-12 23:27 +0000 [r275816]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Don't try to ref authpeer when it isn't set

2010-07-12 17:54 +0000 [r275725]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Add which ITU spec specifies the numbering plan.

2010-07-12 17:21 +0000 [r275682]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 275665 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
	  | 11 lines Change ast_write to not stop generator when called
	  from ast_prod. For SIP channels configured with the
	  progressinband option on, the ringback was being immediately
	  stopped. This problem was due to ast_prod being moved for a
	  deadlock fix in 259858. Prodding the channel after setting up the
	  generator triggered the check in ast_write to stop the generator.
	  The fix here should write the frame the same as was done before
	  the call to ast_prod was moved. (closes issue #17372) Reported
	  by: tech_admin ........

2010-07-12 15:37 +0000 [r275626]  Leif Madsen <lmadsen@digium.com>

	* cdr/cdr_pgsql.c: cdr_pgsql does not detect when a table is found.
	  This change adds an ERROR message to let you know when a failure
	  exists to get the columns from the pgsql database, which
	  typically means that the table does not exist. (closes issue
	  #17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by
	  kobaz (license 834) Tested by: kobaz, russell, lmadsen

2010-07-12 14:55 +0000 [r275587]  Mark Michelson <mmichelson@digium.com>

	* main/netsock2.c: Allow netsock2.c to compile on systems that do
	  not define AI_NUMERICSERV. (closes issue #17617) Reported by:
	  pprindeville Patches: asterisk-trunk-bugid17617.patch uploaded by
	  pprindeville (license 347)

2010-07-12 04:16 +0000 [r275551]  TransNexus OSP Development <support@transnexus.com>

	* configs/osp.conf.sample, apps/app_osplookup.c: Added support for
	  indirect work mode.

2010-07-10 20:49 +0000 [r275509]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_meetme.c: When creating a conference for a unit test, it
	  is not mandatory to open a dahdi pseudo channel, so if we fail
	  doing it, continue creating the conference.

2010-07-10 14:48 +0000 [r275424-275467]  Russell Bryant <russell@digium.com>

	* CHANGES: Make indentation consistent, move some queue features to
	  the queue section.

	* CREDITS, channels/chan_unistim.c, configs/unistim.conf.sample,
	  CHANGES: Add support for devices with less than 3 lines on the
	  LCD. (closes issue #17600) Reported by: minaguib Patches:
	  ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
	  Tested by: minaguib

	* main/features.c, configs/features.conf.sample: Fix some issues
	  related to dynamic feature groups in features.conf. The bridge
	  handling code did not properly consider feature groups when
	  setting parameters that would affect whether or not a native
	  bridge would be attempted. If DYNAMIC_FEATURES only include a
	  feature group, a native bridge would occur that may prevent
	  features from working. Fix a bug in verbose output that would
	  show the key mapping as empty if it was using the default mapping
	  and not a custom mapping in the feature group. Add feature groups
	  to the output of "features show". Adjust the feature execution
	  logic to match that of the logic when executing a feature that
	  was not configured through a feature group. Update
	  features.conf.sample to show that an '=' is still required if
	  using the default key mapping from [applicationmap]. Finally,
	  clean up a little bit of formatting to better coform to coding
	  guidelines while in the area. (closes issue #17589) Reported by:
	  lmadsen Patches: issue_17589.rev4.txt uploaded by russell
	  (license 2) Tested by: russell, lmadsen

2010-07-09 20:58 +0000 [r275385]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix error in parsing SIP registry strings
	  from ASTdb. It was essentially an off-by-one error. The easiest
	  way to fix this was to use the handy-dandy
	  AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the
	  registration string out. Tested and it works wonderfully.

2010-07-09 20:01 +0000 [r275312]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, channels/chan_iax2.c: Get more information
	  about the Bamboo test failures

2010-07-09 19:58 +0000 [r275309-275310]  Russell Bryant <russell@digium.com>

	* main/features.c: Add missing ao2_iterator_destroy().

	* apps/app_voicemail.c: Fix compile error.

2010-07-09 19:46 +0000 [r275308]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix port parsing in check_via. If a Via
	  header contained an IPv6 address, we would not properly parse the
	  port. We would instead get the information after the first colon
	  in the address. (closes issue #17614) Reported by: oej Patches:
	  diff uploaded by sperreault (license 252)

2010-07-09 19:32 +0000 [r275307]  Paul Belanger <paul.belanger@polybeacon.com>

	* CHANGES, apps/app_voicemail.c: Include rdnis in msgXXXX.txt file.
	  (closes issue #17566) Reported by: outcast Patches:
	  voicemail-rdnis.patch uploaded by outcast (license 1071) Tested
	  by: outcast

2010-07-09 19:29 +0000 [r275294]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix an issue where the port for p->ourip was
	  being set to 0. This should fix all the CDR tests that were not
	  passing. When they would originate a call, all fields in the
	  INVITE that contained the source port would have the port set to
	  0. Most troubling of these was the Contact header. Tests are
	  passing locally now and should also pass on the bamboo build
	  agents.

2010-07-09 19:21 +0000 [r275249]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_sip.c: Merged revisions 275241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
	  2010) | 8 lines Fix logging message for stale nonce. (closes
	  issue #17582) Reported by: kenner Patches: chan_sip.c.diff
	  uploaded by kenner (license 1040) Tested by: lmadsen ........

2010-07-09 18:55 +0000 [r275227]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and
	  Bamboo still fails...

2010-07-09 18:24 +0000 [r275186]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/loader.c: Merged revisions 275182 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
	  2010) | 2 lines give a better error message when attempting to
	  unload a module that is not loaded ........

2010-07-09 18:21 +0000 [r275172]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, channels/chan_iax2.c: Add some diagnostic
	  feedback to our data tests

2010-07-09 18:11 +0000 [r275147]  Russell Bryant <russell@digium.com>

	* configs/features.conf.sample: Move parking lot sample config out
	  from the middle of dynamic features sample config.

2010-07-09 17:50 +0000 [r275144]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/loader.c: Merged revisions 275143 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
	  2010) | 2 lines don't unload modules that returned
	  AST_MODULE_LOAD_DECLINE when they were loaded ........

2010-07-09 17:00 +0000 [r275105]  Tilghman Lesher <tlesher@digium.com>

	* main/netsock2.c, tests/test_substitution.c, tests/test_heap.c,
	  apps/app_meetme.c, tests/test_gosub.c, funcs/func_strings.c,
	  tests/test_event.c, channels/sip/reqresp_parser.c,
	  channels/chan_iax2.c, tests/test_stringfields.c,
	  tests/test_time.c, tests/test_devicestate.c, tests/test_utils.c,
	  main/features.c, res/res_agi.c, include/asterisk/netsock2.h,
	  tests/test_astobj2.c, channels/chan_sip.c,
	  tests/test_ast_format_str_reduce.c, tests/test_app.c,
	  funcs/func_math.c, include/asterisk/channel.h,
	  tests/test_sched.c, tests/test_pbx.c, tests/test_strings.c,
	  main/data.c, tests/test_skel.c, tests/test_acl.c,
	  channels/sip/dialplan_functions.c, tests/test_aoc.c, main/test.c,
	  channels/sip/config_parser.c, res/res_timing_kqueue.c,
	  apps/app_voicemail.c: Kill some startup warnings and errors and
	  make some messages more helpful in tracking down the source.

2010-07-09 16:39 +0000 [r275104]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Return logic of sip_debug_test_addr() to its
	  original functionality.

2010-07-09 16:05 +0000 [r275028]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_dial.c, /: Merged revisions 275027 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
	  2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
	  going into the pbx via the G option in app_dial (closes issue
	  #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
	  uploaded by mnicholson (license 96) Tested by: jamicque,
	  mnicholson ........

2010-07-09 15:35 +0000 [r275022]  Russell Bryant <russell@digium.com>

	* include/asterisk/test.h, /, main/test.c: Merged revisions 275021
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
	  | 4 lines Document that a leading and trailing slash is expected
	  for test categories. Also, emit a warning if a test is registered
	  without one of these. ........

2010-07-09 14:27 +0000 [r274984]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/reqresp_parser.c: Fix sip_uri_parse test comparison.
	  Part of the change with the IPv6 changes is to treat a host:port
	  as a single 'domain' entity. This test was not updated to have
	  the correct expectation after calling parse_uri().

2010-07-09 13:30 +0000 [r274909-274947]  <simon.perreault@viagenie.ca>

	* channels/chan_sip.c: Copy the address into the peer structure
	  after we set the default port

	* main/netsock2.c: Sadly we can't dereference a pointer cast and
	  use it as an lvalue without getting this warning (at least with
	  gcc 4.4.4): netsock2.c:492: warning: dereferencing pointer
	  ‘({anonymous})’ does break strict-aliasing rules So we're back to
	  using memcpy()...

2010-07-09 12:48 +0000 [r274907]  Russell Bryant <russell@digium.com>

	* include/asterisk/indications.h: Extend length limit on country
	  name in indications.conf.

2010-07-09 11:06 +0000 [r274866]  Olle Johansson <oej@edvina.net>

	* configs/cdr.conf.sample, cdr/cdr_csv.c: Make it possible to
	  disable individual cdr files per accountcode in cdr_csv Review:
	  https://reviewboard.asterisk.org/r/678/

2010-07-08 23:46 +0000 [r274827-274828]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_jingle.c, channels/chan_h323.c,
	  channels/chan_gtalk.c: Fix calls of ast_sockaddr_from_sin() from
	  IPv6 integration.

	* addons/chan_ooh323.c: Fix compile of chan_ooh323.c from IPv6
	  integration.

2010-07-08 22:16 +0000 [r274783-274786]  Mark Michelson <mmichelson@digium.com>

	* /: And the automerge property.

	* /: Delete properties I merged during v6-new merge.

	* channels/chan_unistim.c, include/asterisk/acl.h, main/netsock2.c
	  (added), channels/sip/include/dialog.h,
	  channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
	  main/rtp_engine.c, /, channels/sip/reqresp_parser.c,
	  include/asterisk/tcptls.h, channels/chan_gtalk.c,
	  channels/chan_iax2.c, main/config.c, res/res_rtp_multicast.c,
	  main/manager.c, channels/chan_skinny.c,
	  channels/sip/include/globals.h, main/http.c, main/app.c,
	  include/asterisk/netsock2.h (added), apps/app_externalivr.c,
	  configs/sip.conf.sample, include/asterisk/rtp_engine.h,
	  channels/sip/include/sip.h, channels/chan_mgcp.c,
	  channels/sip/include/reqresp_parser.h, res/res_rtp_asterisk.c,
	  main/dnsmgr.c, channels/chan_sip.c, include/asterisk/config.h,
	  main/acl.c, CHANGES, channels/chan_jingle.c, main/tcptls.c,
	  channels/sip/dialplan_functions.c, channels/chan_h323.c,
	  include/asterisk/dnsmgr.h: Add IPv6 to Asterisk. This adds a
	  generic API for accommodating IPv6 and IPv4 addresses within
	  Asterisk. While many files have been updated to make use of the
	  API, chan_sip and the RTP code are the files which actually
	  support IPv6 addresses at the time of this commit. The way has
	  been paved for easier upgrading for other files in the near
	  future, though. Big thanks go to Simon Perrault, Marc Blanchet,
	  and Jean-Philippe Dionne for their hard work on this. (closes
	  issue #17565) Reported by: russell Patches:
	  asteriskv6-test-report.pdf uploaded by russell (license 2)
	  Review: https://reviewboard.asterisk.org/r/743

2010-07-08 22:05 +0000 [r274773-274782]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Generate a correct AstData string for
	  ast_callerid.cid_ton

	* main/channel.c: Fix trunk compile.

2010-07-08 14:48 +0000 [r274727]  Eliel C. Sardanons <eliels@gmail.com>

	* main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
	  include/asterisk/indications.h, channels/chan_agent.c,
	  include/asterisk/channel.h, include/asterisk/cdr.h,
	  include/asterisk/data.h, channels/chan_iax2.c, apps/app_queue.c,
	  main/indications.c, main/channel.c, main/cdr.c,
	  channels/chan_dahdi.c, main/data.c, res/res_odbc.c,
	  apps/app_voicemail.c: Implement AstData API data providers as
	  part of the GSOC 2010 project, midterm evaluation. Review:
	  https://reviewboard.asterisk.org/r/757/

2010-07-07 20:09 +0000 [r274686]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes some ref count issues introduced by
	  r274539

2010-07-07 18:32 +0000 [r274595-274639]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Add missing conditional around chan_dahdi
	  mfcr2_skip_category config parameter.

	* channels/chan_dahdi.c, /: Merged revisions 274579 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
	  Jul 2010) | 1 line Close the DAHDI FD on error when processing
	  chan_dahdi toneduration config parameter. ........

2010-07-07 16:40 +0000 [r274540]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Set proper FAXOPT(status), FAXOPT(statusstr), and
	  FAXOPT(error) values where possible. Previously some failure
	  cases did not result in proper FAXOPT values. FAX-203

2010-07-07 16:21 +0000 [r274539]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Use the relatedpeer field of a sip_pvt
	  during INVITE processing. Review:
	  https://reviewboard.asterisk.org/r/629

2010-07-07 07:07 +0000 [r274492]  TransNexus OSP Development <support@transnexus.com>

	* configs/osp.conf.sample, doc/osp.txt: Changed OSP TCP port from
	  1080 to 5045.

2010-07-07 06:32 +0000 [r274418-274491]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES, apps/app_voicemail.c: Also run the externnotify script
	  when the pollmailboxes thread notices a change.

	* /, configs/say.conf.sample: Merged revisions 274417 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07
	  Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also
	  add the crazy British numbers. (closes issue #16102) Reported by:
	  Delvar Patches: say.conf.fix.patch uploaded by Delvar (license
	  908) (plus a few additional fixes and simplifications by me)
	  ........

2010-07-06 22:23 +0000 [r274316]  Jeff Peeler <jpeeler@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 274283 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06
	  Jul 2010) | 7 lines Correct sip.conf.sample comments for
	  prematuremedia option. (closes issue #17513) Reported by: festr
	  Patches: patch uploaded by festr (license 443) ........

2010-07-06 22:15 +0000 [r274284]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c, UPGRADE.txt: Merged revisions 274280 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
	  | 9 lines Add option to not do a call forward on 482 Loop
	  Detected Asterisk has always set up a forwarded call when
	  receiving a 482 Loop Detected. This prevents handling the call
	  failure by just continuing on in the dialplan. Since this would
	  be a change in behavior, the new option to disable this behavior
	  is forwardloopdetected which defaults to 'yes'. Review:
	  https://reviewboard.asterisk.org/r/764/ ........ (no option for
	  trunk, just changing the behavior)

2010-07-06 22:09 +0000 [r274281]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Status shows all non-CRC4 lines as
	  "yellow", even if "yellow" was not in the bitfield.

2010-07-06 19:53 +0000 [r274243]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Properly detect and report invalid maxrate and
	  maxrate values in the FAXOPT dialplan function. Also make
	  fax_rate_str_to_int() return an unsigned int and return 0 instead
	  of -1 in the event of an error. FAX-202

2010-07-06 14:31 +0000 [r274164]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, /: Merged revisions 274157 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue,
	  06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
	  accepted. A recent check was added to ensure that we did not
	  erroneously detect duplicate DTMF when we received packets out of
	  order. The problem was that the check did not account for the
	  fact that the seqno of an RTP stream will roll over back to 0
	  after hitting 65535. Now, we have a secondary check that will
	  ensure that the seqno rolling over will not cause us to stop
	  accepting DTMF. (closes issue #17571) Reported by: mdeneen
	  Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
	  60) Tested by: richardf, maxochoa, JJCinAZ ........

2010-07-06 06:01 +0000 [r274053]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Uh, yeah.

2010-07-05 13:53 +0000 [r273886]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, main/config.c: Merged revisions 273884 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
	  2010) | 8 lines Remove extra line breaks from 'core show config
	  mappings' (closes issue #17583) Reported by: pabelanger Patches:
	  issue17583.patch uploaded by pabelanger (license 224) Tested by:
	  lmadsen ........

2010-07-03 02:36 +0000 [r273714-273830]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /, channels/chan_agent.c,
	  channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
	  273793 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
	  | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
	  fails, to help catch potentially large software bugs. (closes
	  issue #17407) Reported by: pdf Patches:
	  20100527__issue17407.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/751/ ........

	* main/autoservice.c, /: Merged revisions 273717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
	  | 8 lines Autoservice loop optimization causes a busy loop, when
	  channels are serviced while in hangup. (closes issue #17564)
	  Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
	  uploaded by tilghman (license 14) Tested by: ramonpeek ........

	* apps/app_queue.c: The switch fallthrough could create some
	  errorneous situations, so best to force directly to the default
	  case.

2010-07-02 15:57 +0000 [r273641]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c, channels/chan_misdn.c,
	  channels/chan_sip.c, main/say.c, main/fixedjitterbuf.c,
	  res/res_agi.c, channels/chan_h323.c, main/utils.c,
	  channels/chan_iax2.c, addons/chan_mobile.c, apps/app_rpt.c,
	  channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
	  apps/app_while.c: Fix various typos reported by Lintian (Also fix
	  the typos in the comments)

2010-07-01 22:16 +0000 [r273566]  Russell Bryant <russell@digium.com>

	* /, main/datastore.c: Merged revisions 273565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
	  | 7 lines Don't return a partially initialized datastore. If
	  memory allocation fails in ast_strdup(), don't return a partially
	  initialized datastore. Bad things may happen. (related to
	  ABE-2415) ........

2010-07-01 20:28 +0000 [r273522]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_meetme.c: Merged revisions 273474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
	  | 14 lines Allow admin user to join conference without using
	  admin mode and no user pin. Configuring the conference in
	  meetme.conf like the following: conf => 2345,,6666 did not prompt
	  for pin when used without admin mode. This meant that the
	  conference could not be joined as an admin even if the user knew
	  the correct pin. The original bug report was submitted claiming
	  that the blank user pin should deny entry into the conference. I
	  think a better way to handle this would be with a feature
	  enhancement that used the following syntax: conf => 2345,X,6666 -
	  where X denotes no acceptable pin allowed (closes issue #15704)
	  Reported by: modelnine ........

2010-07-01 19:34 +0000 [r273464]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c: Properly handle failures of fax->start_session()
	  FAX-177

2010-07-01 16:40 +0000 [r273427]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/include/sip.h: correct handling
	  of get_destination return values A failure when calling the
	  get_destination can mean multiple things. If the extension is not
	  found, a 404 error is appropriate, but if the URI scheme is
	  incorrect, a 404 is not approperiate. This patch adds the
	  get_destination_result enum to differentiate between these and
	  other failure types. The only logical difference in this patch is
	  that we now send a "416 Unsupported URI scheme" response instead
	  of a "404" when the scheme is not recognized. This indicates to
	  the initiator of the INVITE to retry the request with a correct
	  URI.

2010-07-01 15:12 +0000 [r273355]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_meetme.c: Merged revisions 273354 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
	  | 12 lines Ensure channel placed in meetme in ringing state is
	  properly hung up. An outgoing channel placed in meetme while
	  still ringing which was then hung up would not exit meetme and
	  the channel was not properly destroyed. Specifically checking for
	  this scenario by looking at the appropriate control frames
	  resolves the issue. (closes issue #15871) Reported by: Ivan
	  Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
	  (license 229) ........

2010-07-01 14:37 +0000 [r273270-273352]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c: Fixed whitespace problems

	* main/manager.c: Altered my comment about TCP_NODELAY

	* addons/chan_mobile.c: Don't free written frames in chan_mobile's
	  mbl_write() function. (closes issue #16430) Reported by: azbest
	  Tested by: azbest

	* main/manager.c: Set TCP_NODELAY on manager TCP sockets to prevent
	  delays on outgoing packets. This regression was introduced in
	  r48338. AST-359

2010-06-30 17:28 +0000 [r273233]  Paul Belanger <paul.belanger@polybeacon.com>

	* res/res_rtp_asterisk.c: Fix rt(c)p set debug ip taking wrong
	  argument Also clean up some coding errors. (closes issue #17469)
	  Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch
	  uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger

2010-06-30 17:17 +0000 [r273197-273198]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/config.h: Remove unnecessary if test in
	  CV_DSTR()

	* include/asterisk/config.h: Misc doxygen cleanup in config.h

2010-06-30 01:07 +0000 [r273054-273144]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Permission checking for the system application is
	  backwards. (closes issue #17550) Reported by: kenner Patches:
	  manager.c.diff uploaded by kenner (license 1040) Tested by:
	  kenner

	* main/config.c: Don't attempt to proceed if our internal parser
	  indicates an invalid file. (closes issue #17560) Reported by:
	  Nick_Lewis

	* /, channels/chan_sip.c: Merged revisions 273060 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
	  | 10 lines Allow the "useragent" value to be restored into memory
	  from the realtime backend. This value is purely informational. It
	  does not alter configuration at all. (closes issue #16029)
	  Reported by: Guggemand Patches: realtime-useragent.patch uploaded
	  by Guggemand (license 897) Tested by: Guggemand ........

	* /: Recorded merge of revisions 273057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
	  | 4 lines _Really_ skip the channel... don't just retry for
	  another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Exclude libical for insufficient versions.

	* main/pbx.c: Send DialPlanComplete as a response, not as a
	  separate event. Otherwise, it goes to all manager sessions and
	  may exclude the current session, if the Events mask excludes it.
	  (closes issue #17504) Reported by: rrb3942 Patches:
	  showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
	  by: rrb3942

2010-06-29 20:44 +0000 [r272981]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: send a 400 Bad Request on malformed sip
	  request RFC 2361 section 24.4.1 send a 400 Bad Request if the
	  request can not be understood due to malformed syntax. Currently
	  we simply ignore a packet with a missing callid, to, from, or via
	  header. Instead of ignoring we now send the 400 Bad request.

2010-06-28 21:50 +0000 [r272923-272926]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 272925 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
	  | 8 lines Don't change ownership/group/permissions on run
	  directory, if it already exists. (closes issue #17076) Reported
	  by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
	  tilghman (license 14) Tested by: stuarth ........

	* /, main/config.c: Merged revisions 272921-272922 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28
	  Jun 2010) | 8 lines Change the way that we read include files, to
	  accommodate for changes in GCC 4.4. (closes issue #17472)
	  Reported by: seandarcy Patches: config2.patch uploaded by nivan
	  (license 1066) Tested by: nivan ........ r272922 | tilghman |
	  2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
	  trailing blanks on #includes ........

2010-06-28 18:38 +0000 [r272880]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c,
	  channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h: rfc compliant sip option
	  parsing + new unit test RFC 3261 section 8.2.2.3 states that if
	  any unsupported options are found in the Require header field, a
	  "420 (Bad Extension)" response should be sent with an Unsupported
	  header field containing only the unsupported options. This is not
	  currently being done correctly. Right now, if Asterisk detects
	  any unsupported sip options in a Require header the entire list
	  of options are returned in the Unsupported header even if some of
	  those options are in fact supported. This patch fixes that by
	  building an unsupported options character buffer when parsing the
	  options that can be sent with the 420 response. A unit test
	  verifying this functionality has been created. Some code
	  refactoring was required. Review:
	  https://reviewboard.asterisk.org/r/680/

2010-06-28 17:33 +0000 [r272805]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 272804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
	  2010) | 5 lines Decode URI in contact header of 302 response.
	  ABE-2352 ........

2010-06-28 15:33 +0000 [r272684]  Russell Bryant <russell@digium.com>

	* doc/tex/chan-mobile.tex (added), doc/tex/celdriver.tex,
	  doc/tex/chan_mobile.tex (removed), doc/tex/cdrdriver.tex,
	  doc/tex/asterisk.tex, doc/tex/cel-doc.tex: Use the underscore
	  package so that underscores do not need to be escaped.

2010-06-28 14:55 +0000 [r272652]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: code guidelines cleanup for retrans_pkt()
	  function I am doing work in this function. I noticed a large
	  number of coding guidline fixes that needed to be made. Rather
	  than have those changes distract from my functional changes I
	  decided to separate these into a separate patch.

2010-06-25 20:18 +0000 [r272568]  Tilghman Lesher <tlesher@digium.com>

	* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272562 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
	  | 5 lines Make the structure of the table specified before match
	  the queries and results. (closes issue #17557) Reported by: cmaj
	  ........

2010-06-25 19:42 +0000 [r272558]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c, include/asterisk/res_fax.h: Implemement support
	  for handling multiple documents when sending.

2010-06-25 19:39 +0000 [r272557]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: chan_sip: more accurate retransmissions
	  RFC3261 states that Timer A should start at 500ms (T1) by
	  default. In chan_sip this value initially started at 1000ms and I
	  changed it to 500ms recently. After doing that I noticed in my
	  packet captures that it still occasionally retransmitted starting
	  at 1000ms instead of 500ms like I told it to. This occurs because
	  the scheduler runs in the do_monitor thread. If a new
	  retransmission is added while the do_monitor thread is sleeping
	  then it may not detect that retransmission for nearly 1000ms. To
	  fix this I just poke the do_monitor thread to wake up when a new
	  packet is sent reliably requiring retransmits. The thread then
	  detects the new scheduler entry and adjusts its sleep time to
	  account for it. Review: https://reviewboard.asterisk.org/r/747

2010-06-25 19:17 +0000 [r272533]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile: Symlink sounds files, to save disk space, when
	  multiple tarballs/checkouts are on the same system.

2010-06-24 22:11 +0000 [r272447]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 272446 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
	  | 10 lines ss_thread calls pri_grab without lock during overlap
	  dial Recent changes to chan_dahdi with relation to overlap
	  dialing call pri_grab without first obtaining a lock. (closes
	  issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
	  by jpeeler (license 325) ........

2010-06-23 23:09 +0000 [r272370]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Resolve some errors produced during module
	  unload of chan_iax2. The external test suite stops Asterisk using
	  the "core stop gracefully" command. The logs from the tests show
	  that there are a number of problems with Asterisk trying to
	  cleanly shut down. This patch addresses the following type of
	  error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]:
	  lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371
	  (iax2_process_thread_cleanup): Error destroying mutex
	  &thread->lock: Device or resource busy For an example in the
	  context of a build, see:
	  http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
	  purpose of this patch is to change the thread pool shutdown
	  procedure to be more explicit to ensure that the thread exits
	  from a point where it is not holding a lock. While testing that,
	  I encountered various crashes due to the order of operations in
	  unload_module() being problematic. I reordered some things there,
	  as well. Review: https://reviewboard.asterisk.org/r/736/

2010-06-23 22:36 +0000 [r272368]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 272367 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
	  of the patch only adds AgentComplete for attended transfers. It
	  was already present for blind transfers. ........ r272367 |
	  mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
	  lines Send AgentComplete manager events in the event of blind and
	  attended transfers. (closes issue #16819) Reported by: elbriga
	  Patches: app_queue.diff uploaded by elbriga (license 482)
	  ........

2010-06-23 21:53 +0000 [r272260-272332]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: If there is realtime configuration, it
	  does not get re-read on reload unless the config file also
	  changes. (closes issue #16982) Reported by: dmitri Patches:
	  res_musiconhold.patch uploaded by dmitri (license 1001) Tested
	  by: atis

	* res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c,
	  res/ael/ael.flex: Ensure a NULL file while debugging cannot crash
	  AEL. (closes issue #17215) Reported by: vazir Patches:
	  20100518__issue17215.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman

2010-06-23 21:06 +0000 [r272257-272259]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_meetme.c: Fix previous merge. ast_test_flag !=
	  ast_test_flag64

	* /, apps/app_meetme.c: Merged revisions 272255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
	  2010) | 12 lines First caller into a dynamic conference now enter
	  pin once. If MeetMe is configured to use dynamic conference
	  numbers, then the first caller (which creates the conference) had
	  to enter the PIN number twice. (closes issue #15878) Reported by:
	  shawkris Patches: issue15878.patch uploaded by pabelanger
	  (license 224) Tested by: pabelanger ........

2010-06-23 20:59 +0000 [r272254-272256]  Terry Wilson <twilson@digium.com>

	* configure, include/asterisk/autoconfig.h.in: Update configure
	  when changing autconf m4 files...

	* autoconf/ast_ext_tool_check.m4: Honor the --with-${library}=path
	  for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by:
	  pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson
	  (license 396) Tested by: twilson Review:
	  https://reviewboard.asterisk.org/r/739/

2010-06-23 20:35 +0000 [r272243-272252]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/manager.c: Correct manager variable 'EventList' case.
	  (closes issue #17520) Reported by: kobaz Patches: manager.patch
	  uploaded by kobaz (license 834) Tested by: lmadsen

	* configs/say.conf.sample: Add localization support for Spanish
	  (closes issue #17548) Reported by: cjacobsen Patches:
	  say.conf.sample.diff uploaded by cjacobsen (license 1029)

2010-06-23 19:59 +0000 [r272218]  Tim Ringenbach <tim.ringenbach@gmail.com>

	* channels/chan_local.c: Add new AMI command LocalOptimizeAway.
	  This command lets you request a "/n" local channel optimize
	  itself out of the way anyway. Review:
	  https://reviewboard.asterisk.org/r/732/

2010-06-23 18:45 +0000 [r272148-272150]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c: D'oh! Defaultenabled FTL.

	* /: Recorded merge of revisions 272147 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010)
	  | 5 lines Backport part of revision 136715 to fix callerid in
	  voicemail text files (IMAP only). (closes issue #16945) Reported
	  by: mneuhauser ........

2010-06-23 18:39 +0000 [r272146]  Terry Wilson <twilson@digium.com>

	* apps/app_meetme.c: Don't start the sla thread unless we realy
	  need it

2010-06-23 18:25 +0000 [r272145]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c: Load all lines from realtime, not just the
	  first one. (closes issue #17144) Reported by: nahuelgreco
	  Patches: 20100513__issue17144__trunk.diff.txt uploaded by
	  tilghman (license 14) Tested by: tilghman

2010-06-23 17:21 +0000 [r272109]  Terry Wilson <twilson@digium.com>

	* apps/app_meetme.c: Make sure reload updates SLA config Even if
	  there are no stations or trunks defined, we need to start the sla
	  thread to make sure we get the reload event. Also, when doing a
	  reload we need to remove the existing trunks and stations or they
	  end up hanging around. (closes issue #16818) Reported by: mbonin
	  Patches: sla_reload.patch uploaded by twilson (license 396)
	  Tested by: twilson

2010-06-23 17:08 +0000 [r272090]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add extra protection for reinvite glare
	  scenario. Testing proved that if Asterisk sent a connected line
	  reinvite, and the endpoint to which the reinvite were being sent
	  sent a reinvite, Asterisk would not properly respond with a 491
	  response. The reason is that on connected line reinvites, we set
	  the dialog's invitestate to INV_CALLING to prevent Asterisk from
	  sending a rapid flurry of connected line reinvites. For other
	  reinvites we do not do this. Because of the current invitestate,
	  when Asterisk received the reinvite, we interpreted this as a
	  spiraled INVITE, and thus did not behave properly. The fix for
	  this is to not enter the loop detection or spiral logic in
	  handle_request_invite if the channel state is currently up. This
	  way, no mid-call reinvites will be misinterpreted, no matter what
	  the nature of the reinvite may have been.

2010-06-22 23:20 +0000 [r272052]  Russell Bryant <russell@digium.com>

	* channels/chan_dahdi.c: Don't try to lock/unlock an uninitialized
	  lock on a dahdi_pri. This small changes prevents
	  destroy_all_channels() from accessing a lock on an unused
	  dahdi_pri struct, resolving a ton of ERRORs that get spewed out
	  when shutting Asterisk down gracefully.

2010-06-22 22:11 +0000 [r271905-272014]  David Vossel <dvossel@digium.com>

	* pbx/pbx_config.c: fixes issue with 'dialplan remove extension
	  blah' segfaulting with tab completion (closes issue #17440)
	  Reported by: kobaz

	* channels/chan_sip.c: ignore CANCEL request after having already
	  received final response to INVITE RFC 3261 section 9 states that
	  a CANCEL has no effect on a request to a UAS that has already
	  given a final response. This patch checks to make sure there is a
	  pending invite before allowing a CANCEL request to be processed,
	  otherwise it responds to the CANCEL with a "481 Call/Transaction
	  Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/

	* main/manager.c: minor fixes for white/black event filters This
	  fixes a ref count leak in event filters and checks for a filter
	  container allocation failure during session creation.

2010-06-22 17:35 +0000 [r271903]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 271902 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
	  2010) | 8 lines Decrease the module ref count in sip_hangup when
	  SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
	  ref count correct. (closes issue #16815) Reported by: rain
	  Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
	  (modified) Tested by: rain ........

2010-06-22 16:29 +0000 [r271868]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, configs/manager.conf.sample, CHANGES: Add regular
	  expression filtering for manager events. This patch as documented
	  in the sample config allows one to optionally apply white, black,
	  or both types of filtering to manager events. The new
	  'eventfilter' option is set per user. (closes issue #14861)
	  Reported by: fnordian Patches: eventfilter3.patch uploaded by
	  fnordian (license 110), modified by me Review:
	  https://reviewboard.asterisk.org/r/673/

2010-06-22 16:28 +0000 [r271833-271867]  Russell Bryant <russell@digium.com>

	* res/ais/clm.c, res/ais/evt.c: Resolve some errors that occur on a
	  graceful shutdown. Don't Finalize() if Initialize() did not
	  succeed. This resulted in an error about trying to Finalize() an
	  invalid handle. Also trim some trailing whitespace while in the
	  area.

	* res/res_fax.c: Change the method of retrieving the Asterisk
	  version string. Using this method makes it so res_fax doesn't
	  have to be rebuilt on every svn update.

2010-06-22 15:46 +0000 [r271831]  David Vossel <dvossel@digium.com>

	* main/features.c: fixes attended transfer behavior when both
	  transferee and transferer hung up If both the transferer and
	  transferee of a attended transfer hangup before the new channel
	  picks up, the new channel should be hung up as well as it has no
	  endpoint to talk to. This mirrors the expected behavior used in
	  1.4. (closes issue #17444) Reported by: corruptor

2010-06-22 15:08 +0000 [r271690-271764]  Matthew Nicholson <mnicholson@digium.com>

	* CHANGES: Updated the CHANGES file documenting the addition of a
	  configurable port in the dundi config file.

	* configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
	  271761 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
	  2010) | 9 lines Allow users to specify a port for dundi peers.
	  (closes issue #17056) Reported by: klaus3000 Patches:
	  dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
	  Tested by: klaus3000 ........

	* /, channels/chan_sip.c, include/asterisk/strings.h,
	  channels/sip/include/sip.h: Merged revisions 271689 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue,
	  22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to
	  automatically calculate the Content-Length. This is done by
	  storing packet content in a buffer until it is actually time to
	  send the packet, at which time the size of the packet is
	  calculated. This change was made to ensure that the
	  Content-Length is always correct. (closes issue #17326) Reported
	  by: kenner Tested by: mnicholson, kenner Review:
	  https://reviewboard.asterisk.org/r/693/ ........ This change also
	  adds an ast_str_copy_string() function (similar to
	  ast_copy_string), that copies one ast_str into another, properly
	  handling embedded nulls.

2010-06-21 22:41 +0000 [r271657]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/menuselect-deps.in, configure, configure.ac,
	  res/res_timing_kqueue.c: Conflict kqueue on OS X, since it
	  doesn't work there yet, anyway.

2010-06-21 21:58 +0000 [r271625]  David Vossel <dvossel@digium.com>

	* codecs/codec_speex.c, codecs/ex_speex.h,
	  contrib/editors/asterisk.vim: add speex 16khz sample frame so
	  codec cost can be calculated (closes issue #17534) Reported by:
	  fabled Patches: speex-wb-sample.diff uploaded by fabled (license
	  448)

2010-06-21 20:46 +0000 [r271554]  Jeff Peeler <jpeeler@digium.com>

	* res/ael/pval.c, /: Merged revisions 271552 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
	  | 7 lines Do not use sizeof to calculate size of a heap allocated
	  character array. Change left out from 271399. (closes issue
	  #16053) Reported by: diLLec ........

2010-06-21 20:46 +0000 [r271551-271553]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c: fixes crash
	  when From header URI is missing "sip:" (closes issue #17437)
	  Reported by: klaus3000 Patches: sip_crash uploaded by dvossel
	  (license 671) Tested by: klaus3000

	* res/res_rtp_asterisk.c: fixes logic error introduced by slin16
	  sip support

2010-06-21 05:10 +0000 [r271520]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_saycounted.c (added), CHANGES: Add new application for
	  declining counting words in multiple languages. (closes issue
	  #16869) Reported by: chappell Patches: app_say_counted-20100317.c
	  uploaded by chappell (license 8) Tested by: chappell

2010-06-18 21:32 +0000 [r271483]  Jeff Peeler <jpeeler@digium.com>

	* res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
	  revisions 271399 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
	  | 11 lines Fix crash when parsing some heavily nested statements
	  in AEL on reload. Due to the recursion used when compiling AEL in
	  gen_prios, all the stack space was being consumed when parsing
	  some AEL that contained nesting 13 levels deep. Changing a few
	  large buffers to be heap allocated fixed the crash, although I
	  did not test how many more levels can now be safely used. (closes
	  issue #16053) Reported by: diLLec Tested by: jpeeler ........

2010-06-18 18:59 +0000 [r271341]  David Vossel <dvossel@digium.com>

	* main/file.c: file.c was truncating audio file formats to the
	  lower 32bits.

2010-06-18 18:36 +0000 [r271336]  Jeff Peeler <jpeeler@digium.com>

	* /: Recorded merge of revisions 271335 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010)
	  | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This
	  is a backport of 269307, committed to trunk by rmudgett.) Calling
	  dahdi_indicate() when the channel private lock is already held
	  can cause a deadlock if the PRI lock is needed because
	  dahdi_indicate() will also get the channel private lock. The
	  pri_grab() function assumes that the channel private lock is held
	  once to avoid deadlock. (closes issue #17261) Reported by: aragon
	  ........

2010-06-17 21:23 +0000 [r271231-271300]  David Vossel <dvossel@digium.com>

	* channels/sip/reqresp_parser.c: fixes some coding guideline issue

	* channels/sip/include/dialog.h, channels/chan_sip.c,
	  channels/sip/include/sip.h: retransmit response to BYE requests
	  until timer J expires According to RFC 3261 section 17.2.2, which
	  describes non-INVITE server transaction, when a dialog enters the
	  Completed state it must destroy the dialog after Timer J (T1*64)
	  fires. For a BYE transaction Asterisk terminates the dialog
	  immediately during sip_hangup() when it should be waiting T1*64
	  ms. This results in some odd behavior. For instance if Asterisk
	  receives a BYE and transmits a 200ok in response, if the endpoint
	  never receives the 200ok it will retransmit the BYE to which
	  Asterisk responds with a "481 Call leg/transaction does not
	  exist" because the dialog is already gone. To resolve this I made
	  a function called sip_scheddestroy_final(). This differs slightly
	  from sip_schedestroy() in that it enables a flag that will
	  prevent the destruction from ever being rescheduled or canceled
	  afterwards. It also prevents the pvt's needdestroy flag from
	  being set which triggers the destruction of the dialog within the
	  do_monitor thread(). By using this function we are guaranteed
	  destruction will not occur until the scheduled time. This allows
	  Asterisk to respond to any possible retransmits for a dialog
	  after we process the initial BYE request for T1*64 ms. Other
	  changes: I removed two instances where sip_cancel_destroy is used
	  right before calling sip_scheddestroy. sip_scheddestroy always
	  calls sip_cancel_destroy before scheduling the new destruction so
	  it is completely unnecessary. Review:
	  https://reviewboard.asterisk.org/r/694/

	* res/res_rtp_asterisk.c, main/rtp_engine.c, CHANGES: adds support
	  for slin16 in sip (closes issue #16153) Reported by: kfister
	  Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license
	  912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested
	  by: kfister, malcolmd

	* main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
	  main/rtp_engine.c, codecs/codec_speex.c, CHANGES,
	  include/asterisk/frame.h: adds speex 16khz audio support (closes
	  issue #17501) Reported by: fabled Patches:
	  asterisk-trunk-speex-wideband-v2.patch uploaded by fabled
	  (license 448) Tested by: malcolmd, fabled, dvossel

2010-06-17 15:34 +0000 [r271192]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_analog.c: Change expected operation from error to
	  debug message

2010-06-17 00:30 +0000 [r271089]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_meetme.c: option w[(secs)] incorrectly capitalized in
	  xmldoc (closes issue #17516) Reported by: karlfife

2010-06-16 22:37 +0000 [r271056]  David Vossel <dvossel@digium.com>

	* channels/sip/reqresp_parser.c: addition of more parse_uri test
	  cases

2010-06-16 21:17 +0000 [r270987]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, configs/extensions.conf.sample: Merged revisions 270979 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun
	  2010) | 4 lines Fixed typo in macro-page Reported to
	  #asterisk-dev by a student of jsmith. ........

2010-06-16 21:12 +0000 [r270981-270983]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c: Fix the actual place that was pointed out,
	  for previous commit.

	* /, channels/chan_agent.c: Merged revisions 270980 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun
	  2010) | 4 lines Need to lock the agent chan before access its
	  internal bits. Pointed out by russellb on asterisk-dev mailing
	  list. ........

2010-06-16 20:34 +0000 [r270974]  Matthew Nicholson <mnicholson@digium.com>

	* main/dnsmgr.c, main/acl.c: Set sin_family to AF_INET when doing
	  lookups, also reset sin_port the first time the ip address
	  changes. (closes issue #17496) Reported by: ManChicken (closes
	  issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch
	  uploaded by chappell (license 8) Tested by: DennisD, gentlec,
	  damage, wimpy

2010-06-16 19:03 +0000 [r270940]  David Vossel <dvossel@digium.com>

	* main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
	  main/rtp_engine.c, channels/chan_sip.c, CHANGES,
	  channels/chan_iax2.c, include/asterisk/frame.h,
	  formats/format_g719.c (added): addition of G.719 pass-through
	  support (closes issue #16293) Reported by: malcolmd Patches:
	  g719.passthrough.patch.7 uploaded by malcolmd (license 924)
	  format_g719.c uploaded by malcolmd (license 924)

2010-06-16 18:43 +0000 [r270936]  Paul Belanger <paul.belanger@polybeacon.com>

	* res/res_agi.c, CHANGES: MSG_OOB flag on HANGUP packet removed.
	  Per Tilghman's request on IRC (#asterisk-bugs). (closes issue
	  #17506) Reported by: brycebaril Tested by: pabelanger, tilghman

2010-06-16 17:36 +0000 [r270867]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 270866 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16
	  Jun 2010) | 22 lines fixes chan_iax2 race condition There is code
	  in chan_iax2.c that attempts to guarantee that only a single
	  active thread will handle a call number at a time. This code
	  works once the thread is added to an active_list of threads, but
	  we are not currently guaranteed that a newly activated thread
	  will enter the active_list immediately because it is left up to
	  the thread to add itself after frames have been queued to it.
	  This means that if two frames come in for the same call number at
	  the same time, it is possible for them to grab two separate
	  threads because the first thread did not add itself to the
	  active_list fast enough. This causes some pretty complex
	  problems. This patch resolves this race condition by immediately
	  adding an activated thread to the active_list within the network
	  thread and only depending on the thread to remove itself once it
	  is done processing the frames queued to it. By doing this we are
	  guaranteed that if another frame for the same call number comes
	  in at the same time, that this thread will immediately be found
	  in the active_list of threads. Review:
	  https://reviewboard.asterisk.org/r/720/ ........

2010-06-16 16:45 +0000 [r270836]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_analog.c: Fix no call waiting caller ID Clearing the
	  callwaitcas flag in analog_call was causing the incoming D digit
	  to be ignored which triggers sending the caller ID.

2010-06-16 15:05 +0000 [r270801]  Paul Belanger <paul.belanger@polybeacon.com>

	* doc/tex/channelvariables.tex: Update formatting for
	  channelvariables.tex (closes issue #17511) Reported by: klaus3000
	  Patches: channelvariables.tex-patch.txt uploaded by klaus3000
	  (license 65) Tested by: pabelanger

2010-06-15 22:48 +0000 [r270726]  Russell Bryant <russell@digium.com>

	* channels/sig_analog.c: Don't blow up if an ast_channel doesn't
	  get allocated.

2010-06-15 21:42 +0000 [r270658-270692]  Terry Wilson <twilson@digium.com>

	* main/http.c: Don't continue sending the file when there has been
	  an error If there is a problem with a firmware file, Polycom
	  phones will close the connection. We were continuing to send the
	  file anyway. There should be no reason to continue sending a file
	  if there is an error writing it. (closes issue #16682) Reported
	  by: lmadsen

	* res/res_phoneprov.c: Don't send files twice and remove extra \r\n
	  from header After the manager http auth changes, we forgot to
	  remove the manual sending of the file. Also, ast_http_send adds
	  two \r\n to the header that is passed to it, so a trailing \r\n
	  is removed from the Content-type header. It might be better to
	  change ast_http_send, but I don't like changing the behavior of
	  an API function. (closes issue #17239) Reported by: cjacobsen
	  Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested
	  by: lathama, cjacobsen

	* channels/chan_sip.c: Make contactdeny apply to src ip when
	  nat=yes chan_sip's "contactdeny" feature screens the "to be
	  registered contact". In case of nat=yes it should not use the
	  address information from the Contact header (which is not used at
	  all for routing), but the source IP address of the request. Thus,
	  if nat=yes and a client sends a request from a denied IP address
	  (e.g. by spoofing the src-IP address) it can bypass the
	  screening. This commit makes contactdeny apply to the src ip when
	  nat=yes instead. (closes issue #17276) Reported by: klaus3000
	  Patches: patch-asterisk-trunk-contactdeny.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000

2010-06-15 18:26 +0000 [r270519-270584]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 270583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010)
	  | 5 lines Variables have always been case-sensitive, so we should
	  not be removing case-insensitive matches. Bug reported via the
	  -dev list. See
	  http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
	  ........

	* res/res_jabber.c: Argh, mixed declarations and code.

	* configs/jabber.conf.sample, include/asterisk/jabber.h,
	  doc/distributed_devstate-XMPP.txt (added), CHANGES,
	  res/res_jabber.c: Add distributed devicestate via the XMPP
	  protocol. (closes issue #15757) Reported by: Marquis Patches:
	  distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
	  Tested by: Marquis, lmadsen, marcelloceschia Review:
	  https://reviewboard.asterisk.org/r/351/

2010-06-15 12:51 +0000 [r270443]  Leif Madsen <lmadsen@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 270442 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010)
	  | 1 line Move information about zonemessages into the
	  [zonemessages] section. ........

2010-06-14 21:33 +0000 [r270332]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, res/res_musiconhold.c: Merged revisions 270331 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon,
	  14 Jun 2010) | 14 lines Properly play first file in sort list.
	  When using sort=alpha we would always skip the first file in the
	  list first time through. We now check for that properly. (closes
	  issue #17470) Reported by: pabelanger Patches: sort.aplha.patch
	  uploaded by pabelanger (license 224) Tested by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/703/ ........

2010-06-14 20:51 +0000 [r270298]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
	  Extract sig_ss7_init_linkset() to sig_ss7. Also found a place
	  where sig_pri_init_pri() was inlined and called it instead.

2010-06-14 19:41 +0000 [r270260]  Jason Parker <jparker@digium.com>

	* channels/chan_agent.c: Add option to get untruncated channel name
	  from AGENT function. The "channel" option would chop the channel
	  name at the last '-', which made it useless for something like a
	  channel transfer from the dialplan. The "fullchannel" option will
	  return the channel name as-is. ABE-2218

2010-06-14 15:55 +0000 [r270219]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add digit
	  manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
	  Add the append_msn_to_cid_tag option to chan_dahdi like
	  chan_misdn. Review: https://reviewboard.asterisk.org/r/696/

2010-06-13 09:16 +0000 [r270184]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* autoconf/ast_check_pwlib.m4, configure: bashism in configure
	  script Theoretically the ./configure script is a pure
	  bourne-shell script. Practically it may be run by bash if /bin/sh
	  is not good enough. But we should not count on it. See bug report
	  for the gory details. (closes issue #17485) Patches:
	  0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by
	  tzafrir (license 46)

2010-06-13 01:53 +0000 [r270042-270151]  Paul Belanger <paul.belanger@polybeacon.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Reverting patch and reopening issue #16155, as patch breaks
	  FreeBSD / OSX builds.

	* /, doc/HOWTO_collect_debug_information.txt: Merged revisions
	  270078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun
	  2010) | 2 lines Fix typo in example ........

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Use
	  pkg-config to find gmime libraries This way the libraries can be
	  found even if they are in non-standard locations. (closes issue
	  #16155) Reported by: jcollie Patches:
	  0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch
	  uploaded by jcollie (license 412) Tested by: jsmith, tilghman,
	  pabelanger

2010-06-11 18:31 +0000 [r269936-269976]  Tilghman Lesher <tlesher@digium.com>

	* main/frame.c, /: Merged revisions 269960 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010)
	  | 8 lines For SpeeX, 0 bits remaining is valid and does not need
	  an emitted warning. (closes issue #15762) Reported by: nblasgen
	  Patches: issue15672.patch uploaded by pabelanger (license 224)
	  Tested by: nblasgen ........

	* CHANGES, main/db.c: Add DBGetComplete event after a
	  DBGetResponse. (closes issue #16965) Reported by: rrb3942
	  Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003)

	* main/logger.c: Remove lines from the output related to the
	  backtrace itself.

2010-06-10 20:30 +0000 [r269889]  Paul Belanger <paul.belanger@polybeacon.com>

	* Makefile, makeopts.in: Remove ASTBINDIR variable (closes issue
	  #17031) Reported by: pabelanger Patches:
	  Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
	  Tested by: pabelanger, tilghman

2010-06-10 19:34 +0000 [r269749-269822]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 269821 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun
	  2010) | 19 lines Fix potential crash when writing raw SLIN audio
	  on a PLC-enabled channel. The issue here was that the frame
	  created when adjusting for PLC had no offset to its audio data.
	  If this frame were translated to another format prior to being
	  sent out an RTP socket, all went well because the translation
	  code would put an appropriate offset into the frame. However, if
	  the SLIN audio were not translated before being sent out the RTP
	  socket, bad things would happen. Specifically, the
	  ast_rtp_raw_write makes the assumption that the frame has at
	  least enough of an offset that it can accommodate an RTP header.
	  This was not the case. As such, data was being written prior to
	  the allocation, likely corrupting the data the memory allocator
	  had written. Thus when the time came to free the data, all hell
	  broke loose. ....Well, Asterisk crashed at least. The fix was
	  just what one would expect. Offset the data in the frame by a
	  reasonable amount. The method I used is a bit odd since the data
	  in the frame is 16 bit integers and not bytes. I left a big ol'
	  comment about it. This can be improved on if someone is
	  interested. I was more interested in getting the crash resolved.
	  ........

	* doc/tex/plc.tex (added), doc/tex/asterisk.tex: Add documentation
	  explaining PLC in Asterisk. Review:
	  https://reviewboard.asterisk.org/r/688/

2010-06-10 13:17 +0000 [r269711]  Russell Bryant <russell@digium.com>

	* tests/test_heap.c: Fix an off by one error that caused a unit
	  test to occasionally crash.

2010-06-10 12:28 +0000 [r269707]  Kevin P. Fleming <kpfleming@digium.com>

	* main/logger.c: Ensure that 'logger show channels' works properly
	  when wildcards are used in logger.conf.

2010-06-10 08:15 +0000 [r269636]  Tilghman Lesher <tlesher@digium.com>

	* /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
	  revisions 269635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010)
	  | 9 lines Ensure restartable system calls can restart (BSD signal
	  semantics). This eliminates the annoying <beep> on the console.
	  (closes issue #17477) Reported by: jvandal Patches:
	  20100610__issue17477.diff.txt uploaded by tilghman (license 14)
	  ........

2010-06-10 00:32 +0000 [r269417-269602]  Russell Bryant <russell@digium.com>

	* channels/chan_dahdi.c: Attempt to fix a FreeBSD build error by
	  including sys/stat.h.
	  http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log

	* main/lock.c: Attempt to fix FreeBSD build problem.

	* /, channels/chan_oss.c: Merged revisions 269495 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010)
	  | 2 lines Don't stop Asterisk if chan_oss fails to register
	  'Console' (due to another channel driver already claiming it).
	  ........

	* include/asterisk/event.h, main/event.c: Resolve an invalid memory
	  read on an event. Valgrind pointed out that attempting to get an
	  IE value from an event that has no IEs produces an invalid memory
	  read past the end of the event. Thanks to mmichelson for pointing
	  the problem out to me and then testing the fix.

2010-06-09 17:32 +0000 [r269346]  Paul Belanger <paul.belanger@polybeacon.com>

	* contrib/init.d/rc.debian.asterisk, /, main/term.c: Merged
	  revisions 269334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun
	  2010) | 12 lines Fix Debian init script to not use -c. When using
	  the init script as-is currently, it could cause issues on Debian
	  such as high CPU usage. This fix has worked for several people so
	  I'm implementing the change. We now handle color displays
	  properly. (closes issue #16784) Reported by: pabelanger Patches:
	  20100530__issue16784__2.diff.txt uploaded by tilghman (license
	  14) Tested by: pabelanger, tilghman ........

2010-06-09 17:06 +0000 [r269307-269308]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
	  Add missing API function to sig_ss7: sig_ss7_fixup().

	* channels/chan_dahdi.c: Eliminate deadlock potential in
	  dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup()
	  while the owner pointers are in a potentially inconsistent state
	  is a potentially bad thing in principle. However, calling
	  dahdi_indicate() when the channel private lock is already held
	  can cause a deadlock if the PRI lock is needed because
	  dahdi_indicate() will also get the channel private lock. The
	  pri_grab() function assumes that the channel private lock is held
	  once to avoid deadlock.

2010-06-09 15:09 +0000 [r269271]  David Vossel <dvossel@digium.com>

	* res/res_musiconhold.c: fixes crash in moh when cachertclasses
	  flag is used The result for moh_register was not verified to
	  guarantee the mohclass as added to the container. (closes issue
	  #16993) Reported by: dmitri Patches:
	  res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
	  moh_crash2.diff uploaded by dvossel (license 671) Tested by:
	  dmitri

2010-06-09 13:17 +0000 [r269238]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
	  dial by name in chan_dahdi * chan_dahdi supports dialing
	  configuring and dialing by device file name.
	  DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 .
	  Likewise it may appear in chan_dahdi.conf as 'channel =>
	  span-name!local!1'. * A new options for chan_dahdi.conf:
	  'ignore_failed_channels'. Boolean. False by default. If set,
	  chan_dahdi will ignore failed 'channel' entries. Handy for the
	  above name-based syntax as it does not depend on initialization
	  order. * have my_pri_make_cc_dialstring() only manupulate
	  dial-strings of group (gGrR) dialing, which make it lsightly more
	  complicated. https://reviewboard.asterisk.org/r/535/

2010-06-09 10:55 +0000 [r269187-269205]  Russell Bryant <russell@digium.com>

	* contrib/scripts/install_prereq: Add libjack-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add libpopt-dev, libical-dev, and
	  libspandsp-dev to install_prereq.

	* contrib/scripts/install_prereq: Add libnewt-dev to
	  install-prereq.

	* contrib/scripts/install_prereq: Add libopenais-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add an "install-unpackaged"
	  command to install_prereq for installing unpackaged dependencies
	  (such as NBS and libresample).

	* contrib/scripts/install_prereq: Add libcurl to install_prereq.

	* contrib/scripts/install_prereq: Add freetds-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add libradiusclient-ng-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add libbluetooth-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add libmysqlclient-dev to
	  install_prereq.

	* contrib/scripts/install_prereq: Add libgtk2.0-dev to the packages
	  list for install_prereq.

2010-06-08 23:48 +0000 [r269153]  Bradley Latus <brad.latus@gmail.com>

	* configs/cdr_custom.conf.sample, configs/cdr_tds.conf.sample,
	  cdr/cdr_sqlite.c, configs/cdr_sqlite3_custom.conf.sample,
	  funcs/func_cdr.c, configs/cdr_syslog.conf.sample, UPGRADE.txt,
	  cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, cdr/cdr_pgsql.c,
	  CHANGES, cdr/cdr_odbc.c, cdr/cdr_tds.c,
	  configs/cdr_odbc.conf.sample: Add High Resolution Times to CDRs
	  for Asterisk People expressed an interest in having access to the
	  exact length of calls to a finer degree than seconds. See the
	  CHANGES and UPGRADE.txt for usage also updated the sample configs
	  to note the change. Patch by snuffy. (closes issue #16559)
	  Reported by: cianmaher Tested by: cianmaher, snuffy Review:
	  https://reviewboard.asterisk.org/r/461/

2010-06-08 22:45 +0000 [r269119]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/localtime.h: Fix build on Mac OS X (and maybe
	  FreeBSD, too)

2010-06-08 18:50 +0000 [r269083]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_fax.c: Don't pass null to manager_event() (closes issue
	  #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff
	  uploaded by mnicholson (license 96) Tested by: bklang

2010-06-08 15:41 +0000 [r269008]  Russell Bryant <russell@digium.com>

	* Makefile.rules: Ensure CONFIG_FLAGS makes it into the build rules
	  when doing out of tree builds. (closes issue #16685) Reported by:
	  pprindeville

2010-06-08 15:39 +0000 [r269007]  Sean Bright <sean@malleable.com>

	* /, cdr/cdr_tds.c: Merged revisions 269006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun
	  2010) | 11 lines Reduce startup time for cdr_tds with large CDR
	  tables. Since we are just checking for table existence, add a
	  WHERE clause that will return no rows but will raise an error if
	  the table doesn't exist. (closes issue #17380) Reported by:
	  kkwong Patches: issue17380-01.patch uploaded by seanbright
	  (license 71) Tested by: kkwong ........

2010-06-08 15:23 +0000 [r268969-268988]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Update note in sip.conf.sample. Update
	  note in sip.conf.sample about externip and externhost with STUN.
	  (closes issue #16323) Reported by: klaus3000 Patches:
	  sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)

	* apps/app_meetme.c, main/ccss.c, include/asterisk/data.h,
	  res/res_jabber.c, res/res_config_sqlite.c,
	  include/asterisk/callerid.h, channels/chan_dahdi.c,
	  include/asterisk/bridging_technology.h,
	  include/asterisk/doxyref.h, include/asterisk/event.h,
	  include/asterisk/astmm.h, main/ast_expr2f.c, main/features.c,
	  include/asterisk/timing.h, include/asterisk/rtp_engine.h,
	  include/asterisk/ccss.h, include/asterisk/threadstorage.h,
	  include/asterisk/xml.h, main/pbx.c, channels/chan_sip.c,
	  include/asterisk/astobj2.h, include/asterisk/channel.h,
	  include/asterisk/calendar.h, include/asterisk/manager.h,
	  include/asterisk/features.h, include/asterisk/logger.h,
	  include/asterisk/http.h, channels/sig_pri.h,
	  include/asterisk/app.h, main/audiohook.c, include/asterisk/pbx.h,
	  include/asterisk/dnsmgr.h, include/asterisk/smdi.h,
	  apps/app_voicemail.c: Fix some doxygen warnings. (closes issue
	  #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded
	  by snuffy (license 35) Tested by: russell

2010-06-08 06:57 +0000 [r268896-268933]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c: Release list lock before returning on
	  error.

	* utils/extconf.c: Fix trunk build on Mac OS X.

2010-06-08 05:29 +0000 [r268894]  Terry Wilson <twilson@digium.com>

	* channels/sip/sdp_crypto.c (added), res/res_rtp_asterisk.c,
	  main/global_datastores.c, main/rtp_engine.c,
	  include/asterisk/res_srtp.h (added), channels/sip/srtp.c (added),
	  channels/chan_sip.c, include/asterisk/autoconfig.h.in,
	  res/res_srtp.exports.in (added), configure.ac, CHANGES,
	  channels/chan_iax2.c, res/res_srtp.c (added), main/channel.c,
	  build_tools/menuselect-deps.in, main/asterisk.exports.in,
	  configure, funcs/func_channel.c,
	  channels/sip/dialplan_functions.c,
	  channels/sip/include/sdp_crypto.h (added),
	  doc/tex/secure-calls.tex (added),
	  include/asterisk/global_datastores.h, channels/sip/include/srtp.h
	  (added), makeopts.in, include/asterisk/rtp_engine.h,
	  include/asterisk/frame.h, doc/tex/asterisk.tex,
	  channels/sip/include/sip.h: Add SRTP support for Asterisk After 5
	  years in mantis and over a year on reviewboard, SRTP support is
	  finally being comitted. This includes generic CHANNEL dialplan
	  functions that work for getting the status of whether a call has
	  secure media or signaling as defined by the underlying channel
	  technology and for setting whether or not a new channel being
	  bridged to a calling channel should have secure signaling or
	  media. See doc/tex/secure-calls.tex for examples. Original patch
	  by mikma, updated for trunk and revised by me. (closes issue
	  #5413) Reported by: mikma Tested by: twilson, notthematrix,
	  hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/

2010-06-08 00:45 +0000 [r268857]  Richard Mudgett <rmudgett@digium.com>

	* channels/sip/dialplan_functions.c: Make SIP tests compile again.

2010-06-07 22:56 +0000 [r268817-268818]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Use the mailbox destructor function,
	  instead.

	* channels/chan_sip.c, channels/sip/include/sip.h: Mailbox list
	  would previously grow at each reload, containing duplicates.
	  Also, optimize the allocation of mailboxes to avoid additional
	  memory structures. (closes issue #16320) Reported by: Marquis
	  Patches: 20100525__issue16320.diff.txt uploaded by tilghman
	  (license 14)

2010-06-07 20:04 +0000 [r268774]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h
	  (added), channels/Makefile, channels/sig_pri.c,
	  channels/sig_ss7.c (added): Extract sig_ss7 out of chan_dahdi.
	  Extract the SS7 specific code out of chan_dahdi like what was
	  done to ISDN/PRI and analog signaling. The new SS7 structures
	  were modeled on sig_pri. The changes to sig_pri are an
	  enhancement and a bug fix made possible because SS7 was
	  extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable
	  should have been set unconditionally in
	  sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability
	  interaction in dahdi_new() fixed because of SS7 extraction. 3)
	  Module ref count error in dahdi_new() if startpbx failed to start
	  the PBX for some reason. Review:
	  https://reviewboard.asterisk.org/r/661/

2010-06-07 19:52 +0000 [r268773]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp_engine.c, channels/chan_sip.c,
	  channels/sip/dialplan_functions.c, include/asterisk/rtp_engine.h:
	  Seems strange (and the code backs up) that if the max and min of
	  a statistic is expressed as a double, the last value would not
	  also need to be a double. (closes issue #15807) Reported by:
	  klaus3000

2010-06-07 19:06 +0000 [r268734]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Moved AOC request code out of the middle of
	  code parsing the dialed number.

2010-06-07 18:59 +0000 [r268731]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Event well was going dry. (issue #17234)

2010-06-07 17:34 +0000 [r268690]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/dsp.c: Set threshold for silence detection defaults to 256
	  (closes issue #15685) Reported by: david_s5 Patches:
	  dsp-silence-threshold-init.diff uploaded by dant (license 670)
	  issue15685.patch.v5 uploaded by pabelanger (license 224) Tested
	  by: danti Review: https://reviewboard.asterisk.org/r/670/

2010-06-07 17:14 +0000 [r268653]  Tilghman Lesher <tlesher@digium.com>

	* res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue
	  #17237) Reported by: pabelanger

2010-06-07 15:51 +0000 [r268578]  Richard Mudgett <rmudgett@digium.com>

	* main/file.c: Suppress warning in waitstream_core(). Suppress the
	  warning about unexpected control subclass frames for
	  AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and
	  AST_CONTROL_AOC in file.c:waitstream_core().

2010-06-06 05:29 +0000 [r268454-268534]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.redhat.asterisk: Take advantage of variable
	  substitution already in the Makefile to specify the correct
	  location for files in init.d. (closes issue #16979) Reported by:
	  jw-asterisk (issue #15691) Reported by: itamarjp

	* channels/chan_iax2.c: Finally track down and eliminate the
	  "FRACK! warnings from chan_iax2".

	* main/dsp.c: Fix crash in DTMF detection. What I did not
	  originally see in my previous commit was that even though the
	  next digit could be detected before the previous was considered
	  ended, the detection of the next digit effectively ends the
	  detection of the previous. Therefore, the length moves in
	  lockstep with the digit, and no separate counter is needed for
	  the length alone. (closes issue #17371) Reported by: alecdavis
	  (closes issue #17474) Reported by: kenner

	* main/manager.c: Verify event is not NULL before attempting to
	  lower its usecount. (closes issue #17234) Reported by: mav3rick

2010-06-05 05:23 +0000 [r268395-268417]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: Typo fix.

	* CHANGES: Grammatical error fix.

2010-06-05 02:51 +0000 [r268321]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 268320 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010)
	  | 3 lines Rest In Peace
	  http://www.outandaboutnewspaper.com/article/4061 ........

2010-06-04 22:37 +0000 [r268205-268281]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes compile error from uninitialized
	  variable

	* channels/chan_sip.c: RFC3261 compliant sip unreliable retransmit
	  timing + 'registerattempts' option tweak Changes. 1. RFC 3261
	  states in section 17.1.2.2 and 17.1.1.2 that retransmission
	  timers should initially be set to timer T1. T1 by default is
	  500ms. Asterisk was starting the retransmission timers at T1*2
	  which shouldn't cause any problems, but is not RFC compliant. 2.
	  RFC 3261 states in section 17.1.2.2 that for a non-INVITE client
	  transaction, if the retransmit timer fires while in the
	  proceeding state that the request must be retransmitted. Asterisk
	  currently ack's requests for both INVITE and non-INVITE
	  transactions when a 1XX response is received, this patch changes
	  this for non-INVITE requests. 3. The 'registerattempts' option in
	  sip.conf is supposed to set how many registry attempts will be
	  made before giving up. When this option is set to 1, I would
	  expect only one registry attempt to be made before stopping
	  because of a failure, but instead two are made. In my opinion
	  this is not expected behavior. This option does not indicate that
	  these are re-attempts. The logic behind this option has been
	  changed to only attempt registers the exact number of times this
	  option is set to. If this option is 0, it still continues to
	  re-attempt the registration forever. Review:
	  https://reviewboard.asterisk.org/r/687/

2010-06-04 20:42 +0000 [r267972-268127]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 268126 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04
	  Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on
	  cross-compiles. ........

	* Makefile, /, makeopts.in: Merged revisions 268050 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04
	  Jun 2010) | 6 lines Build menuselect with the build environment's
	  compiler, not the host (target)'s compiler. (closes issue #17464)
	  Reported by: pprindeville Tested by: tilghman ........

	* /, configure, configure.ac, autoconf/libcurl.m4: Merged revisions
	  267971 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010)
	  | 2 lines As-fixiate the build process ........

2010-06-04 14:45 +0000 [r267928]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Incoming overlap dialing no longer works
	  after sig_pri extraction. The problem would manifest itself if
	  your dialplan matching could accept more digits to match than
	  were actually dialed. The time out waiting for overlap digits
	  disconnected the call instead of matching any accumulated digits
	  to the dialplan. Accidental conversion of a break out of loop as
	  a break out of switch. (closes issue #17401) Reported by:
	  avalentin Patches: issue17401_digit_timeout.patch uploaded by
	  rmudgett (license 664) Tested by: avalentin, rmudgett

2010-06-04 03:20 +0000 [r267877]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/slin.h: As signed linear audio data is accessed
	  as 16-bit values, certain processors require the values to be
	  aligned in memory. (closes issue #16912) Reported by:
	  michaelevdokimov Patches: asterisk.patch uploaded by
	  michaelevdokimov (license 997) Tested by: michaelevdokimov

2010-06-04 03:11 +0000 [r267863]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Send an ACK for every final response
	  received for an INVITE From issue ABE-2247. RFC 3261 compliance
	  for sections 13.2.24 and 17.1.1.2. Review:
	  https://reviewboard.asterisk.org/r/692/

2010-06-04 02:58 +0000 [r267775-267862]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/slin.h: As signed linear audio data is accessed
	  as 16-bit values, certain processors require the values to be
	  aligned in memory. (closes issue #16912) Reported by:
	  michaelevdokimov

	* configure, autoconf/ast_ext_lib.m4: If there's a default, turn it
	  on, even when the option isn't specified.

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Merged revisions 267759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010)
	  | 7 lines Make the default install path appear to be /usr on
	  Linux, instead of /usr/local. Also, reorganize the options, so
	  that they're more alphabetical. (closes issue #17013) Reported
	  by: klaus3000 ........

2010-06-03 20:41 +0000 [r267714]  Russell Bryant <russell@digium.com>

	* main/ccss.c: Remove a LOG_WARNING. This came up when using the
	  sample configs, and just indicates expected behavior.

2010-06-03 19:46 +0000 [r267669]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Handle OOM errors more gracefully. (closes
	  issue #17084) Reported by: falves11 Patches:
	  issue17084_162_A.diff uploaded by falves11 (license 374) Tested
	  by: falves11

2010-06-03 18:53 +0000 [r267624]  Leif Madsen <lmadsen@digium.com>

	* UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGE for CDR
	  functionality changes. Updated the UPGRADE.txt and CHANGES file
	  stating that CDR records will not be explicity written unless
	  cdr.conf exists and is configured. (closes issue #17373) Reported
	  by: wdoekes Tested by: pabelanger

2010-06-03 18:38 +0000 [r267622]  Richard Mudgett <rmudgett@digium.com>

	* codecs/codec_dahdi.c: Make compile again.

2010-06-03 17:31 +0000 [r267537]  Russell Bryant <russell@digium.com>

	* channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio
	  isn't configured.

2010-06-03 17:09 +0000 [r267492]  Mark Michelson <mmichelson@digium.com>

	* codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
	  codecs/codec_alaw.c, main/translate.c, codecs/codec_g726.c,
	  codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
	  include/asterisk/translate.h: Remove unnecessary code relating to
	  PLC. The logic for handling generic PLC is now handled in
	  ast_write in channel.c instead of in translation code. Review:
	  https://reviewboard.asterisk.org/r/683/

2010-06-03 17:05 +0000 [r267445-267490]  Russell Bryant <russell@digium.com>

	* channels/chan_usbradio.c: Remove a line that was killing Asterisk
	  on startup.

	* channels/h323/Makefile.in: Comment out a rule that likes to run
	  implicitly unnecessarily, breaking builds

2010-06-03 00:02 +0000 [r267399]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
	  channels/sig_pri.c: Add ETSI Message Waiting Indication (MWI)
	  support. Add the ability to report waiting messages to ISDN
	  endpoints (phones). Relevant specification: EN 300 650 and EN 300
	  745 Review: https://reviewboard.asterisk.org/r/599/

2010-06-02 22:46 +0000 [r267352]  Russell Bryant <russell@digium.com>

	* channels/Makefile, channels/h323/Makefile.in: try to fix some
	  random chan_h323 compilation failures After some debugging, the
	  random chan_h323 build failures appear to be due to complications
	  introduced by some chan_h323 specific build stuff getting
	  triggered during a clean. Simplify this by moving the h323 clean
	  commands down into channels/makefile.

2010-06-02 22:28 +0000 [r267350]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, include/asterisk/channel.h, CHANGES,
	  channels/sig_pri.c: Add ETSI Malicious Call ID support. Add the
	  ability to report malicious callers as an AMI event in the call
	  event class. Relevant specification: EN 300 180 Review:
	  https://reviewboard.asterisk.org/r/576/

2010-06-02 21:44 +0000 [r267303-267305]  Russell Bryant <russell@digium.com>

	* utils/extconf.c: Fix a build error on mac.

	* main/Makefile: Ensure the -Wno-strict-aliasing flag makes it,
	  even if ASTCFLAGS has been specified. When ASTCFLAGS was
	  specified with the make command, Makefile.rules was using the
	  specified value from the command line and not the one here,
	  making it so this flag would go missing.

2010-06-02 21:05 +0000 [r267261]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
	  channels/sig_pri.c: Add ETSI Call Waiting support. Add the
	  ability to announce a call to an endpoint when there are no B
	  channels available. A call waiting call is a SETUP message with
	  no B channel selected. Relevant specification: EN 300 056, EN 300
	  057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan
	  function now supports the "no_media_path" option. * Returns "0"
	  if there is a B channel associated with the call. * Returns "1"
	  if no B channel is associated with the call. The call is either
	  on hold or is a call waiting call. If you are going to allow
	  incoming call waiting calls then you need to use
	  CHANNEL(no_media_path) do determine if you must drop a call to
	  accept the new call. Review:
	  https://reviewboard.asterisk.org/r/568/

2010-06-02 19:33 +0000 [r267181]  David Vossel <dvossel@digium.com>

	* CHANGES, doc/advice_of_charge.txt: Update CHANGES and aoc help
	  doc to reflect AOC additions

2010-06-02 18:53 +0000 [r267138]  Russell Bryant <russell@digium.com>

	* main/cli.c: Add a CLI command that blocks until Asterisk has
	  fully booted. Review: https://reviewboard.asterisk.org/r/684/

2010-06-02 18:13 +0000 [r267097]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Prevent use of uninitialized values. Two
	  struct sockaddr_ins are created when applying directmedia host
	  access rules. The addresses of these are passed to the RTP engine
	  to be filled in. However, the RTP engine inspects the fields of
	  the structs before actually taking action. This inspection caused
	  valgrind to be a bit unhappy.

2010-06-02 18:10 +0000 [r267096]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, configs/chan_dahdi.conf.sample,
	  include/asterisk/aoc.h (added), channels/chan_sip.c,
	  configs/manager.conf.sample, main/aoc.c (added),
	  apps/app_queue.c, channels/sig_pri.c, doc/advice_of_charge.txt
	  (added), main/channel.c, channels/sig_pri.h,
	  channels/chan_dahdi.c, main/manager.c, main/features.c,
	  tests/test_aoc.c (added), configs/sip.conf.sample,
	  include/asterisk/frame.h, main/asterisk.c,
	  channels/sip/include/sip.h: Generic Advice of Charge. Asterisk
	  Generic AOC Representation - Generic AOC encode/decode routines.
	  (Generic AOC must be encoded to be passed on the wire in the
	  AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent
	  generic encoded AOC data - Manager events for AOC-S, AOC-D, and
	  AOC-E messages Asterisk App Support - app_dial AOC-S pass-through
	  support on call setup - app_queue AOC-S pass-through support on
	  call setup AOC Unit Tests - AOC Unit Tests for encode/decode
	  routines - AOC Unit Test for manager event representation. SIP
	  AOC Support - Pass-through of generic AOC-D and AOC-E messages to
	  snom phones via the snom AOC specification. - Creation of
	  chan_sip page3 flags for the addition of the new
	  'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively
	  supports AOC pass-through through the use of the new
	  AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC
	  Pass-through support - 'aoc_enable' chan_dahdi.conf option for
	  independently enabling pass-through of AOC-S, AOC-D, AOC-E. -
	  'aoce_delayhangup' option for retrieving AOC-E on disconnect. -
	  DAHDI A() dial string option for requesting AOC services. example
	  usage: ;requests AOC-S, AOC-D, and AOC-E on call setup
	  exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review:
	  https://reviewboard.asterisk.org/r/552/

2010-06-02 17:57 +0000 [r267093]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Silence a compiler warning.

2010-06-02 17:29 +0000 [r267065]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/slin.h: Fix infinite loop when loading codec
	  speex This changes the sample slinear frame data to contain
	  non-zero data so that translation calculations for speex works
	  when preprocessing and VAD is turned on. The encoder expects
	  samples to be returned, but when attempted with the mentioned two
	  options and silent sample frames everything was discarded.
	  (closes issue #17240) Reported by: seandarcy Review:
	  https://reviewboard.asterisk.org/r/682/

2010-06-02 17:25 +0000 [r267041]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, main/ast_expr2.y: Merged revisions 267009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun
	  2010) | 7 lines Cleanup error/warning messages in AEL2 parser
	  (closes issue #16684) Reported by: Silmaril Patches:
	  patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
	  ........

2010-06-02 17:13 +0000 [r266926-267008]  Richard Mudgett <rmudgett@digium.com>

	* main/manager.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, configs/manager.conf.sample, CHANGES,
	  channels/sig_pri.c, include/asterisk/manager.h: Add ETSI Advice
	  Of Charge (AOC) event reporting. This feature generates AMI
	  events in the new aoc event class from the events passed up by
	  libpri. Review: https://reviewboard.asterisk.org/r/537/

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
	  channels/sig_pri.c: Add ETSI Explicit Call Transfer (ECT)
	  support. Added ability to send and receive ETSI Explicit Call
	  Transfer (ECT) messages to eliminate tromboned calls. Note:
	  Asterisk already supported initiating the transfer of calls to
	  eliminate tromboned calls to libpri so there was nothing to do
	  for the asterisk portion. Review:
	  https://reviewboard.asterisk.org/r/520/

2010-06-02 13:32 +0000 [r266877]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/bridging.c: pthread_join to assure the thread is really gone
	  (closes issue #15465) Reported by: fnordian Patches:
	  bridging.patch uploaded by fnordian (license 110) Tested by:
	  lmadsen, fnordian, peterh Review:
	  https://reviewboard.asterisk.org/r/679/

2010-06-01 22:14 +0000 [r266832]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_exchange.c: Use the correct ical.h file

2010-06-01 21:28 +0000 [r266828]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, tests/test_locale.c
	  (added), configure.ac, configs/voicemail.conf.sample,
	  include/asterisk/localtime.h, main/stdtime/localtime.c, CHANGES,
	  apps/app_voicemail.c: Support setting locale per-mailbox (changes
	  date/time languages for email, pager messages). (closes issue
	  #14333) Reported by: klaus3000 Patches:
	  20090515__issue14333.diff.txt uploaded by tilghman (license 14)
	  app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000

2010-06-01 21:12 +0000 [r266786]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, UPGRADE.txt: Set app and appdata fields when a
	  Dial is redirected (closes issue #17204) Reported by: one47
	  Tested by: twilson, one47

2010-06-01 18:02 +0000 [r266592-266735]  Tilghman Lesher <tlesher@digium.com>

	* res/res_smdi.c: Don't register functions until the last possible
	  point, so they're not unloaded unnecessarily. (closes issue
	  #15996) Reported by: junky Patches: sdmi_wait.diff uploaded by
	  junky (license 177)

	* main/manager.c: Eliminate stale manager events after a set
	  interval, even if AMI clients don't query for them. Actions (or
	  failures to act) by external clients should not cause memory
	  leaks in Asterisk, especially when those continued leaks could
	  cause Asterisk to misbehave later. (closes issue #17234) Reported
	  by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by
	  tilghman (license 14) 20100517__issue17234__trunk.diff.txt
	  uploaded by tilghman (license 14) Tested by: mav3rick, davidw
	  (closes issue #17365) Reported by: davidw

	* /, main/asterisk.c: Merged revisions 266585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
	  | 11 lines Prevent CLI prompt from distorting output of lines
	  shorter than the prompt. Uses the VT100 method of clearing the
	  line from the cursor position to the end of the line: Esc-0K
	  (closes issue #17160) Reported by: coolmig Patches:
	  20100531__issue17160.diff.txt uploaded by tilghman (license 14)
	  Tested by: coolmig ........

2010-05-30 20:18 +0000 [r266438-266522]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Needs to be wrapped in <para>

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 266437 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010)
	  | 2 lines Reverting patch and reopening issue #16784, as patch
	  breaks color display. ........

2010-05-28 22:54 +0000 [r266386]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_icalendar.c, configure, configure.ac,
	  res/res_calendar_caldav.c: Fix ical library handling (again)
	  Newer versions of libical (which we require) store the header
	  file in a libical/ subfolder and include an ical.h file that does
	  a #warning for deprecation and then #includes <libical/ical.h>.
	  Since we now test for libical/ical.h, we can change the #includes
	  back to <libical/ical.h> and remove the test which specifically
	  adds /usr/include/libical as an include directory.

2010-05-28 22:50 +0000 [r266337-266385]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c, UPGRADE.txt, main/asterisk.c: Setup environment
	  variables for the benefit of child processes and disallow
	  changing them. (closes issue #14899) Reported by: jmls Patches:
	  20090916__issue14899.diff.txt uploaded by tilghman (license 14)
	  Tested by: jmls

	* main/asterisk.c: Only report swap on platforms which can examine
	  those statistics

2010-05-28 17:55 +0000 [r266292]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes crash when creation of UDPTL fails
	  (closes issue #17264) Reported by: falves11 Patches:
	  issue_17264_reviewboard_fix.diff uploaded by dvossel (license
	  671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
	  (license 671) Tested by: falves11

2010-05-28 17:34 +0000 [r266289]  Terry Wilson <twilson@digium.com>

	* configure, configure.ac, makeopts.in: More build fixes for
	  ical/neon and res_calendar_ews

2010-05-27 20:08 +0000 [r266240]  Jeff Peeler <jpeeler@digium.com>

	* pbx/pbx_realtime.c: fix compile error

2010-05-27 19:25 +0000 [r266146-266238]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_realtime.c, CHANGES: Cache query results for one second.
	  Queries from the PBX core come in 3's. Caching avoids the
	  additional performance penalty from those two additional queries
	  hitting the database. (closes issue #16521) Reported by: tilghman
	  Patches: 20091229__issue16521.diff.txt uploaded by tilghman
	  (license 14) Tested by: Hubguru, tilghman

	* /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
	  revisions 266142 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
	  | 14 lines Use sigaction for signals which should persist past
	  the initial trigger, not signal. If you call signal() in a
	  Solaris signal handler, instead of just resetting the signal
	  handler, it causes the signal to refire, because the signal is
	  not marked as handled prior to the signal handler being called.
	  This effectively causes Solaris to immediately exceed the
	  threadstack in recursive signal handlers and crash. (closes issue
	  #17000) Reported by: rmcgilvr Patches:
	  20100526__issue17000.diff.txt uploaded by tilghman (license 14)
	  Tested by: rmcgilvr ........

2010-05-26 20:17 +0000 [r266092-266098]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Remove redundant ast_conntected_line_free call.
	  This wouldn't cause any problems, but it's certainly not needed
	  either.

	* res/res_musiconhold.c: Remove unrelated MOH change from previous
	  commit. Thanks Kevin!

	* main/channel.c, res/res_musiconhold.c: Fix misspelling of macro
	  args.

2010-05-26 19:46 +0000 [r266006-266090]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, main/app.c, channels/sip/config_parser.c,
	  channels/sip/include/sip.h: do all sip registry parsing before
	  transmit_register This patch breaks up every part of the sip
	  registry string during config parsing and removes all parsing
	  from transmit_register(). Thanks to Nick_Lewis for contributing
	  this patch! (closes issue #14331) Reported by: Nick_Lewis
	  Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis
	  (license 657) chan_sip.c.patch uploaded by Nick Lewis (license
	  657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis
	  (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis
	  (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis
	  (license 657) nicklewispatch.diff uploaded by dvossel (license
	  671) Tested by: Nick_Lewis, dvossel Review:
	  https://reviewboard.asterisk.org/r/628/

	* channels/chan_sip.c: fixes failed SIP Directed pickup resulting
	  in dead channel (closes issue #17339) Reported by: one47 Patches:
	  sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
	  one47, dvossel

2010-05-26 16:23 +0000 [r265894-265923]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, /: Merged revisions 265910 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26
	  May 2010) | 7 lines Not finding rows in the DB does not rise to
	  the level of a warning. (closes issue #17062) Reported by:
	  drookie Patches: 20100525__issue17062.diff.txt uploaded by
	  tilghman (license 14) ........

	* res/res_config_pgsql.c, configs/res_pgsql.conf.sample: Construct
	  socket name, according to the Postgres docs, and document as
	  such. (closes issue #17392) Reported by: dps Patches:
	  20100525__issue17392.diff.txt uploaded by tilghman (license 14)
	  Tested by: dps

2010-05-26 14:45 +0000 [r265842-265844]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: .......

	* channels/chan_sip.c: Re-enable "always" option for videosupport
	  option in sip.conf. (closes issue #17016) Reported by: twilson
	  Patches: 17016.patch uploaded by mmichelson (license 60) Tested
	  by: devmod

2010-05-26 05:33 +0000 [r265793]  Terry Wilson <twilson@digium.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  res/res_calendar_ews.c: Ensure that libneon > 0.29.0 is installed
	  for res_calendar_ews This uses a modified version of pabelanger's
	  patch that checks for NTLM support instead, which was added in
	  0.29.0 which is what is required for res_calendar_ews. (closes
	  issue #17391) Reported by: loloski Patches: issue17391.patch.v2
	  uploaded by pabelanger (license 224) Tested by: twilson

2010-05-26 00:29 +0000 [r265747]  Tilghman Lesher <tlesher@digium.com>

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  configure, include/asterisk/autoconfig.h.in, configure.ac,
	  pbx/pbx_lua.c, res/res_calendar_caldav.c, res/res_calendar_ews.c:
	  Use configure to determine the prefixes and include directories
	  properly. This ensures cross-platform compatibility, even among
	  Linux distributions, which don't always put headers in the same
	  place. (closes issue #17391) Reported by: loloski

2010-05-25 20:59 +0000 [r265698]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Properly use peer's outboundproxy for
	  outbound REGISTERs. The logic used in transmit_register to get
	  the outboundproxy for a peer was flawed since this value would be
	  overridden shortly afterwards when create_addr was called. In
	  addition, this also fixes some logic used when parsing users.conf
	  so that the peer name is placed in the internally-generated
	  register string so that an outboundproxy set in the Asterisk GUI
	  will be used for outbound REGISTERs.

2010-05-25 17:00 +0000 [r265611]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 265610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
	  2010) | 8 lines Don't mark the cdr records of unanswered queue
	  calls with "NOANSWER". This restores the behavior prior to
	  r258670. (closes issue #17334) Reported by: jvandal Patches:
	  queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
	  by: aragon, jvandal ........

2010-05-25 16:23 +0000 [r265608]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Memory leak in connected line data when SIP blond
	  transfer done. The handling of the control subclass
	  AST_CONTROL_READ_ACTION frame leaked connected line string memory
	  in __ast_read(). Also in __ast_read() the frame type switch
	  should not have had a case for AST_CONTROL_READ_ACTION.
	  AST_CONTROL_READ_ACTION is not a frame type.

2010-05-25 08:31 +0000 [r265525]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian)

2010-05-24 22:21 +0000 [r265467]  Terry Wilson <twilson@digium.com>

	* doc/manager_1_1.txt, main/manager.c, main/asterisk.c: Merge the
	  rest of the FullyBooted patch

2010-05-24 22:16 +0000 [r265449-265453]  Mark Michelson <mmichelson@digium.com>

	* apps/app_senddtmf.c: Allow SendDTMF to play digits to a specified
	  channel. Patch supplied by reporter was modified to use
	  autoservice and prevent a potential channel ref leak but is
	  otherwise as the reporter uploaded it. (closes issue #17182)
	  Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded
	  by rcasas (license 641)

	* channels/h323/ast_h323.cxx: Print openh323 log to the Asterisk
	  console. (closes issue #17109) Reported by: under Patches:
	  logstream.diff uploaded by under (license 914)

	* channels/chan_sip.c: Allow type=user SIP endpoints to be loaded
	  properly from realtime. (closes issue #16021) Reported by:
	  Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand
	  (license 897) (altered by me slightly to avoid ref leaks) Tested
	  by: Guggemand

2010-05-24 20:08 +0000 [r265367]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_rpt.c: Make app_rpt.c able to compile again.

2010-05-24 19:42 +0000 [r265366]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: reverses incorrect logic introduced by
	  r243200 The decoding of the replace_id did not need to be broken
	  up in this instance. This was brought to my attention again
	  because it caused a segfault when the from or to tags were not
	  present in the "Replaces" header.

2010-05-24 19:06 +0000 [r265317-265320]  Terry Wilson <twilson@digium.com>

	* doc/tex/manager.tex: Add the FullyBooted AMI event It is possible
	  to connect to the manager interface before all Asterisk modules
	  are loaded. To ensure that an application does not send AMI
	  actions that might require a module that has not yet loaded, the
	  application can listen for the FullyBooted manager event. It will
	  be sent upon connection if all modules have been loaded, or as
	  soon as loading is complete. The event: Event: FullyBooted
	  Privilege: system,all Status: Fully Booted Review:
	  https://reviewboard.asterisk.org/r/639/

	* CREDITS, configs/calendar.conf.sample, CHANGES,
	  res/res_calendar_ews.c (added), res/res_calendar.c: Calendaring
	  support for Exchange Server 2007+ via EWS This commit adds
	  support for calendaring with Exchange Server 2007+ via Exchange
	  Web Services. Full write support and for querying attendees. Many
	  thanks to Jan Kaláb for the feature. (closes issue #17022)
	  Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel
	  (license 1008) Tested by: pitel, twilson Review:
	  https://reviewboard.asterisk.org/r/557/ Review:
	  https://reviewboard.asterisk.org/r/668/

2010-05-24 18:19 +0000 [r265316]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: On systems with a LOT of RAM, a signed integer
	  sometimes printed negative. (closes issue #16837) Reported by:
	  jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by
	  tilghman (license 14)

2010-05-24 16:10 +0000 [r265273]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes segfault when using generic plc

2010-05-23 18:23 +0000 [r265227]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: small changes to avoiding 'freeing unused
	  memory...'

2010-05-21 22:46 +0000 [r265174]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Channel initialization failure causes crashes.
	  __ast_channel_alloc_ap() has several points in the initialization
	  of a new channel structure where it could fail. Since the channel
	  structure is now an ao2 object, the destructor callback needs to
	  be able to handle clean up when the structure setup is
	  incomplete. Problems corrected: 1) Failing to setup the alertpipe
	  would not unreference the structure but free it directly. Doing
	  this to an ao2_object is very bad. 2) File descriptors need to be
	  initialized to -1 before a construction failure could occur so
	  the destructor will not close unopened descriptors. 3) The
	  destructor needs to check that the string field has been
	  initialized before using any string field values. Crashes
	  expected. 4) The destructor should not notify devstate if the
	  device name is empty. It is a waste of cycles and a couple ERROR
	  log messages are generated. Review:
	  https://reviewboard.asterisk.org/r/675/

2010-05-21 21:08 +0000 [r264953-265090]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/file.h, /, apps/app_queue.c: Merged revisions
	  265089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
	  2010) | 8 lines Don't hang up on a queue caller if the file we
	  attempt to play does not exist. This also fixes a documentation
	  mistake in file.h that made my original attempt to correct this
	  problem not work correctly. (closes issue #17061) Reported by:
	  RoadKill ........

	* channels/chan_sip.c: Be sure to set the sin_family on the proxy
	  when allocating. (closes issue #17157) Reported by: stuarth

	* /, include/asterisk/channel.h: Merged revisions 264999 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May
	  2010) | 3 lines Fix grammatical error in comment. ........

	* main/channel.c, main/autoservice.c, /,
	  include/asterisk/channel.h: Merged revisions 264996 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri,
	  21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific
	  frames until after the sleep has concluded. From reviewboard
	  Background: A Digium customer discovered a somewhat odd bug. The
	  setup is that parties A and B are bridged, and party A places
	  party B on hold. While party B is listening to hold music, he
	  mashes a bunch of DTMF. Party A takes party B off hold while this
	  is happening, but party B continues to hear hold music. I could
	  reproduce this about 1 in 5 times. The issue: When DTMF features
	  are enabled and a user presses keys, the channel that the DTMF is
	  streamed to is placed in an ast_safe_sleep for 100 ms, the
	  duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is
	  read from the channel during the sleep, the frame is dropped.
	  Thus the unhold indication is never made to the channel that was
	  originally placed on hold. The fix: Originally, I discussed with
	  Kevin possible ways of fixing the specific problem reported.
	  However, we determined that the same type of problem could happen
	  in other situations where ast_safe_sleep() is used. Using
	  autoservice as a model, I modified ast_safe_sleep_conditional()
	  to defer specific frame types so they can be re-queued once the
	  sleep has finished. I made a common function for determining if a
	  frame should be deferred so that there are not two identical
	  switch blocks to maintain. Review:
	  https://reviewboard.asterisk.org/r/674/ ........

	* res/res_fax.c, include/asterisk/res_fax.h,
	  res/res_fax.exports.in, res/res_fax_spandsp.c: Log spandsp's fax
	  debug output to the FAX logger level. Review:
	  https://reviewboard.asterisk.org/r/658

2010-05-21 01:00 +0000 [r264905]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Take dup'd code for directmedia ACLs and
	  make utility func The same code was repeated in lots of different
	  places, so I made a utility fuction for it. This should make the
	  merge in the v6-new branch easier.

2010-05-20 23:29 +0000 [r264828]  Richard Mudgett <rmudgett@digium.com>

	* /, main/callerid.c: Merged revisions 264820 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
	  | 6 lines ast_callerid_parse() had a path that left name
	  uninitialized. Several callers of ast_callerid_parse() do not
	  initialize the name parameter before calling thus there is the
	  potential to use an uninitialized pointer. ........

2010-05-20 22:23 +0000 [r264752-264779]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Let ExtensionState resolve dynamic hints. (closes
	  issue #16623) Reported by: tilghman Patches:
	  20100116__issue16623.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen

	* apps/app_stack.c: Error message fix. (closes issue #17356)
	  Reported by: kenner Patches: app_stack.c.diff uploaded by kenner
	  (license 1040)

2010-05-20 20:49 +0000 [r264669-264711]  Richard Mudgett <rmudgett@digium.com>

	* main/ccss.c: Avoid crash in generic CC agent init if caller name
	  or number is NULL.

	* apps/app_dial.c, apps/app_queue.c: Dial and queue connected line
	  update macro not always run when expected. The connected line
	  update macro would not get run if the connected line number
	  string was empty. The number could be empty if the connected line
	  update did not update a number but the name. It should be run if
	  there was an AST_CONTROL_CONNECTED_LINE frame received for
	  pending dials and queues. Renamed and added some more comments
	  for some confusing identifiers directly connected to the related
	  code. Also fixed a memory leak in app_queue. Review:
	  https://reviewboard.asterisk.org/r/669/

2010-05-20 17:54 +0000 [r264626]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
	  channels/sip/include/sip.h: Add support for direct media ACLs
	  directmediapermit/directmediadeny support to restrict which peers
	  can do directmedia based on ip address. In some networks not all
	  phones are fully routed, i.e. not all phones can ping each other.
	  This patch adds a way to restrict directmedia for certain peers
	  between certain networks. (closes issue #16645) Reported by:
	  raarts Patches: directmediapermit.patch uploaded by raarts
	  (license 937) Tested by: raarts Review:
	  https://reviewboard.asterisk.org/r/467/

2010-05-20 15:30 +0000 [r264497-264540]  Kevin P. Fleming <kpfleming@digium.com>

	* addons/ooh323c/src/h323, addons/ooh323c/src: Ignore pre-processed
	  source files generated during DONT_OPTIMIZE dev-mode builds.

	* main/logger.c: Correct 'all logger levels' patch to work
	  properly. Nick Lewis pointed out that the patch as committed
	  wouldn't actually include dynamic logger levels, which was missed
	  by the other reviewers. Thanks!

2010-05-19 21:29 +0000 [r264452]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, channels/chan_sip.c, include/asterisk/_private.h,
	  include/asterisk/options.h, main/asterisk.c, main/loader.c: Fix
	  transcode_via_sln option with SIP calls and improve PLC usage.
	  From reviewboard: The problem here is a bit complex, so try to
	  bear with me... It was noticed by a Digium customer that generic
	  PLC (as configured in codecs.conf) did not appear to actually be
	  having any sort of benefit when packet loss was introduced on an
	  RTP stream. I reproduced this issue myself by streaming a file
	  across an RTP stream and dropping approx. 5% of the RTP packets.
	  I saw no real difference between when PLC was enabled or disabled
	  when using wireshark to analyze the RTP streams. After analyzing
	  what was going on, it became clear that one of the problems faced
	  was that when running my tests, the translation paths were being
	  set up in such a way that PLC could not possibly work as
	  expected. To illustrate, if packets are lost on channel A's read
	  stream, then we expect that PLC will be applied to channel B's
	  write stream. The problem is that generic PLC can only be done
	  when there is a translation path that moves from some codec to
	  SLINEAR. When I would run my tests, I found that every single
	  time, read and write translation paths would be set up on channel
	  A instead of channel B. There appeared to be no real way to
	  predict which channel the translation paths would be set up on.
	  This is where Kevin swooped in to let me know about the
	  transcode_via_sln option in asterisk.conf. It is supposed to work
	  by placing a read translation path on both channels from the
	  channel's rawreadformat to SLINEAR. It also will place a write
	  translation path on both channels from SLINEAR to the channel's
	  rawwriteformat. Using this option allows one to predictably set
	  up translation paths on all channels. There are two problems with
	  this, though. First and foremost, the transcode_via_sln option
	  did not appear to be working properly when I was placing a SIP
	  call between two endpoints which did not share any common
	  formats. Second, even if this option were to work, for PLC to be
	  applied, there had to be a write translation path that would go
	  from some format to SLINEAR. It would not work properly if the
	  starting format of translation was SLINEAR. The one-line change
	  presented in this review request in chan_sip.c fixed the first
	  issue for me. The problem was that in sip_request_call, the
	  jointcapability of the outbound channel was being set to the
	  format passed to sip_request_call. This is nativeformats of the
	  inbound channel. Because of this, when
	  ast_channel_make_compatible was called by app_dial, both channels
	  already had compatibly read and write formats. Thus, no
	  translation path was set up at the time. My change is to set the
	  jointcapability of the sip_pvt created during sip_request_call to
	  the intersection of the inbound channel's nativeformats and the
	  configured peer capability that we determined during the earlier
	  call to create_addr. Doing this got the translation paths set up
	  as expected when using transcode_via_sln. The changes presented
	  in channel.c fixed the second issue for me. First and foremost,
	  when Asterisk is started, we'll read codecs.conf to see the value
	  of the genericplc option. If this option is set, and ast_write is
	  called for a frame with no data, then we will attempt to fill in
	  the missing samples for the frame. The implementation uses a
	  channel datastore for maintaining the PLC state and for creating
	  a buffer to store PLC samples in. Even when we receive a frame
	  with data, we'll call plc_rx so that the PLC state will have
	  knowledge of the previous voice frame, which it can use as a
	  basis for when it comes time to actually do a PLC fill-in. So,
	  reviewers, now I ask for your help. First off, there's the one
	  line change in chan_sip that I have put in. Is it right? By my
	  logic it seems correct, but I'm sure someone can tell me why it
	  is not going to work. This is probably the change I'm least
	  concerned about, though. What concerns me much more is the set of
	  changes in channel.c. First off, am I even doing it right? When I
	  run tests, I can clearly see that when PLC is activated, I see a
	  significant increase in RTP traffic where I would expect it to
	  be. However, in my humble opinion, the audio sounds kind of
	  crappy whenever the PLC fill-in is done. It sounds worse to me
	  than when no PLC is used at all. I need someone to review the
	  logic I have used to be sure that I'm not misusing anything. As
	  far as I can see my pointer arithmetic is correct, and my use of
	  AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
	  someone can point out somewhere where I've done something
	  incorrectly. As I was writing this review request up, I decided
	  to give the code a test run under valgrind, and I find that for
	  some reason, calls to plc_rx are causing some invalid reads.
	  Apparently I'm reading past the end of a buffer somehow. I'll
	  have to dig around a bit to see why that is the case. If it's
	  obvious to someone reviewing, speak up! Finally, I have one other
	  proposal that is not reflected in my code review. Since without
	  transcode_via_sln set, one cannot predict or control where a
	  translation path will be up, it seems to me that the current
	  practice of using PLC only when transcoding to SLINEAR is not
	  useful. I recommend that once it has been determined that the
	  method used in this code review is correct and works as expected,
	  then the code in translate.c that invokes PLC should be removed.
	  Review: https://reviewboard.asterisk.org/r/622/

2010-05-19 20:30 +0000 [r264400]  David Vossel <dvossel@digium.com>

	* main/udptl.c: fixes infinite loop during udptl.c's
	  decode_open_type When decode_length returns the length there is a
	  check to see if that length is negative, if so the decode loop
	  breaks as this means the limit has been reached. The problem here
	  is that length is an unsigned int, so length can never be
	  negative. This resulted in an infinite loop. (issue #17352)

2010-05-19 20:26 +0000 [r264335-264379]  Matthew Nicholson <mnicholson@digium.com>

	* main/udptl.c: Cast an unsigned int to a signed int when comparing
	  it with 0. (AST-377)

	* /, apps/app_speech_utils.c: Merged revisions 264334 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed,
	  19 May 2010) | 5 lines Set quieted flag when receiving a dtmf
	  tone during playback in speechbackground. (closes issue #16966)
	  Reported by: asackheim ........

2010-05-19 19:21 +0000 [r264331]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes crash in check_rtp_timeout During
	  deadlock avoidance the sip dialog pvt is locked and unlocked.
	  When this occurs we have no guarantee the pvt's owner is still
	  valid. We were trying to access the pvt's owner after this
	  without checking to see if it still existed first. (closes issue
	  #17271) Reported by: under Patches: check_rtp_timeout.diff
	  uploaded by under (license 914) Tested by: dvossel

2010-05-19 17:48 +0000 [r264204-264249]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/options.h: Merged revisions 264248 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19
	  May 2010) | 17 lines Internal timing is now on by default, if
	  you're using DAHDI 2.3 or above. The reason for ensuring DAHDI
	  2.3 or above is that this version ensures that a timer is always
	  available, whereas in previous versions, it was possible for
	  DAHDI to be loaded, but have no drivers to actually generate
	  timing. If internal_timing was turned on in this circumstance, a
	  complete lack of audio would result. This is the reason why
	  internal_timing was not on by default. However, now that DAHDI
	  ensures the availability of a timer, there is no reason for this
	  setting to be off (and in fact, it solves a great many initial
	  user problems). (closes issue #15932) Reported by: dimas Patches:
	  20100519__issue15932.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman ........

	* main/dsp.c: Keep track of digit duration, when we're decoding
	  inband to pass DTMF frames. (closes issue #17235) Reported by:
	  frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license
	  610) 20100518__issue17235.diff.txt uploaded by tilghman (license
	  14) Tested by: frawd

2010-05-19 15:39 +0000 [r264161]  Leif Madsen <lmadsen@digium.com>

	* main/cli.c: Fix compilation problem with previous commit. (issue
	  #16009)

2010-05-19 15:29 +0000 [r264160]  Kevin P. Fleming <kpfleming@digium.com>

	* main/logger.c, configs/logger.conf.sample: Add ability for logger
	  channels to include *all* levels. Now that Asterisk modules can
	  dynamically create and destroy logger levels on demand, it's
	  useful to be able to configure a logger channel (console, file,
	  whatever) to be able to accept log messages from *all* levels,
	  even levels created dynamically. This patch adds support for
	  this, by allowing the '*' level name to be used in logger.conf.
	  Review: https://reviewboard.asterisk.org/r/663/

2010-05-19 15:12 +0000 [r264117]  Leif Madsen <lmadsen@digium.com>

	* CHANGES, main/cli.c: Add ability to hangup all channels from the
	  CLI. Added the keyword 'all' to the 'channel hangup request' CLI
	  command so that you can request all channels to be hungup without
	  having to restart Asterisk. (closes issue #16009) Reported by:
	  moy Patches: hangup-all-rev-221688.patch uploaded by moy (license
	  222) Tested by: moy, russell

2010-05-19 14:38 +0000 [r264114]  David Vossel <dvossel@digium.com>

	* res/res_rtp_asterisk.c: fixes crash during dtmf During the
	  processing of Cisco dtmf the dtmf samples were not being
	  calculated correctly. In an attempt to determine what sample rate
	  was being used, a NULL frame was processed which caused a crash.
	  This patch resolves this. (closes issue #17248) Reported by:
	  falves11 Patches: issue_17248.diff uploaded by dvossel (license
	  671)

2010-05-19 08:09 +0000 [r264031]  Alec L Davis <sivad.a@paradise.net.nz>

	* configs/indications.conf.sample: fix incorrectly typed
	  indications for [nz] stutter and dialrecall (closes issue #17359)
	  Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
	  alecdavis (license 585)

2010-05-19 06:41 +0000 [r263905-263950]  Tilghman Lesher <tlesher@digium.com>

	* /, main/dsp.c: Merged revisions 263949 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
	  | 8 lines Because progress is called multiple times, across
	  several frames, we must persist states when detecting multitone
	  sequences. (closes issue #16749) Reported by: dant Patches:
	  dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
	  dant ........

	* configure, configure.ac, build_tools/sha1sum-sh (added),
	  makeopts.in, sounds/Makefile: Add an sha1sum-workalike for
	  platforms which don't have it (like Mac OS X)

2010-05-18 22:48 +0000 [r263904]  David Vossel <dvossel@digium.com>

	* main/strings.c: fixes segfault on logging (closes issue #17331)
	  Reported by: under Patches: utils.diff uploaded by under (license
	  914) segfault_on_logging.diff uploaded by dvossel (license 671)
	  Tested by: under, dvossel

2010-05-18 21:09 +0000 [r263860]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Be sure to heap-allocate the redirecting to
	  tag so as not to cause crashiness.

2010-05-18 20:49 +0000 [r263858]  Tilghman Lesher <tlesher@digium.com>

	* res/res_timing_kqueue.c: Make happy green color come back

2010-05-18 20:09 +0000 [r263810]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix memory leaks in redirecting structures
	  in chan_sip.c Thanks to Richard for pointing this out.

2010-05-18 19:30 +0000 [r263807-263808]  Jeff Peeler <jpeeler@digium.com>

	* CHANGES: put changes with the correct version

	* /, CHANGES, apps/app_directory.c: Merged revisions 263769 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
	  | 10 lines Modify directory name reading to be interrupted with
	  operator or pound escape. In the case of accidentally entering
	  the wrong first three letters for the reading, users could be
	  very frustrated if the name listing is very long. This allows
	  interrupting the reading by pressing 0 or #. 0 will attempt to
	  execute a configured operator (o) extension and # will exit and
	  proceed in the dialplan. ABE-2200 ........

2010-05-17 23:49 +0000 [r263724]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  makeopts.in, sounds/Makefile, autoconf/ast_ext_lib.m4: Cache
	  sound tarfiles in a common directory, such that a clean reinstall
	  does not force a re-download of the tarballs. (closes issue
	  #15370) Reported by: pprindeville Patches:
	  asterisk-trunk-bugid15370.patch uploaded by pprindeville (license
	  347) Tested by: pprindeville, tilghman, seanbright

2010-05-17 22:08 +0000 [r263640]  Mark Michelson <mmichelson@digium.com>

	* /, main/devicestate.c: Merged revisions 263639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
	  2010) | 10 lines Fix logic error when checking for a devstate
	  provider. When using strsep, if one of the list of specified
	  separators is not found, it is the first parameter to strsep
	  which is now NULL, not the pointer returned by strsep. This issue
	  isn't especially severe in that the worst it is likely to do is
	  waste some cycles when a device with no '/' and no ':' is passed
	  to ast_device_state. ........

2010-05-17 19:31 +0000 [r263589]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: With IMAP backend, messages in INBOX were
	  counted twice for MWI. (closes issue #17135) Reported by:
	  edhorton Patches: 20100513__issue17135.diff.txt uploaded by
	  tilghman (license 14) 17135_2.diff uploaded by ebroad (license
	  878) Tested by: edhorton, ebroad

2010-05-17 15:36 +0000 [r263541]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, channels/chan_local.c, main/rtp_engine.c,
	  channels/chan_sip.c, include/asterisk/channel.h,
	  configs/misdn.conf.sample, apps/app_queue.c,
	  funcs/func_redirecting.c, channels/misdn_config.c,
	  main/channel.c, main/dial.c, channels/chan_dahdi.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h, main/features.c,
	  funcs/func_connectedline.c, include/asterisk/frame.h,
	  funcs/func_callerid.c, channels/sip/include/sip.h: Enhancements
	  to connected line and redirecting work. From reviewboard: Digium
	  has a commercial customer who has made extensive use of the
	  connected party and redirecting information present in later
	  versions of Asterisk Business Edition and which is to be in the
	  upcoming 1.8 release. Through their use of the feature, new
	  problems and solutions have come about. This patch adds several
	  enhancements to maximize usage of the connected party and
	  redirecting information functionality. First, Asterisk trunk
	  already had connected line interception macros. These macros
	  allow you to manipulate connected line information before it was
	  sent out to its target. This patch adds the same feature except
	  for redirecting information instead. Second, the ast_callerid and
	  ast_party_id structures have been enhanced to provide a "tag."
	  This tag can be set with func_callerid, func_connectedline,
	  func_redirecting, and in the case of DAHDI, mISDN, and SIP
	  channels, can be set in a configuration file. The idea behind the
	  callerid tag is that it can be set to whatever value the
	  administrator likes. Later, when running connected line and
	  redirecting macros, the admin can read the tag off the
	  appropriate structure to determine what action to take. You can
	  think of this sort of like a channel variable, except that
	  instead of having the variable associated with a channel, the
	  variable is associated with a specific identity within Asterisk.
	  Third, app_dial has two new options, s and u. The s option lets a
	  dialplan writer force a specific caller ID tag to be placed on
	  the outgoing channel. The u option allows the dialplan writer to
	  force a specific calling presentation value on the outgoing
	  channel. Fourth, there is a new control frame subclass called
	  AST_CONTROL_READ_ACTION added. This was added to correct a very
	  specific situation. In the case of SIP semi-attended (blond)
	  transfers, the party being transferred would not have the
	  opportunity to run a connected line interception macro to
	  possibly alter the transfer target's connected line information.
	  The issue here was that during a blond transfer, the SIP transfer
	  code has no bridged channel on which to queue the connected line
	  update. The way this was corrected was to add this new control
	  frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on
	  the channel on which the connected line interception macro should
	  be run. When ast_read is called to read the frame, ast_read
	  responds by calling a callback function associated with the
	  specific read action the control frame describes. In this case,
	  the action taken is to run the connected line interception macro
	  on the transferee's channel. Review:
	  https://reviewboard.asterisk.org/r/652/

2010-05-17 15:14 +0000 [r263375-263460]  Leif Madsen <lmadsen@digium.com>

	* main/manager.c: Missing newlines added to Set-Cookie line in
	  manager.c Sean Bright pointed out that we lost a set of newline
	  characters in commit 190349 on a line I had recently changed. Yay
	  for code review on commits. (issue #17231, #10961)

	* main/manager.c, /: Recorded merge of revisions 263456 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
	  | 11 lines Manager cookies are not compatible with RFC2109. The
	  Version field in the cookies we're setting contain quotes around
	  the version number which is not compatible with RFC2109 and
	  breaks some implementations. (closes issue #17231) Reported by:
	  ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
	  ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
	  ecarruda (license 559) Tested by: ecarruda, russell ........

	* /, sounds/Makefile: Merged revisions 263374 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
	  | 8 lines Update link to new version of core sounds. The latest
	  version of the core sounds files 1.4.19 now includes the missing
	  queue-minute sound file which is called by app_queue but which
	  has been missing. (closes issue #17123) Reported by: n8ideas
	  ........

2010-05-17 13:05 +0000 [r263294]  David Vossel <dvossel@digium.com>

	* CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option
	  backport to 1.6.2

2010-05-16 16:31 +0000 [r263250]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/live_ast: live_ast: add commands 'rsync' and
	  'gen-live-asterisk' This adds the following two commands to
	  live_ast: * rsync [user]@host directory Copy over all generated
	  files to <directory> at remote host. Would allow running live_ast
	  there. Hence allows separating a build machine from a test
	  machine. * gen-live-asteris: regenerate live/asterisk . Useful if
	  copying over files to a different directory.

2010-05-16 11:14 +0000 [r263208]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c: Improve some very confusing structure names in
	  astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code
	  here called a list of bucket entries a 'bucket', and the entries
	  within the bucket were called 'bucket_list'. This made the code
	  very hard to understand without reading all of it... so I've
	  renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of
	  the structure.

2010-05-14 18:53 +0000 [r263151]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fix iax_frame double free Very unfortunate
	  things happen if we add an iax_frame to the frame queue and let
	  go of the lock before scheduling the frame's transmit... There is
	  a race condition that exists where the frame can be removed from
	  the frame_queue and freed before the transmit is scheduled if we
	  do not hold on to that lock. This results in a freed frame being
	  scheduled for transmit later.

2010-05-13 22:01 +0000 [r263069]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set
	  debug on/off

2010-05-13 20:25 +0000 [r263028]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configure, configure.ac: Remove "untested" feature PRI_VERSION
	  Nobody seems to actually test PRI_VERSION. It is only useful for
	  failing PRI support in chan_dahdi.

2010-05-13 17:49 +0000 [r262940-262987]  Tilghman Lesher <tlesher@digium.com>

	* res/res_timing_kqueue.c: For FreeBSD

	* res/res_timing_kqueue.c: Hmmm, probably should have read the
	  manpage more thoroughly.

2010-05-13 15:36 +0000 [r262895-262897]  Russell Bryant <russell@digium.com>

	* channels/chan_console.c: Fix an off by one error that causes a
	  crash. Thanks to Raymond Burke for pointing it out.

	* main/stdtime/localtime.c: Fix build on linux.

	* pbx/pbx_spool.c: Fix build on linux.

2010-05-13 05:37 +0000 [r262852]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, pbx/pbx_spool.c, tests/test_time.c,
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  main/stdtime/localtime.c, res/res_timing_kqueue.c (added): Add
	  kqueue(2) implementation to Asterisk in various places. This will
	  save a considerable amount of CPU on the BSDs, including Mac OS
	  X, as it eliminates several places in the code that we previously
	  used a busy loop. Additionally, this adds a res_timing interface,
	  using kqueue timers. Review:
	  https://reviewboard.asterisk.org/r/543/

2010-05-12 19:59 +0000 [r262800]  Paul Belanger <paul.belanger@polybeacon.com>

	* main/loader.c, main/cli.c: Notify CLI when modules is loaded /
	  unloaded (closes issue #17308) Reported by: pabelanger Patches:
	  cli.modules.patch uploaded by pabelanger (license 224) Tested by:
	  pabelanger, russell

2010-05-12 19:53 +0000 [r262796-262798]  Leif Madsen <lmadsen@digium.com>

	* res/ael/pval.c: Revert previous WARNING message removal.
	  Marquis42 suggested a better method of doing what I wanted
	  because I ended up removing the WARNING message for all instances
	  when really I just wanted to remove it for the 'return' keyword,
	  not everything. (issue #17145)

	* res/ael/pval.c: Remove unnecessary WARNING message in ael/pval.c
	  (closes issue #17145) Reported by: okrief

2010-05-12 18:01 +0000 [r262744]  David Vossel <dvossel@digium.com>

	* /, apps/app_meetme.c: Merged revisions 262662 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
	  | 11 lines fixes app_meetme dsp error We attempted to detect
	  silence after translating a frame from signed linear. This caused
	  a flooding of errors. To resolve this the code to detect silence
	  was moved before the translation. (closes issue #17133) Reported
	  by: jsdyer ........

2010-05-12 17:57 +0000 [r262661-262743]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Don't crash when destroying chan_dahdi
	  pseudo channels. Must do a deep copy of the cc_params in
	  duplicate_pseudo(). Otherwise, when the duplicate pseudo channel
	  is destroyed, it frees the original pseudo channel cc_params. The
	  original pseudo channel is then left with a dangling pointer for
	  when the next duplicated pseudo channel is created.

	* channels/chan_misdn.c: Merged revisions 262657,262660 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
	  .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed,
	  12 May 2010) | 4 lines Forgot some conditionals around the
	  callrerouting facility help text. JIRA ABE-2223 ..........
	  r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010)
	  | 22 lines Add mISDN Call rerouting facility for point-to-point
	  ISDN lines (exchange line) In the case of ISDN
	  point-to-multipoint (multidevice) you can use the mISDN "facility
	  calldeflect" application for call diversions from external (PSTN)
	  to external (PSTN). In that case this is the only way to get rid
	  of the two call legs to the PBX and let the calling number at the
	  C party become the number of the A party. In the case of ISDN
	  point-to-point (exchange line) the call deflection facility may
	  not be used. Instead a call rerouting facility has to be used.
	  This patch for chan_misdn.c is an extension to realize this
	  service (facility rerouting application). It can accept either
	  spelling: "callrerouting" or "callrerouteing". The patch is
	  tested towards Deutsche Telekom and requires a modified version
	  of mISDN from Digium, Inc. Patches:
	  misdn_rerouteing_corrected.patch (Slightly modified.) JIRA
	  ABE-2223

2010-05-12 16:23 +0000 [r262656]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_privacy.c: Ensure the arguments are initialized. Also
	  miscellaneous CG cleanup. (closes issue #16576) Reported by:
	  uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman
	  (license 14) Tested by: uxbod

2010-05-12 01:00 +0000 [r262613]  Paul Belanger <paul.belanger@polybeacon.com>

	* channels/chan_sip.c, include/asterisk/cli.h: Convert to
	  AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new
	  AST_CLI functions (closes issue #17287) Reported by: pabelanger
	  Patches: issue17287.patch uploaded by pabelanger (license 224)
	  Tested by: russell

2010-05-11 23:18 +0000 [r262569]  Richard Mudgett <rmudgett@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  channels/sig_pri.c: Dialing an invalid extension causes
	  incomplete hangup sequence. Revision -r1489 of the libpri 1.4
	  branch corrected a deviation from Q.931 Section 5.3.2. However,
	  this resulted in an unexpected behaviour change to the upper
	  layer (Asterisk). This change uses pri_hangup_fix_enable() to
	  follow Q.931 Section 5.3.2 call hangup better if the version of
	  libpri supports it. (issue #17104) Reported by: shawkris Tested
	  by: rmudgett

2010-05-11 21:25 +0000 [r262513]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/causes.h: Move cause 200 to cause 26, as
	  specified in Q.850. Also cleanup the formatting and add a few
	  more that seem like good candidates. (closes issue #16157)
	  Reported by: wimpy

2010-05-11 19:57 +0000 [r262422]  Jason Parker <jparker@digium.com>

	* /, res/Makefile: Merged revisions 262421 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
	  11 lines Use a less silly method for modifying a flex-generated
	  file. The sed syntax that was used wasn't actually valid, causing
	  some versions to choke. This is the method that is used in 1.6.x+
	  for similar changes. (closes issue #16696) Reported by: bklang
	  Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
	  by: qwell ........

2010-05-11 19:40 +0000 [r262414-262419]  Paul Belanger <paul.belanger@polybeacon.com>

	* pbx/pbx_config.c: Improve logging by displaying line number
	  (closes issue #16303) Reported by: dant Patches:
	  issue16303.patch.v2 uploaded by pabelanger (license 224) Tested
	  by: dant, lmadsen, pabelanger

	* channels/chan_sip.c: Improve logging information for
	  misconfigured contexts (closes issue #17238) Reported by:
	  pprindeville Patches: chan_sip-bug17238.patch uploaded by
	  pprindeville (license 347) Tested by: pprindeville

2010-05-11 17:23 +0000 [r262330]  Tilghman Lesher <tlesher@digium.com>

	* /, Makefile.rules, apps/app_voicemail.c: Merged revisions 262321
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010)
	  | 2 lines Fix issue #17302 a slightly different way (mad props to
	  Qwell) ........

2010-05-11 16:43 +0000 [r262299]  Jason Parker <jparker@digium.com>

	* bootstrap.sh: Allow bootstrap script to work on Solaris. As
	  usual, the way they do things is different, so we need to account
	  for that. automake is versioned ala BSD/Linux, but autoconf is
	  not. We don't actually need to specify a version there, since
	  AC_PREREQ will cover it for us. Things will fail pretty loudly if
	  AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang
	  Patches: opensolaris_bootstrap.sh uploaded by bklang (license
	  919)

2010-05-10 19:06 +0000 [r262236-262240]  David Vossel <dvossel@digium.com>

	* apps/app_directed_pickup.c: fixes PickupChan application (closes
	  issue #16863) Reported by: schern Patches:
	  app_directed_pickup.c.patch uploaded by schern (license 995)
	  for_trunk.diff uploaded by cjacobsen (license 1029) Tested by:
	  Graber, cjacobsen, lathama, rickead2000, dvossel

	* channels/chan_console.c: fixes crash in chan_console There is a
	  race condition between console_hangup() and start_stream(). It is
	  possible for console_hangup() to be called and then the stream
	  thread to begin after the hangup. To avoid this a check in
	  start_stream() to make sure the pvt-owner still exists while the
	  pvt lock is held is made. If the owner is gone that means the
	  channel hung up and start_stream should be aborted.

2010-05-10 16:36 +0000 [r262152]  Tilghman Lesher <tlesher@digium.com>

	* /, Makefile.rules: Merged revisions 262151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
	  | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
	  issue #17297) Reported by: jcovert Patches:
	  20100506__issue17297.diff.txt uploaded by tilghman (license 14)
	  (closes issue #17302) Reported by: jcovert ........

2010-05-09 02:14 +0000 [r262048-262102]  Tilghman Lesher <tlesher@digium.com>

	* autoconf/ast_c_define_check.m4, configure,
	  include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4,
	  autoconf/ast_c_compile_check.m4: Cleanup a bit more by getting
	  rid of useless version defines. Also make library detection use
	  passed CFLAGS. (closes issue #17309) Reported by: stuarth

	* configure, configure.ac: Use CPPFLAGS to pass PTHREAD_CFLAGS for
	  vpb only

2010-05-07 23:54 +0000 [r262005]  Alec L Davis <sivad.a@paradise.net.nz>

	* UPGRADE.txt, apps/app_voicemail.c: VoicemailMain and
	  VMauthenticate, allow escape to the 'a' extension when a single
	  '*' is entered Where a site uses VoicemailMain(mailbox) the users
	  have to be at their own extension to clear their voicemail, they
	  have no way of escaping VoicemailMain to allow entry of new
	  boxnumber. This patch, allows a site to include to 'a' priority
	  in the VoicemailMain context, to allow an escape. If the 'a'
	  priority doesn't exist in the context that VoicemailMain was
	  called from then it acts as the old behaviour. Reported by:
	  alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt
	  uploaded by alecdavis (license 585) Review:
	  https://reviewboard.asterisk.org/r/489/

2010-05-07 22:09 +0000 [r261913-261964]  Tilghman Lesher <tlesher@digium.com>

	* addons/ooh323c/src/ooh323.c: Fix build on Linux

	* funcs/func_odbc.c: Double free crash (closes issue #17245)
	  Reported by: thedavidfactor Patches:
	  20100426__issue17245.diff.txt uploaded by tilghman (license 14)
	  Tested by: murraytm

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Use
	  the detected pthread building flags in every place, instead of
	  hardcoding -lpthread. We nicely detect the right flags on each
	  system for building Asterisk with pthreads, then ignore it for
	  every other build option that requires us to build with pthreads.
	  This caused some items to return a false negative. Also cleanup
	  some minor naming issues that caused "library library" redundancy
	  in the output. (closes issue #17303) Reported by: stuarth
	  Patches: 20100507__issue17303.diff.txt uploaded by tilghman
	  (license 14) Tested by: stuarth

2010-05-07 16:05 +0000 [r261867]  Leif Madsen <lmadsen@digium.com>

	* UPGRADE-1.6.txt: Update UPGRADE-1.6.txt stating insecure=very has
	  been removed. (closes issue #17282) Reported by: stuarth Tested
	  by: stuarth

2010-05-07 15:33 +0000 [r261866]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.c: Fix deadlock in sig_pri when hanging up. The
	  pri_dchannel thread currently violates locking order by locking
	  the private and then attempting to queue a frame, which needs to
	  lock the channel. Queueing a frame is unneccesary though and is
	  actually a regression since sig_pri. All the places that
	  currently use ast_softhangup_nolock now will just set the
	  softhangup value directly as before. (closes issue #17216)
	  Reported by: lmsteffan Patches: bug17216.patch uploaded by
	  jpeeler (license 325)

2010-05-06 23:41 +0000 [r261822]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Some code optimizations. * Made more places
	  use pri_queue_control() instead of pri_queue_frame() and a local
	  frame variable. * Made pri_queue_frame() use
	  sig_pri_lock_owner(). pri_queue_frame() no longer releases the
	  libpri access lock unless it is required. * Made the
	  pri_queue_frame() and pri_queue_control() parameter list similar
	  to sig_pri_lock_owner().

2010-05-06 20:11 +0000 [r261736]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 261735 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06
	  May 2010) | 8 lines Only allow the operator key to be accepted
	  after leaving a voicemail. Or rather disallow the operator key
	  from being accepted when not offered, such as after finishing a
	  recording from within the mailbox options menu. ABE-2121 SWP-1267
	  ........

2010-05-06 17:06 +0000 [r261609]  Jason Parker <jparker@digium.com>

	* /, sounds/Makefile: Merged revisions 261608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
	  4 lines Use the versioned MOH tarballs, now that we have them.
	  This makes for more reproducibility. Prompted by a discussion in
	  #asterisk-dev ........

2010-05-06 15:39 +0000 [r261560]  Tilghman Lesher <tlesher@digium.com>

	* channels/sip/include/sip.h: Permit more lines within a SIP body
	  to be parsed. The example given within the related issue showed
	  120 lines, which was mostly a result of the body being XML.
	  (closes issue #17179) Reported by: khw

2010-05-06 14:15 +0000 [r261496-261500]  Russell Bryant <russell@digium.com>

	* tests/test_heap.c: Add test case for removing random elements
	  from a heap. I modified the original patch for trunk to use the
	  unit test API. (issue #17277) Reported by: cappucinoking Patches:
	  test_heap.diff uploaded by cappucinoking (license 1036) Tested
	  by: cappucinoking, russell

	* main/heap.c: Fix handling of removing nodes from the middle of a
	  heap. This bug surfaced in 1.6.2 and does not affect code in any
	  other released version of Asterisk. It manifested itself as SIP
	  qualify not happening when it should, causing peers to go
	  unreachable. This was debugged down to scheduler entries
	  sometimes not getting executed when they were supposed to, which
	  was in turn caused by an error in the heap code. The problem only
	  sometimes occurs, and it is due to the logic for removing an
	  entry in the heap from an arbitrary location (not just popping
	  off the top). The scheduler performs this operation frequently
	  when entries are removed before they run (when ast_sched_del() is
	  used). In a normal pop off of the top of the heap, a node is
	  taken off the bottom, placed at the top, and then bubbled down
	  until the max heap property is restored (see max_heapify()). This
	  same logic was used for removing an arbitrary node from the
	  middle of the heap. Unfortunately, that logic is full of fail.
	  This patch fixes that by fully restoring the max heap property
	  when a node is thrown into the middle of the heap. Instead of
	  just pushing it down as appropriate, it first pushes it up as
	  high as it will go, and _then_ pushes it down. Lastly, fix a
	  minor problem in ast_heap_verify(), which is only used for
	  debugging. If a parent and child node have the same value, that
	  is not an error. The only error is if a parent's value is less
	  than its children. A huge thanks goes out to cappucinoking for
	  debugging this down to the scheduler, and then producing an
	  ast_heap test case that demonstrated the breakage. That made it
	  very easy for me to focus on the heap logic and produce a fix.
	  Open source projects are awesome. (closes issue #16936) Reported
	  by: ib2 Tested by: cappucinoking, crjw (closes issue #17277)
	  Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded
	  by russell (license 2) Tested by: cappucinoking, russell

2010-05-06 07:27 +0000 [r261451]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c: When failing to configure, don't destroy
	  'cfg' twice Fixes a crash when some config section had an
	  incorrect channel config.

2010-05-05 22:22 +0000 [r261405]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Avoid a crash on SS7 channels.

2010-05-05 20:48 +0000 [r261364]  Russell Bryant <russell@digium.com>

	* Makefile, configs/asterisk.conf.sample: Restore previous
	  asterisk.conf syntax, where the directories aren't commented out.
	  This fixes some breakage in the test suite, that uses the
	  contents of asterisk.conf to discover the install layout on the
	  system.

2010-05-05 19:13 +0000 [r261316]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes sip native transfer The Refer-To
	  header field containing the Replaces header in the URI was not
	  being decoded properly. This caused invalid parsing between the
	  caller id field and the domain resulting in a failed transfer.
	  (closes issue #17284) Reported by: dvossel

2010-05-05 18:43 +0000 [r261314]  Paul Belanger <paul.belanger@polybeacon.com>

	* /, channels/chan_sip.c: Merged revisions 261274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
	  2010) | 12 lines Registration fix for SIP realtime. Make sure
	  realtime fields are not empty. (closes issue #17266) Reported by:
	  Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
	  Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
	  https://reviewboard.asterisk.org/r/643/ ........

2010-05-05 18:28 +0000 [r261313]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/dialplan_functions.c: Prevent unnecessary warnings
	  when getting rtpsource or rtpdest. If a recognized media type was
	  present, but the media type was not enabled for the channel, then
	  a warning would be emitted. For instance, attempting to get
	  CHANNEL(rtpsource,video) on a call with no video would cause a
	  warning message to appear. With this change, the warning will
	  only appear if the stream argument is not recognized as being a
	  media type that can be specified.

2010-05-05 15:42 +0000 [r261124-261232]  Paul Belanger <paul.belanger@polybeacon.com>

	* apps/app_queue.c: 'queue reset stats' erroneously clears
	  wrapuptime configuration. Resets each member's lastcall to 0 now.
	  (closes issue #17262) Reported by: rain Patches:
	  wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
	  by: rain

	* main/manager.c, include/asterisk/cli.h, CHANGES,
	  include/asterisk/manager.h: New 'manager show settings' CLI
	  command. See the CHANGES file for more details. (closes issue
	  #16343) Reported by: pabelanger Patches: issue16343.patch.v5
	  uploaded by pabelanger (license 224) Tested by: pabelanger,
	  tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/

	* Makefile, configs/asterisk.conf.sample (added): New static
	  asterisk.conf.sample file. This simply moves the functionality
	  from the Makefile (cleaning it up) into an external
	  asterisk.conf.samples file. Also updates formatting (easier to
	  read) and grammar changes to asterisk.conf.samples. (closes issue
	  #17027) Reported by: pabelanger Patches:
	  0017027.asterisk.conf.v6.patch uploaded by pabelanger (license
	  224) Tested by: qwell, lmadsen, pabelanger, chappell Review:
	  https://reviewboard.asterisk.org/r/616/

2010-05-04 23:51 +0000 [r261095]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 261093-261094 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04
	  May 2010) | 7 lines Protect against overflow, when calculating
	  how long to wait for a frame. (closes issue #17128) Reported by:
	  under Patches: d.diff uploaded by under (license 914) ........
	  r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010)
	  | 2 lines Add a tiny corner case to the previous commit ........

2010-05-04 22:46 +0000 [r261051]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add new
	  possible value to autopause option to allow members to be
	  autopaused in all queues. See the CHANGES file and
	  queues.conf.sample for more details. (closes issue #17008)
	  Reported by: jlpedrosa Patches: queues.autopause_en_review.diff
	  uploaded by jlpedrosa (license 1002) Review:
	  https://reviewboard.asterisk.org/r/581/

2010-05-04 21:10 +0000 [r261007]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h, channels/sig_pri.c: The inalarm flag is
	  not passed up from the sig_analog and sig_pri submodules. The CLI
	  "dahdi show channel" command was not correctly reporting the
	  InAlarm status. The inalarm flag is now consistently passed
	  between chan_dahdi and submodules.

2010-05-04 18:51 +0000 [r260924]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 260923 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04
	  May 2010) | 12 lines Voicemail transfer to operator should occur
	  immediately, not after main menu. There were two scenarios in the
	  advanced options that while using the operator=yes and review=yes
	  options, the transfer occurred only after exiting the main menu
	  (after sending a reply or leaving a message for an extension).
	  Now after the audio is processed for the reply or message the
	  transfer occurs immediately as expected. ABE-2107 ABE-2108
	  ........

2010-05-04 15:49 +0000 [r260802]  Jason Parker <jparker@digium.com>

	* /, build_tools/make_build_h: Merged revisions 260801 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
	  2010) | 1 line Fix fallout from removing from configure script.
	  Pointed out by philipp64 on #asterisk-dev ........

2010-05-03 22:13 +0000 [r260757]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c, CHANGES: Add new admin features to meetme:
	  Roll call, eject all, mute all, record in-conf This patch adds
	  the following in-conference admin DTMF features: *81 - Roll call
	  (or simply user count if INTROUSER isn't enabled) *82 - Eject all
	  non-admins *83 - Mute/unmute all non-admins *84 - Start recording
	  the conference on the fly FWIW, this code uses newly recorded
	  prompts. (closes issue #16379) Reported by: rfinnie Patches:
	  meetme-enhancements-232771-v1.patch uploaded by rfinnie (license
	  940) modified slightly by me

2010-05-03 17:06 +0000 [r260663]  Paul Belanger <paul.belanger@polybeacon.com>

	* Makefile, /: Merged revisions 260661-260662 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
	  2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
	  libdir when executing mkpkgconfig allowing non-root installs to
	  work. (closes issue #17268) Reported by: pabelanger Patches:
	  issue17268.patch uploaded by pabelanger (license 224) Tested by:
	  pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
	  -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
	  part. Thanks Qwell. ........

2010-05-03 14:58 +0000 [r260570]  Leif Madsen <lmadsen@digium.com>

	* doc/HOWTO_collect_debug_information.txt: Merged revisions 260569
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010)
	  | 1 line Minor typo pointed out by pabelanger on IRC. ........

2010-05-02 02:52 +0000 [r260521]  Eliel C. Sardanons <eliels@gmail.com>

	* main/data.c, include/asterisk/data.h: Avoid making AstData depend
	  on libxml2 to compile. We have some functions inside the AstData
	  API to get the tree in XML form, but it is not required at the
	  moment to compile asterisk and we can disable that part of the
	  API if we don't have libxml2 support.

2010-04-30 22:36 +0000 [r260437]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/sig_analog.h: Merged revisions 260434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
	  | 11 lines Ensure channel state is not incorrectly set in the
	  case of a very early answer. The needringing bit was being read
	  in dahdi_read after answering thereby setting the state to
	  ringing from up. This clears needringing upon answering so that
	  is no longer possible. (closes issue #17067) Reported by: tzafrir
	  Patches: needringing.diff uploaded by tzafrir (license 46)
	  ........

2010-04-30 22:24 +0000 [r260435]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7,
	  and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
	  SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS
	  Also fixed the declaration of pollers[] in mfcr2_monitor(). It
	  was dimensioned to the number of bytes in struct
	  dahdi_mfcr2.pvts[] and not to the same dimension of the struct
	  dahdi_mfcr2.pvts[].

2010-04-30 20:11 +0000 [r260344-260346]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 260345 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri,
	  30 Apr 2010) | 18 lines Fix potential crash from race condition
	  due to accessing channel data without the channel locked. In
	  res_musiconhold.c, there are several places where a channel's
	  stream's existence is checked prior to calling ast_closestream on
	  it. The issue here is that in several cases, the channel was not
	  locked while checking the stream. The result was that if two
	  threads checked the state of the channel's stream at
	  approximately the same time, then there could be a situation
	  where both threads attempt to call ast_closestream on the
	  channel's stream. The result here is that the refcount for the
	  stream would go below 0, resulting in a crash. I have added
	  proper channel locking to res_musiconhold.c to ensure that we do
	  not try to check chan->stream without the channel locked. A
	  Digium customer has been using this patch for several weeks and
	  has not had any crashes since applying the patch. ABE-2147
	  ........

	* apps/app_queue.c: Fix logic reversal error when queue callers
	  join the queue. When a specific position is specified for the
	  queue, the idea was that the caller cannot be placed ahead of
	  higher-priority callers. Unfortunately, the logic was reversed so
	  that the caller could ONLY be placed ahead of higher priority
	  callers. Discovered while writing a unit test.

2010-04-30 06:19 +0000 [r260280-260292]  Tilghman Lesher <tlesher@digium.com>

	* main/strcompat.c: Don't allow file descriptors to go above 64k,
	  when we're closing them in a fork(2). This saves time, when, even
	  though the system allows the process limit to be that high, the
	  practical limit is much lower. Also introduce an additional
	  optimization, in the form of using the CLOEXEC flag to close
	  descriptors at the right time. (closes issue #17223) Reported by:
	  dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
	  tilghman (license 14) Tested by: dbackeberg

	* configs/extensions.conf.sample: Logic fixups for a sample FREENUM
	  dialplan context. (closes issue #17263) Reported by: pprindeville
	  Patches: freenum-dialplan.patch#3 uploaded by pprindeville
	  (license 347)

2010-04-29 22:44 +0000 [r260231]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  260195 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
	  | 26 lines DTMF CallerID detection problems. The code handling
	  DTMF CallerID drops digits on long CallerID numbers and may
	  timeout waiting for the first ring with shorter numbers. The DTMF
	  emulation mode was not turned off when processing DTMF CallerID.
	  When the emulation code gets behind in processing the DTMF digits
	  it can skip a digit. For shorter numbers, the timeout may have
	  been too short. I increased it from 2 seconds to 4 seconds. Four
	  seconds is a typical time between rings for many countries.
	  (closes issue #16460) Reported by: sum Patches: issue16460.patch
	  uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
	  uploaded by rmudgett (license 664) Tested by: sum, rmudgett
	  Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
	  AST-334 JIRA SWP-901 ........

2010-04-29 18:15 +0000 [r260148]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.conf.sample: Pattern match fail.

2010-04-29 15:33 +0000 [r260050]  David Vossel <dvossel@digium.com>

	* /, include/asterisk/audiohook.h, main/audiohook.c: Merged
	  revisions 260049 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
	  | 14 lines Fixes crash in audiohook_write_list The middle_frame
	  in the audiohook_write_list function was being freed if a
	  audiohook manipulator returned a failure. This is incorrect
	  logic. This patch resolves this and adds detailed descriptions of
	  how this function should work and why manipulator failures must
	  be ignored. (closes issue #17052) Reported by: dvossel Tested by:
	  dvossel (closes issue #16196) Reported by: atis Review:
	  https://reviewboard.asterisk.org/r/623/ ........

2010-04-29 00:35 +0000 [r260007]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/extconf.h: Fix comment.

2010-04-28 22:34 +0000 [r259957]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, channels/sip/include/sip.h: Don't override
	  peer context with domain context. (closes issue #17040) Reported
	  by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded
	  by pprindeville (license 347) Tested by: pprindeville Review:
	  https://reviewboard.asterisk.org/r/565/

2010-04-28 21:20 +0000 [r259870]  David Vossel <dvossel@digium.com>

	* main/channel.c, channels/chan_local.c, /: Merged revisions 259858
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
	  | 33 lines resolves deadlocks in chan_local Issue_1. In the
	  local_hangup() 3 locks must be held at the same time... pvt,
	  pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
	  the channel to hangup is the outbound chan_local channel, but
	  when it is not the outbound channel we have an issue... We
	  attempt to do deadlock avoidance only on the tech pvt, when both
	  the tech pvt and the pvt->owner are locked coming into that loop.
	  By never giving up the pvt->owner channel deadlock avoidance is
	  not entirely possible. This patch resolves that by doing deadlock
	  avoidance on both the pvt->owner and the pvt when trying to get
	  the pvt->chan lock. Issue_2. ast_prod() is used in
	  ast_activate_generator() to queue a frame on the channel and make
	  the channel's read function get called. This function is used in
	  ast_activate_generator() while the channel is locked, which
	  mean's the channel will have a lock both from the generator code
	  and the frame_queue code by the time it gets to chan_local.c's
	  local_queue_frame code... local_queue_frame contains some of the
	  same crazy deadlock avoidance that local_hangup requires, and
	  this recursive lock prevents that deadlock avoidance from
	  happening correctly. This patch removes ast_prod() from the
	  channel lock so only one lock is held during the
	  local_queue_frame function. (closes issue #17185) Reported by:
	  schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
	  (license 671) issue_17185_v2.diff uploaded by dvossel (license
	  671) Tested by: schmoozecom, GameGamer43 Review:
	  https://reviewboard.asterisk.org/r/631/ ........

2010-04-28 21:08 +0000 [r259853]  Leif Madsen <lmadsen@digium.com>

	* /, config.guess: Merged revisions 259852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
	  | 6 lines Update config.guess. Updating config.guess because
	  after installing Ubuntu Server 9.10 and running all the update
	  scripts, running ./configure would not continue because it was
	  unable to determine what kind of system I had. After updating
	  config.guess things started working again. ........

2010-04-28 20:32 +0000 [r259760-259848]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac: Merged revisions 259847 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
	  2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
	  systems without install can use install-sh from our source dir.
	  ........

	* /, makeopts.in: Merged revisions 259833 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
	  1 line Missed this when removing $ID ........

	* Makefile, /, configure, configure.ac: Merged revisions 259748 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
	  7 lines Remove usage of `id` since it isn't useful and was
	  causing breakge. Solaris `id` doesn't support the -u argument.
	  Instead of figuring out how to fix this to work on Solaris, I
	  decided to check why it was necessary and where else it was used.
	  It was only used in one place, and it hasn't been needed for a
	  very long time (I question whether it was ever needed). ........

2010-04-28 17:18 +0000 [r259672]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 259664 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28
	  Apr 2010) | 4 lines Do not play goodbye prompt after timeout of
	  message review. ABE-2124 ........

2010-04-27 22:47 +0000 [r259587-259617]  Jason Parker <jparker@digium.com>

	* res/res_agi.c: Fix compile on systems without
	  HAVE_NULLSAFE_PRINTF defined.

	* channels/sip/dialplan_functions.c: Be more explicit about field
	  naming in a test.

2010-04-27 22:18 +0000 [r259538]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 259531 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27
	  Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and
	  vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
	  failed: Success" Changed the warning to "Failed to decode
	  CallerID on channel 'name'". The message before it is likely more
	  specific about why the CallerID decode failed. SWP-501 AST-283
	  ........

2010-04-27 22:11 +0000 [r259533]  Mark Michelson <mmichelson@digium.com>

	* main/ccss.c: Shuffle some casts to make builds on bamboo happier.

2010-04-27 21:49 +0000 [r259527]  Leif Madsen <lmadsen@digium.com>

	* /, sounds/Makefile: Merged revisions 259526 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
	  | 15 lines Update sounds files. * Add additional sounds prompts
	  for say_enumeration * Update the English conference sounds
	  prompts so they are better quality and all sound more consistent
	  * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
	  to include all present sound files Both core (en, fr, es) and
	  extra (en, fr) sounds files have been updated. (closes issue
	  #16200) Reported by: murf (closes issue #17137) Reported by:
	  lmadsen ........

2010-04-27 21:18 +0000 [r259439-259451]  Jason Parker <jparker@digium.com>

	* /: Block 259441 instead of recording it as merged.

	* /: Recorded merge of revisions 259441 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) |
	  1 line Add gar to the check for AR for those silly OSes (Solaris)
	  that don't have ar. ........

	* main/editline/configure, main/editline/Makefile.in,
	  main/editline/configure.in: Add gar to the check for AR for those
	  silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't
	  handle AC_PROG_GREP, so I removed it. This is fine, since we
	  don't need to use anything that the configure script doesn't.

2010-04-27 21:10 +0000 [r259438]  Leif Madsen <lmadsen@digium.com>

	* include/asterisk/doxygen/mantisworkflow.h: Update the Mantis
	  Workflow document in doxygen. (closes issue #17175) Reported by:
	  lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by
	  pabelanger (license 224) Tested by: pabelanger, lmadsen

2010-04-27 19:52 +0000 [r259357]  Mark Michelson <mmichelson@digium.com>

	* main/ccss.c: Change cc_ref and cc_unref from macros to inline
	  functions. The hope is that Solaris won't be as whiny after this
	  change.

2010-04-27 19:31 +0000 [r259353]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac: Merged revisions 259352 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr
	  2010) | 5 lines Support the silly OSes that don't have ar and
	  strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path
	  isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
	  switch to AC_CHECK_TOOLS. ........

2010-04-27 18:29 +0000 [r259229-259307]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
	  revisions 259270 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
	  | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
	  #7321 implements a new chan_dahdi configuration option. However,
	  a change mentioned in the issue was never implemented. This is
	  the change that will allow the feature to work. I added a note to
	  chan_dahdi.conf.sample about the feature. (closes issue #17143)
	  Reported by: djensen99 Patches: diff.txt uploaded by djensen99
	  (license NA) (One line change) Tested by: djensen99 ........

	* channels/chan_dahdi.c: Re-fix dahdi_request() iflist locking
	  since CCSS merged.

2010-04-27 15:25 +0000 [r259189]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/etc_default_asterisk (added): Add missing file
	  (pointed out by TheDavidFactor on #asterisk-dev) referenced by
	  revision 239231.

2010-04-26 21:45 +0000 [r259023-259105]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 259104 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
	  2010) | 3 lines Let compilation succeed warning-free when
	  DONT_OPTIMIZE is turned off. ........

	* main/channel.c, /: Merged revisions 259018 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
	  2010) | 13 lines Prevent Newchannel manager events for dummy
	  channels. No Newchannel manager event will be fired for channels
	  that are allocated to not match a registered technology type.
	  Thus bogus channels allocated solely for variable substitution or
	  CDR operations do not result in a Newchannel event. (closes issue
	  #16957) Reported by: atis Review:
	  https://reviewboard.asterisk.org/r/601 ........

2010-04-26 19:05 +0000 [r258974]  David Ruggles <thedavidfactor@gmail.com>

	* contrib/valgrind.supp: Line 24 missed in compatibility fix in
	  revision 233577 added a "fun:" prefix line 24

2010-04-26 15:59 +0000 [r258934]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_sip.c: Small error in the T.140 RTP port verbose
	  log. (closes issue #16988) Reported by: frawd Patches:
	  chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
	  Tested by: russell

2010-04-26 14:18 +0000 [r258896]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c, include/asterisk/res_fax.h, res/res_fax_spandsp.c:
	  Update res_fax and res_fax_spandsp to be compatible with Fax For
	  Asterisk 1.2. The fax session initilization code for T.38 faxes
	  has been rewritten. T.38 session initialization was removed from
	  generic_fax_exec, and split into two different code paths for
	  receive and send. Also the 'z' option (to send a T.38 reinvite if
	  we do not receive one) was added to sendfax. In the output of
	  'fax show sessions', the 'Type' column has been renamed to 'Tech'
	  and replaced with a new 'Tech' column that will report 'G.711' or
	  'T.38'. Control of ECM defaults has been added to res_fax A 'fax
	  show settings' CLI command has been added. Support of the new
	  AST_T38_REQUEST_PARMS control method request to handle channels
	  that have already received a T.38 reinvite before the FAX
	  application is start has been added. Support for the 'fax show
	  settings' command has been added to res_fax_spandsp and handling
	  of the ECM flag has been slightly altered.

2010-04-25 18:51 +0000 [r258838-258855]  Alexandr Anikin <may@telecom-service.ru>

	* addons/chan_ooh323.c: additional checking related to issue 17186

	* addons/chan_ooh323.c: Don't pass zero length callerid to ooh323
	  stack Don't pass zero callerid string to ooh323 stack because it
	  can't encode this properly and can't generate setup message.
	  (closes issue #17186) Reported by: vmikhelson Patches:
	  zero_callerid_num.patch uploaded by may213 (license 454) Tested
	  by: may213

2010-04-25 18:12 +0000 [r258776]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_monitor.c: Merged revisions 258775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
	  | 6 lines When StopMonitor is called, ensure that it will not be
	  restarted by a channel event. (closes issue #16590) Reported by:
	  kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
	  (license 888) ........

2010-04-22 22:19 +0000 [r258685]  Jason Parker <jparker@digium.com>

	* utils/extconf.c: Add another random function that does nothing to
	  make the utils/ dir happy.

2010-04-22 22:11 +0000 [r258675]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c: Fix previous commit.

2010-04-22 22:10 +0000 [r258673-258674]  Jason Parker <jparker@digium.com>

	* utils/Makefile, utils/extconf.c: Make utils/ stuff *actually*
	  compile this time.

	* utils/Makefile, utils/extconf.c: Let utils/ dir compile when
	  DEBUG_THREADS is not enabled.

2010-04-22 21:57 +0000 [r258671]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
	  193391,258670 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
	  2009) | 8 lines Set the proper disposition on originated calls.
	  (closes issue #14167) Reported by: jpt Patches:
	  call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
	  Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
	  mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
	  lines Fix broken CDR behavior. This change allows a CDR record
	  previously marked with disposition ANSWERED to be set as BUSY or
	  NO ANSWER. Additionally this change partially reverts r235635 and
	  does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
	  from ast_call(). To preserve proper CDR behavior, the
	  AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
	  ast_bridge_call(). (closes issue #16797) Reported by:
	  VarnishedOtter Tested by: mnicholson ........ (closes issue
	  #16222) Reported by: telles Tested by: mnicholson

2010-04-22 21:06 +0000 [r258632]  Russell Bryant <russell@digium.com>

	* tests/test_event.c, main/event.c: Add ast_event subscription unit
	  test and fix some ast_event API bugs. This patch introduces
	  another test in test_event.c that exercises most of the
	  subscription related ast_event API calls. I made some minor
	  additions to the existing event allocation test to increase API
	  coverage by the test code. Finally, I made a list in a comment of
	  API calls not yet touched by the test module as a to-do list for
	  future test development. During the development of this test
	  code, I discovered a number of bugs in the event API. 1)
	  subscriptions to AST_EVENT_ALL were not handled appropriately in
	  a couple of different places. The API allows a subscription to
	  all event types, but with IE parameters, just as if it was a
	  subscription to a specific event type. However, the parameters
	  were being ignored. This affected ast_event_check_subscriber()
	  and event distribution to subscribers. 2) Some of the logic in
	  ast_event_check_subscriber() for checking subscriptions against
	  query parameters was wrong. Review:
	  https://reviewboard.asterisk.org/r/617/

2010-04-22 20:04 +0000 [r258595]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_voicemail.c: Pass interactive = 0 and fix a compile
	  error.

2010-04-22 19:08 +0000 [r258557]  Jason Parker <jparker@digium.com>

	* main/lock.c (added), include/asterisk/res_odbc.h,
	  include/asterisk/astobj2.h, main/heap.c, include/asterisk/lock.h,
	  main/astobj2.c, res/res_odbc.c, include/asterisk/heap.h: Remove
	  ABI differences that occured when compiling with DEBUG_THREADS.
	  "Bad Things" would happen if Asterisk was compiled with
	  DEBUG_THREADS, but a loaded module was not (or vice versa). This
	  also immensely simplifies the lock code, since there are no
	  longer 2 separate versions of them. Review:
	  https://reviewboard.asterisk.org/r/508/

2010-04-22 18:07 +0000 [r258517]  Eliel C. Sardanons <eliels@gmail.com>

	* doc/manager_1_1.txt, main/channel.c, include/asterisk/doxyref.h,
	  include/asterisk/xml.h, main/data.c (added), main/xml.c,
	  include/asterisk/channel.h, include/asterisk/_private.h,
	  include/asterisk/data.h (added), CHANGES, apps/app_queue.c,
	  main/asterisk.c, apps/app_voicemail.c: Asterisk data retrieval
	  API. This module implements an abstraction for retrieving and
	  exporting asterisk data. Developed by: Brett Bryant
	  <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY)
	  <eliels@gmail.com> For the Google Summer of code 2009 Project.
	  Documentation can be found in doxygen format and inside the
	  header include/asterisk/data.h Review:
	  https://reviewboard.asterisk.org/r/275/

2010-04-22 17:36 +0000 [r258515]  Russell Bryant <russell@digium.com>

	* doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019.

2010-04-21 21:56 +0000 [r258433]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 258432 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21
	  Apr 2010) | 8 lines Fix looping forever when no input received in
	  certain voicemail menu scenarios. Specifically, prompting for an
	  extension (when leaving or forwarding a message) or when
	  prompting for a digit (when saving a message or changing
	  folders). ABE-2122 SWP-1268 ........

2010-04-21 19:45 +0000 [r258351-258387]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/asterisk.tex: Missed this when reverting the bad version
	  change in asterisk.tex.

	* doc/tex/asterisk.tex: Fix change in asterisk.tex that got merged
	  in after testing. (issue #17220)

	* Makefile, doc/tex/security-events.tex, configure,
	  include/asterisk/autoconfig.h.in, doc/tex/Makefile, configure.ac,
	  doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
	  build_tools/prep_tarball, doc/tex/localchannel.tex,
	  doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex,
	  doc/tex/cel-doc.tex: Add ability to generate ASCII documentation
	  from the TeX files. These changes add the ability to run 'make
	  asterisk.txt' just like the existing 'make asterisk.pdf' commands
	  to generate a text document from the TeX files we have in the
	  doc/tex/ directory. I've also updated a few of the .tex files
	  because they weren't properly escaping certain characters so they
	  would show up as Unicode characters (like [U+021C]). Made changes
	  to the configure scripts so it would detect the catdvi program
	  which is required to convert the .dvi file generated by latex.
	  I've also added a few lines to the build_tools/prep_tarball
	  script so that the text documentation gets generated and added to
	  future tarballs of Asterisk releases. (closes issue #17220)
	  Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
	  lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
	  (license 224) Tested by: lmadsen, pabelanger

2010-04-21 19:07 +0000 [r258345]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_callcompletion.c: Add small documentation update to
	  func_callcompletion.c. This directs users to documents which can
	  help explain the concepts and configuration options settable with
	  the function.

2010-04-21 19:02 +0000 [r258344]  Leif Madsen <lmadsen@digium.com>

	* UPGRADE.txt, CHANGES, channels/chan_iax2.c: IAXpeers output now
	  matches SIPpeers format for manager (AMI). (closes issue #17100)
	  Reported by: secesh Tested by: pabelanger Review:
	  https://reviewboard.asterisk.org/r/594/

2010-04-21 18:13 +0000 [r258305]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes issue with double "sip:" in header
	  field This is a clear mistake in logic. Future discussions about
	  how to avoid having to handle uri's like this should take place
	  in the future, but this fix needs to go in for now. (closes issue
	  #15847) Reported by: ebroad Patches: doublesip.patch uploaded by
	  ebroad (license 878)

2010-04-21 13:26 +0000 [r258265]  Leif Madsen <lmadsen@digium.com>

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  res/res_calendar_caldav.c: Fix the \brief description in the
	  res_calendar_*.c files.

2010-04-21 13:24 +0000 [r258190-258256]  Julian Lyndon-Smith <julian@dotr.com>

	* doc/manager_1_1.txt: fix whitespace issue

	* doc/manager_1_1.txt, doc/tex/manager.tex: Added NEW ACTIONS entry
	  for new MixMonitorMute AMI command. Added State and Direction
	  variables for new MixMonitorMute AMI command.

	* CHANGES: Added CHANGES entry for new MixMonitorMute AMI command.

	* main/frame.c, include/asterisk/audiohook.h, main/audiohook.c,
	  include/asterisk/frame.h, apps/app_mixmonitor.c,
	  res/res_mutestream.c: Added MixMonitorMute manager command Added
	  a new manager command to mute/unmute MixMonitor audio on a
	  channel. Added a new feature to audiohooks so that you can mute
	  either read / write (or both) types of frames - this allows for
	  MixMonitor to mute either side of the conversation without
	  affecting the conversation itself. (closes issue #16740) Reported
	  by: jmls Review: https://reviewboard.asterisk.org/r/487/

2010-04-20 19:02 +0000 [r258106-258149]  Leif Madsen <lmadsen@digium.com>

	* configs/cli_aliases.conf.sample: Add 'soft hangup' alias per
	  Steve Johnson on asterisk-users.

	* configs/extensions.conf.sample: Add example dialplan for dialing
	  ISN numbers (http://www.freenum.org). Minor tweaks and
	  documentation added by me. (closes issue #17058) Reported by:
	  pprindeville Patches: freenum.patch#5 uploaded by pprindeville
	  (license 347) Tested by: lmadsen

	* contrib/scripts/sip-friends.sql: Add missing 'useragent' field to
	  sip-friends.sql file. (closes issue #17171) Reported by: thehar
	  Patches: sip-friends.patch uploaded by thehar (license 831)
	  Tested by: pabelanger, thehar

2010-04-20 17:06 +0000 [r258065]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 258029 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20
	  Apr 2010) | 11 lines Play correct prompt when voicemail store
	  failure occurs after attempted forward. If a user's mailbox was
	  full and a message was attempted to be forwarded to said box,
	  warnings on the console would indicate failure. However, the
	  played prompt was that of success (vm-msgsaved). Now storage
	  failure is taken into account and the correct prompt
	  (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262
	  ........

2010-04-20 12:38 +0000 [r257988]  Leif Madsen <lmadsen@digium.com>

	* formats/format_pcm.c: Update supported file extensions in
	  doxygen. Updated the doxygen \arg line after looking at the file
	  for some other Asterisk documentation and noticing they weren't
	  up to date. Thanks to seanbright for looking at the code for me
	  :)

2010-04-19 21:57 +0000 [r257947-257949]  Jason Parker <jparker@digium.com>

	* main/indications.c: Change log message to match severity.

	* main/indications.c: Don't consider a missing indications.conf to
	  be a critical error. There were many changes in revision 176627
	  which would avoid the error that a missing config would have
	  caused. Other than this, there are no other config files
	  (including asterisk.conf, surprisingly) that are required.

2010-04-19 19:23 +0000 [r257883]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Bad merge fix

2010-04-19 18:42 +0000 [r257851]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_srv.c: Commit compromise I suggested on review 608.
	  This allows for multiple SRV queries to be done from the dialplan
	  for the same service on a single call while still allowing one to
	  bypass the call to SRVQUERY if they so please. Taking action
	  since no comments had been left for a while. This can easily be
	  reverted if needed. External tests still pass.

2010-04-19 17:57 +0000 [r257810]  Terry Wilson <twilson@digium.com>

	* main/features.c: Fix incomplete CDR merge from r195881 Because
	  res/res_features.c was removed and main/cdr.c added, these
	  changes didn't make it to trunk and the 1.6.x branches

2010-04-18 17:25 +0000 [r257768]  Tilghman Lesher <tlesher@digium.com>

	* configs/cdr_odbc.conf.sample: Removing unused configuration
	  parameters

2010-04-16 21:22 +0000 [r257713]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* /, apps/app_mixmonitor.c: Merged revisions 257686 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16
	  Apr 2010) | 21 lines Make the mixmonitor thread process audio
	  frames faster Mantis issue 17078 reports MixMonitor recordings
	  have shorter durations than the call duration. This was because
	  the mixmonitor thread was not processing frames from the
	  audiohook fast enough. The mixmonitor thread would slowly fall
	  behind the most recent audio frame and when the channel hangs up,
	  the mixmonitor thread would exit without processing the same
	  number of frames as the channel; leaving the mixmonitor recording
	  shorter than actual call duration. This revision fixes this issue
	  by moving the ast_audiohook_trigger_wait() and the subsequent
	  audiohook.status check into the block where the
	  ast_audiohook_read_frame() function returns NULL. (closes issue
	  #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
	  by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
	  https://reviewboard.asterisk.org/r/611/ ........

2010-04-16 19:50 +0000 [r257646]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Make sure to fail a monitor if we receive a
	  negative response for a CC SUBSCRIBE.

2010-04-16 19:25 +0000 [r257642]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* channels/chan_dahdi.c: Enable PRI SERVICE message support in
	  chan_dahdi for the 'national' switchtype Revision 1072 of libpri
	  added SERVICE message support for the 'national' switchtype. The
	  attached patch enables the use of 'pri service' CLI commands on
	  dahdi channels that are configured for the 'national' switchtype.
	  (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch
	  uploaded by dhubbard (license 733) Tested by: elguero, dhubbard
	  Review: https://reviewboard.asterisk.org/r/612/

2010-04-15 21:26 +0000 [r257493-257560]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, /, tests/test_app.c, main/app.c: Merged
	  revisions 257544 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
	  | 6 lines Allow application options with arguments to contain
	  parentheses, through a variety of escaping techniques. Fixes
	  SWP-1194 (ABE-2143). Review:
	  https://reviewboard.asterisk.org/r/604/ ........

	* /, channels/chan_sip.c: Merged revisions 257467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
	  | 13 lines Don't recreate peer, when responding to a repeated
	  deregistration attempt. When a reply to a deregistration is lost
	  in transmit, the client retries the deregistration. Previously,
	  this would cause a realtime/autocreate peer to be loaded back
	  into memory, after it had already been correctly purged. Instead,
	  we just want to resend the reply without loading the peer.
	  (closes issue #16908) Reported by: kkm Patches:
	  20100412__issue16908.diff.txt uploaded by tilghman (license 14)
	  Tested by: kkm ........

2010-04-15 19:41 +0000 [r257343-257427]  Leif Madsen <lmadsen@digium.com>

	* /, doc/backtrace.txt: Merged revisions 257426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
	  | 13 lines Update backtrace.txt documentation. Update the
	  backtrace.txt documentation so it conforms to the same layout as
	  other documents we've been working on recently. Additionally, add
	  a bunch of new information about gathering backtraces for crashes
	  and deadlocks, along with ways of verifying your file before
	  uploading it. Create a couple of one line commands for people to
	  generate the files we need. (closes issue #17190) Reported by:
	  lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
	  (license 10) Tested by: lmadsen, pabelanger ........

	* /, doc/backtrace.txt: Merged revisions 257342 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
	  | 1 line Update address of the bug tracker. ........

2010-04-14 22:57 +0000 [r257262]  Tilghman Lesher <tlesher@digium.com>

	* main/features.c, configs/features.conf.sample: Yet another issue
	  where the conversion of the application delimiter to comma caused
	  an issue. Application arguments within the feature map could
	  possibly contain a comma, which conflicts with the syntax of the
	  features.conf configuration file. This patch allows the argument
	  to be wrapped in parentheses or quoted, to allow the application
	  arguments to be interpreted as a single configuration parameter.
	  (closes issue #16646) Reported by: pinga-fogo Patches:
	  20100414__issue16646.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman Review:
	  https://reviewboard.asterisk.org/r/547/

2010-04-13 19:17 +0000 [r257191]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Also unref the pvt when we delete the
	  provisional keepalive job. (closes issue #16774) Reported by:
	  kowalma Patches: 20100315__issue16774.diff.txt uploaded by
	  tilghman (license 14) Tested by: falves11, jamicque Review:
	  https://reviewboard.asterisk.org/r/591/

2010-04-13 18:10 +0000 [r257146]  Matthew Nicholson <mnicholson@digium.com>

	* main/manager.c, /, configs/manager.conf.sample: Merged revisions
	  257070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
	  2010) | 9 lines Add an option to restore past broken behavor of
	  the Events manager action Before r238915, certain values for the
	  EventMask parameter of the Events action would result in no
	  response being returned. This patch adds an option to restore
	  that broken behavior. Also while fixing this bug I discovered
	  that passing an empty EventMasks parameter would also result in
	  no response being returned, this has been fixed as well while
	  being preserved when the broken behavior is requested. (closes
	  issue #17023) Reported by: nblasgen Review:
	  https://reviewboard.asterisk.org/r/602/ ........

2010-04-13 16:33 +0000 [r257065]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_sqlite3_custom.c: Ensure that we can have commas within
	  cdr values. (closes issue #17001) Reported by: snuffy Patches:
	  20100412__issue17001.diff.txt uploaded by tilghman (license 14)
	  Tested by: snuffy

2010-04-13 16:18 +0000 [r256985-257032]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample: Update sample dialstrings in
	  sip.conf.sample file.

	* funcs/func_srv.c: Address Russell's comments on func_srv from
	  reviewboard. * Change copyright date * Place channel in
	  autoservice when doing SRV lookup * Get rid of trailing
	  whitespace * Change logic in load_module function

	* main/ccss.c: Fix issue where recall would not happen when it
	  should. Specifically, the situation would happen when multiple
	  callers would request CC for a single generically-monitored
	  device. If the monitored device became available but the caller
	  did not answer the recall, then there was nothing that would poke
	  the CC core to let it know that it should attempt to recall
	  someone else instead. After careful consideration, I came to the
	  conclusion that the only area of Asterisk that needed to be
	  touched was the generic CC monitor. All other types of CC would
	  require something outside of Asterisk to invoke a recall for a
	  separate device. This was accomplished by changing the generic
	  monitor destructor to poke other generic monitor instances if the
	  device is currently available and the specific instance was
	  currently not suspended. In order to not accidentally trigger
	  recalls at bad times, the fit_for_recall flag was also added to
	  the generic_monitor_instance_list struct. This gets set as soon
	  as a monitored device becomes available. It gets cleared if a
	  CCNR request triggers the creation of a new generic monitor
	  instance. By doing this, we don't accidentally try to recall a
	  device when the monitored device was being monitored for CCNR and
	  never actually became available for recall in the first place.
	  This error was discovered by Steve Pitts during in-house testing
	  at Digium.

2010-04-12 17:29 +0000 [r256860-256901]  Leif Madsen <lmadsen@digium.com>

	* /, doc/HOWTO_collect_debug_information.txt (added): Merged
	  revisions 256900 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
	  | 15 lines Add How-To document on collecting debugging info for
	  issues.asterisk.org Paul Belanger has been helping a lot with bug
	  tracking recently and created this document that we can now point
	  to when additional debugging information is required. This
	  document will help those filing issues to know how to get the
	  information required when filing their issues. This will make
	  things easier on the developers. Initial text and changes by
	  pabelanger. Tweaks and editing by myself. (closes issue #17159)
	  Reported by: pabelanger Patches:
	  HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
	  (license 10) Tested by: tzafrir, pabelanger, lmadsen ........

	* apps/app_voicemail.c: Remove silly debug message that is not
	  useful. (issue #17159)

2010-04-12 14:47 +0000 [r256823]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: gives channel reference before unlocking it
	  and using setvar helper. To guarantee the channel is valid when
	  calling setvar on the MASTER_CHANNEL dialplan function, a channel
	  reference must be taken before unlocking. Thanks to russell for
	  pointing out the error.

2010-04-12 14:39 +0000 [r256821]  Leif Madsen <lmadsen@digium.com>

	* main/logger.c: CLI command logger set level auto complete. A
	  simple patch to enable auto tab complete. (closes issue #17152)
	  Reported by: pabelanger Patches: 0017152.patch uploaded by
	  pabelanger (license 224)

2010-04-12 02:19 +0000 [r256745-256783]  Russell Bryant <russell@digium.com>

	* tests/test_substitution.c: test_substitution expects func_curl to
	  be present to work.

	* tests/test_pbx.c: Add ASTERISK_FILE_VERSION() macro

2010-04-10 08:33 +0000 [r256704]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/safe_asterisk.8, doc/asterisk.8,
	  contrib/scripts/autosupport.8, contrib/scripts/astgenkey.8: fix
	  hyphen vs. minus in man pages In troff '-' is used for a hyphen.
	  A minus is denoted by '\-' . This is normally also used for a
	  dash. This patch converts all '-'-s that are minuses or dashes to
	  '\-'.

2010-04-09 22:20 +0000 [r256646-256661]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, main/ccss.c: Remove status_response
	  callbacks where they are not needed.

	* channels/chan_local.c: Prevent crash when originating a call to a
	  local channel. Call completion code tries to grab the call
	  completion parameters from the requesting channel during
	  local_request. When originating a call to a local channel,
	  however, this channel is NULL. This was causing an issue for me
	  when trying to run a test script.

2010-04-09 19:46 +0000 [r256569-256608]  Richard Mudgett <rmudgett@digium.com>

	* doc/CCSS_architecture.pdf (added): Merge CCSS architecture
	  document from CCSS branch.

	* channels/sig_pri.h, configure, include/asterisk/autoconfig.h.in:
	  Remove PRI CCSS BUGBUG message and update configure script.

2010-04-09 16:04 +0000 [r256485-256530]  Mark Michelson <mmichelson@digium.com>

	* channels/sip/reqresp_parser.c, channels/sip/include/sip.h,
	  channels/sip/include/reqresp_parser.h: Add routines for parsing
	  SIP URIs consistently. From the original issue report opened by
	  Nick Lewis: Many sip headers in many sip methods contain the ABNF
	  structure name-andor-addr = name-addr / addr-spec Examples
	  include the to-header, from-header, contact-header,
	  replyto-header At the moment chan_sip.c makes various different
	  attempts to parse this name-andor-addr structure for each header
	  type and for each sip method with sometimes limited degrees of
	  success. I recommend that this name-andor-addr structure be
	  parsed by a dedicated function and that it be used irrespective
	  of the specific method or header that contains the
	  name-andor-addr structure Nick has also included unit tests for
	  verifying these routines as well, so...heck yeah. (closes issue
	  #16708) Reported by: Nick_Lewis Patches:
	  reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis
	  (license 657 Review: https://reviewboard.asterisk.org/r/549

	* channels/chan_sip.c, tests/test_gosub.c, funcs/func_srv.c: Fix
	  some compiler errors that popped up after the CCSS merge.

	* apps/app_dial.c, configs/chan_dahdi.conf.sample,
	  include/asterisk/devicestate.h, include/asterisk/xml.h,
	  channels/chan_local.c, doc/tex/ccss.tex (added), main/ccss.c
	  (added), channels/chan_sip.c, configure.ac, main/xml.c,
	  include/asterisk/channel.h, configs/manager.conf.sample,
	  include/asterisk/channelstate.h (added),
	  include/asterisk/manager.h, CHANGES, channels/sig_pri.c,
	  channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
	  main/manager.c, funcs/func_callcompletion.c (added),
	  channels/sig_analog.c, channels/sig_analog.h,
	  configs/ccss.conf.sample (added), include/asterisk/rtp_engine.h,
	  include/asterisk/frame.h, include/asterisk/ccss.h (added),
	  doc/tex/asterisk.tex, main/asterisk.c,
	  channels/sip/include/sip.h: Merge Call completion support into
	  trunk. From Reviewboard: CCSS stands for Call Completion
	  Supplementary Services. An admittedly out-of-date overview of the
	  architecture can be found in the file doc/CCSS_architecture.pdf
	  in the CCSS branch. Off the top of my head, the big differences
	  between what is implemented and what is in the document are as
	  follows: 1. We did not end up modifying the Hangup application at
	  all. 2. The document states that a single call completion monitor
	  may be used across multiple calls to the same device. This proved
	  to not be such a good idea when implementing protocol-specific
	  monitors, and so we ended up using one monitor per-device
	  per-call. 3. There are some configuration options which were
	  conceived after the document was written. These are documented in
	  the ccss.conf.sample that is on this review request. For some
	  basic understanding of terminology used throughout this code, see
	  the ccss.tex document that is on this review. This implements
	  CCBS and CCNR in several flavors. First up is a "generic"
	  implementation, which can work over any channel technology
	  provided that the channel technology can accurately report device
	  state. Call completion is requested using the dialplan
	  application CallCompletionRequest and can be canceled using
	  CallCompletionCancel. Device state subscriptions are used in
	  order to monitor the state of called parties. Next, there is a
	  SIP-specific implementation of call completion. This method uses
	  the methods outlined in draft-ietf-bliss-call-completion-06 to
	  implement call completion using SIP signaling. There are a few
	  things to note here: * The agent/monitor terminology used
	  throughout Asterisk sometimes is the reverse of what is defined
	  in the referenced draft. * Implementation of the draft required
	  support for SIP PUBLISH. I attempted to write this in a
	  generic-enough fashion such that if someone were to want to write
	  PUBLISH support for other event packages, such as dialog-state or
	  presence, most of the effort would be in writing callbacks
	  specific to the event package. * A subportion of supporting
	  PUBLISH reception was that we had to implement a PIDF parser. The
	  PIDF support added is a bit minimal. I first wrote a validation
	  routine to ensure that the PIDF document is formatted properly.
	  The rest of the PIDF reading is done in-line in the
	  call-completion-specific PUBLISH-handling code. In other words,
	  while there is PIDF support here, it is not in any state where it
	  could easily be applied to other event packages as is. Finally,
	  there are a variety of ISDN-related call completion protocols
	  supported. These were written by Richard Mudgett, and as such I
	  can't really say much about their implementation. There are notes
	  in the CHANGES file that indicate the ISDN protocols over which
	  call completion is supported. Review:
	  https://reviewboard.asterisk.org/r/523

	* main/srv.c, channels/chan_sip.c, funcs/func_srv.c (added),
	  CHANGES, include/asterisk/srv.h: func_srv and explicit
	  specification of a remote IP for SIP. From Review Board: There
	  are two interrelated changes here. First, there is the
	  introduction of func_srv. This adds two new read-only dialplan
	  functions, SRVQUERY and SRVRESULT. They work very similarly to
	  the ENUMQUERY and ENUMRESULT functions, except that this allows
	  one to query SRV records instead. In order to facilitate this
	  work, I added a couple of new API calls to srv.h.
	  ast_srv_get_record_count tells the number of records returned by
	  an SRV lookup. This number is calculated at the time of the SRV
	  lookup. ast_srv_get_nth_record allows one to get a numbered SRV
	  record. Second, there is the modification to chan_sip that allows
	  one to specify a hostname or IP address (along with a port) to
	  send an outgoing INVITE to when dialing a SIP peer. This goes
	  hand-in-hand with func_srv. You can query SRV records and then
	  use the host and port from the results to dial via a specific
	  host instead of what is configured in sip.conf. Review:
	  https://reviewboard.asterisk.org/r/608 SWP-1200

2010-04-08 16:35 +0000 [r256428]  Kevin P. Fleming <kpfleming@digium.com>

	* /, Makefile.rules, build_tools/make_linker_version_script: Ensure
	  that linker version scripts (used for symbol export control)
	  always exist. Using wildcard matching in the Makefile is not
	  adequate to determine whether an export file should exist for a
	  module or not, so instead we'll just create one if the module
	  needs one, or copy the default one if it does not.

2010-04-06 19:28 +0000 [r256370]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Mac OS X does not support comparing a
	  mutex to its initializer. Create a test for this.

2010-04-06 14:42 +0000 [r256319]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes deadlock in chan_sip caused by usage
	  of MASTER_CHANNEL dialplan function (closes issue #16767)
	  Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by
	  dvossel (license 671) Review:
	  https://reviewboard.asterisk.org/r/606/

2010-04-06 00:39 +0000 [r256265]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 256225 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05
	  Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not
	  protected by PRI lock. SWP-1231 ABE-2163 ........

2010-04-05 15:14 +0000 [r256161]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/localchannel.tex: Fix for localchannel.tex to allow PDFs
	  to be generated again.

2010-04-03 02:12 +0000 [r256103-256104]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
	  include/asterisk/channel.h, main/cel.c, channels/sig_pri.c,
	  channels/chan_iax2.c, apps/app_queue.c, channels/chan_oss.c,
	  funcs/func_redirecting.c, main/channel.c, main/dial.c,
	  channels/chan_dahdi.c, channels/chan_misdn.c,
	  apps/app_dumpchan.c, res/res_agi.c, channels/chan_h323.c,
	  res/snmp/agent.c, apps/app_amd.c, funcs/func_callerid.c:
	  Consolidate ast_channel.cid.cid_rdnis into
	  ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure
	  chan_local.c:local_call() will not leak cid.cid_dnid when
	  copying.

	* apps/app_dial.c: Using the Dial application f option when the
	  call is forwarded will likely crash. Fix app_dial.c:do_forward()
	  OPT_FORCECLID setting cid.cid_num with a stack allocated string
	  instead of a heap allocated string.

2010-04-02 23:55 +0000 [r256010-256019]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Export MEETMEBOOKID and fix pin-less
	  conferences with realtime conferences (closes issue #16866)
	  Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA
	  (license 3) Tested by: DEA Review:
	  https://reviewboard.asterisk.org/r/582/

	* channels/chan_local.c, /: Merged revisions 256014 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02
	  Apr 2010) | 9 lines Resolve a deadlock that occurs due to a
	  pointless call to ast_bridged_channel() (closes issue #16840)
	  Reported by: bzing2 Patches: patch.txt uploaded by bzing2
	  (license 902) issue_16840.rev1.diff uploaded by russell (license
	  2) Tested by: bzing2, russell ........

	* main/channel.c, /: Merged revisions 256009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
	  | 2 lines Remove extremely verbose debug message. ........

2010-04-02 20:19 +0000 [r255952]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Pass the PID of the Asterisk process, not the
	  PID of the canary. (closes issue #17065) Reported by:
	  globalnetinc Patches: astcanary.patch uploaded by makoto (license
	  38) Tested by: frawd, globalnetinc

2010-04-02 18:57 +0000 [r255906]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_ael_share.exports.in (added), codecs,
	  res/res_pktccops.exports.in (added), utils,
	  res/res_monitor.exports.in (added), Makefile.moddir_rules,
	  res/res_smdi.exports.in (added), Makefile.rules, cdr,
	  res/res_agi.exports.in (added), formats, main/asterisk.exports
	  (removed), res/res_odbc.exports (removed),
	  res/res_calendar.exports (removed), apps/app_voicemail.exports
	  (removed), bridges, res/res_odbc.exports.in (added),
	  main/asterisk.exports.in (added), apps/app_voicemail.exports.in
	  (added), res/res_calendar.exports.in (added),
	  res/res_features.exports (removed), res/res_fax.exports.in
	  (added), pbx, res/res_adsi.exports.in (added),
	  res/res_jabber.exports (removed), res/res_pktccops.exports
	  (removed), channels, res/res_jabber.exports.in (added),
	  main/Makefile, res/res_smdi.exports (removed), tests, apps, cel,
	  res/res_agi.exports (removed), addons, res/res_speech.exports
	  (removed), Makefile, funcs, res/res_speech.exports.in (added),
	  res/res_fax.exports (removed), main, res/res_adsi.exports
	  (removed), res/res_features.exports.in (added),
	  res/res_ael_share.exports (removed),
	  build_tools/make_linker_version_script (added), res,
	  res/res_monitor.exports (removed): Allow symbol export filtering
	  to work properly on platforms that have symbol prefixes. Some
	  platforms prefix externally-visible symbols in object files
	  generated from C sources (most commonly, '_' is the prefix). On
	  these platforms, the existing symbol export filtering process
	  ends up suppressing all the symbols that are supposed to be left
	  visible. This patch allows the prefix string to be supplied to
	  the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and
	  then generates the linker scripts as required to include the
	  prefix supplied.

2010-04-02 06:45 +0000 [r255850-255851]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Ignore Redial softkey when no previous
	  dialed number is known (closes issue #17126) Reported by: wedhorn
	  Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30)

	* channels/chan_skinny.c: Cleanup transmit_* functions Bulk lot of
	  generally trivial changes for cleaning up the transmit stuff.
	  Line state request has been modified for line only responses.
	  (closes issue #16994) Reported by: wedhorn Patches:
	  skinny-clean07.diff uploaded by wedhorn (license 30) Tested by:
	  wedhorn

2010-04-01 18:16 +0000 [r255796]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix DEBUG_THREADS build on Darwin.
	  (closes issue #16828) Reported by: oej Patches:
	  20100331__issue16828.diff.txt uploaded by tilghman (license 14)

2010-04-01 16:09 +0000 [r255751]  Matthew Nicholson <mnicholson@digium.com>

	* configs/sip.conf.sample: Removed documentation of the non
	  existent 'both' option to 'faxdetect' in sip.conf

2010-03-31 22:35 +0000 [r255701]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix improper comaparison of anonymous URI
	  when getting P-Asserted-Identity. There was a bug where we split
	  the URI on the @ sign and then attempted to compare to
	  "anonymous@anonymous.invalid" afterwards. This comparison could
	  never evaluate true. So now we keep a copy of the URI prior to
	  the split so that the comparison is valid.

2010-03-31 19:13 +0000 [r255592]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Recorded merge of revisions 255591 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
	  | 15 lines Ensure line terminators in email are consistent. Fixes
	  an issue with certain Mail Transport Agents, where attachments
	  are not interpreted correctly. (closes issue #16557) Reported by:
	  jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
	  tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
	  uploaded by tilghman (license 14)
	  20100308__issue16557__trunk.diff.txt uploaded by tilghman
	  (license 14) Tested by: ebroad, zktech Reviewboard:
	  https://reviewboard.asterisk.org/r/544/ ........

2010-03-31 17:48 +0000 [r255504]  Leif Madsen <lmadsen@digium.com>

	* apps/app_dial.c, /, configs/sip.conf.sample: Add documentation
	  clarifying when 't' and 'T' can be used. (closes issue #17021)
	  Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad

2010-03-30 20:56 +0000 [r255323-255410]  Russell Bryant <russell@digium.com>

	* /, channels/chan_h323.c: Merged revisions 255409 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
	  Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
	  not start. ........

	* /, pbx/pbx_dundi.c: Merged revisions 255322 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
	  | 2 lines Don't make Asterisk not start if pbx_dundi fails to
	  initialize. ........

2010-03-29 14:07 +0000 [r255281]  Jared Smith <jaredsmith@jaredsmith.net>

	* apps/app_confbridge.c, CHANGES: This patch adds custom device
	  state handling for ConfBridge conferences, matching the devstate
	  handling of the MeetMe conferences. Review:
	  https://reviewboard.asterisk.org/r/572/ Closes issue #16972

2010-03-29 05:10 +0000 [r255240]  Russell Bryant <russell@digium.com>

	* main/event.c: Remove a debugging log entry.

2010-03-27 23:51 +0000 [r255199]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
	  addons/chan_ooh323.c, addons/ooh323c/src/ooh323.h,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c:
	  corrections in gk interface, small fixes in call clearing.

2010-03-27 14:44 +0000 [r255158]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: We need to inclde sys/wait.h on OpenBSD to
	  get WEXITSTATUS.

2010-03-27 06:09 +0000 [r255117]  Tilghman Lesher <tlesher@digium.com>

	* pbx/pbx_spool.c: inotify support for pbx_spool This should give a
	  good speed boost, in that one particular thread isn't waking up
	  once a second to read directory contents. Reviewboard:
	  https://reviewboard.asterisk.org/r/137/

2010-03-26 19:27 +0000 [r255021-255066]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Replace some documentation from 1.6.x
	  back into trunk. This documentation associated wth tlsbindaddr is
	  still useful so lets synchronize it between trunk and 1.6.x
	  branches. (issue #17054)

	* configs/sip.conf.sample: Update confusing documentation for
	  tlsbindaddr. Update some confusing documentation for the
	  tlsbindaddr option in sip.conf.sample. Point at a link instead
	  which has better documentation. (closes issue #17054) Reported
	  by: klaus3000

2010-03-26 16:27 +0000 [r254976]  Sean Bright <sean@malleable.com>

	* contrib/scripts/live_ast: Work around a bug in dash on Ubuntu by
	  checking the number of arguments before shift'ing. Reported and
	  tested by pabelanger.

2010-03-25 23:38 +0000 [r254931]  Kevin P. Fleming <kpfleming@digium.com>

	* addons/chan_ooh323.h, addons/ooh323c/src/ooasn1.h,
	  addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
	  addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/dlist.c, addons/ooh323c/src/eventHandler.c,
	  addons/ooh323c/src/ooCapability.c, addons/ooh323cDriver.c,
	  addons/mp3/interface.c, addons/ooh323cDriver.h,
	  addons/ooh323c/src/rtctype.c, addons/ooh323c/src/ooCalls.c,
	  addons/ooh323c/src/encode.c, addons/ooh323c/src/ooUtils.c,
	  addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooh323ep.c,
	  addons/ooh323c/src/ooports.c, addons/mp3/decode_ntom.c,
	  addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
	  addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
	  addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
	  addons/ooh323c/src/perutil.c, addons/mp3/layer3.c,
	  addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooCmdChannel.c,
	  addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ootrace.c: Use "local" instead of "system"
	  header file inclusion. Now that these files are in the tree, they
	  should prefer the tree's local copy of all Asterisk headers over
	  any that may be installed.

2010-03-25 21:39 +0000 [r254884]  Russell Bryant <russell@digium.com>

	* addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooSocket.h: Fix
	  a number of other build problems on Mac OS X.

2010-03-25 20:41 +0000 [r254802]  Jason Parker <jparker@digium.com>

	* utils/Makefile, /: Merged revisions 254800 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
	  1 line Don't remove local copies of utils in uninstall. ........

2010-03-25 20:41 +0000 [r254718-254801]  Russell Bryant <russell@digium.com>

	* addons/chan_ooh323.h: Resolve compiler warning on FreeBSD.

	* addons/ooh323c/src/ooh323.c, addons/Makefile,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c: Fix
	  chan_ooh323 so it works on Mac OS X, as well.

	* channels/chan_usbradio.c: chan_usbradio depends on alsa.

2010-03-25 18:38 +0000 [r254636-254638]  Kevin P. Fleming <kpfleming@digium.com>

	* .cleancount: Bump cleancount due to ast_channel change.

	* include/asterisk/channel.h: Remove no-longer-used (and unsafe)
	  field in ast_channel for linked lists. The ast_channel structure
	  had a field used for linking a channel into a linked list, but
	  now that ast_channel structures are ao2 objects, this is no
	  longer needed, and could be harmful as ao2 objects really
	  shouldn't ever be placed into linked lists (since those lists
	  don't assist with reference count management on the objects).

	* addons/Makefile: Get chan_ooh323 building again after recent
	  build system changes.

2010-03-25 17:52 +0000 [r254454-254557]  Mark Michelson <mmichelson@digium.com>

	* tests/test_acl.c (added): Add unit test for testing ACL
	  functionality. There are two unit tests contained here. 1.
	  "Invalid ACL" This attempts to read a bunch of badly formatted
	  ACL entries and add them to a host access rule. The goal of this
	  test is to be sure that all invalid entries are rejected as they
	  should be. 2. "ACL" This sets up four ACLs. One is a permit all,
	  one is a deny all, and the other two have specific rules about
	  which subnets are allowed and which are not. Then a set of test
	  addresses is used to determine whether we would allow those
	  addresses to access us when each ACL is applied. This test, by
	  the way, was what resulted in AST-2010-003's creation. Review:
	  https://reviewboard.asterisk.org/r/532

	* include/asterisk/acl.h, /: Merged revisions 254552 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu,
	  25 Mar 2010) | 5 lines Add doxygen for acl.h Review:
	  https://reviewboard.asterisk.org/r/528 ........

	* channels/sip/dialplan_functions.c: Add new rtpsource options to
	  the CHANNEL function. This adds rtpsource options analogous to
	  the rtpdest functions that already exist. In addition, this fixes
	  potential crashes which could result due to trying to read values
	  from nonexistent RTP streams.

	* res/res_rtp_asterisk.c, /: Recorded merge of revisions 254452 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
	  2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
	  Here is a copy and paste of the details from my request on
	  reviewboard that dealt with these changes: Fix 1. The first
	  change in place is to fix Mantis issue 15811, which deals with a
	  situation where Asterisk will incorrectly interpret out of order
	  RFC2833 frames as duplicate DTMF digits. For instance, we would
	  receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
	  DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
	  seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
	  when we received the frame with seqno 5, we would interpret this
	  as a new DTMF 1. With this patch, we will check the seqno of the
	  incoming digit and not process the frame if the seqno is lower
	  than the last recorded seqno. Note that we do not record the
	  seqno of the dropped DTMF frame for future processing. While the
	  above situation is what was designed to be fixed, the patch is
	  written in such a way that the following would also be fixed too:
	  seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
	  seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
	  15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
	  this second situation, the beginning of the DTMF 2 arrives before
	  the final end frame of the DTMF 1. With the patch, seqno 12 is no
	  processed and thus we properly interpret the DTMF. Fix 2. The
	  second change in place is to fix an issue like the following:
	  seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
	  lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
	  *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
	  code in place that was supposed to properly end the previously
	  unended DTMF 1. The problem was that the code was essentially a
	  no-op. The code would set up an end frame for the DTMF 1 but
	  would immediately overwrite the frame with the begin for DTMF 2.
	  I changed process_dtmf_rfc2833() so that instead of returning a
	  single frame, it is given as an output parameter a list of
	  frames. Each frame that needs to be returned is appended to this
	  list. Fix 3. The final change is a minor one where an
	  AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
	  DTMF or an RFC 3389 frame and no frame was returned, then we
	  would return &ast_null_frame. The problem is that earlier in the
	  function, we may have generated an AST_CONTROL_SRCCHANGE frame
	  and put it in the list of frames we wish to return. This frame
	  would be lost in such a case. The patch fixes this problem
	  ........

2010-03-25 16:03 +0000 [r254453]  Terry Wilson <twilson@digium.com>

	* /, main/file.c: Merged revisions 254451 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
	  | 2 lines Handle new SRCCHANGE control message here too ........

2010-03-25 15:27 +0000 [r254450]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, channels/chan_sip.c, res/res_fax.c,
	  configs/sip.conf.sample, include/asterisk/frame.h,
	  channels/sip/include/sip.h: Improve handling of T.38 re-INVITEs
	  that arrive before a T.38-capable application is executing on a
	  channel. This patch addresses an issue found during working with
	  end-users using res_fax. If an incoming call is answered in the
	  dialplan, or jumps to the 'fax' extension due to reception of a
	  CNG tone (with faxdetect enabled), and then the remote endpoint
	  sends a T.38 re-INVITE, it is possible for the channel's T.38
	  state to be 'T38_STATE_NEGOTIATING' when the application starts
	  up. Unfortunately, even if the application wants to use T.38, it
	  can't respond to the peer's negotiation request, because the
	  AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
	  originally has been lost, and the application needs the content
	  of that frame to be able to formulate a reply. This patch adds a
	  new 'request' type to AST_CONTROL_T38_PARAMETERS,
	  AST_T38_REQUEST_PARMS. If the application sends this request,
	  chan_sip will re-send the original control frame (with
	  AST_T38_REQUEST_NEGOTIATE as the request type), and the
	  application can respond as normal. If this occurs within the five
	  second timeout in chan_sip, the automatic cancellation of the
	  peer reinvite will be stopped, and the application will 'own' the
	  negotiation process from that point onwards. This also improves
	  the code path in chan_sip to allow sip_indicate(), when called
	  for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
	  response, which should have been in place before since the
	  control frame *can* fail to be processed properly. It also
	  modifies ast_indicate() to return whatever result the channel
	  driver returned for this control frame, rather than converting
	  all non-zero results into '-1'. Finally, the new request type
	  intentionally returns a positive value, so that an application
	  that sends AST_T38_REQUEST_PARMS can know for certain whether the
	  channel driver accepted it and will be replying with a control
	  frame of its own, or whether it was ignored (if the
	  sip_indicate()/ast_indicate() path had properly supported failure
	  responses before, this would not be necessary). This patch also
	  modifies res_fax to take advantage of the new request. In
	  addition, this patch makes sip_t38_abort() actually lock the
	  private structure before doing its work... bad programmer, no
	  donut. This patch also enhances chan_sip's 'faxdetect' support to
	  allow triggering on T.38 re-INVITEs received as well as CNG tone
	  detection. Review: https://reviewboard.asterisk.org/r/556/

2010-03-25 15:21 +0000 [r254446]  Leif Madsen <lmadsen@digium.com>

	* res/res_agi.c: handle_speechset has 4 arguments. Update code to
	  reflect that handle_speechset has 4 arguments. (closes issue
	  #17093) Reported by: gpatri Patches: res_agi.patch uploaded by
	  gpatri (license 1014) Tested by: pabelanger, mmichelson

2010-03-25 10:09 +0000 [r254406]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c: remove unneeded explicit channel in dahdi
	  ioctls This patch removes some cases where the channel number for
	  an ioctl was passed as a member in a struct rather then through
	  the file descriptor. The gain setting functions passed around a
	  channel which is always 0, and thus this parameter is simply
	  dropped. Review: https://reviewboard.asterisk.org/r/584/

2010-03-24 21:10 +0000 [r254362]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Fix potential invalid reads that could occur in pbx.c
	  Here is a cut and paste of my review request for this change:
	  This past weekend, Russell ran our current suite of unit tests
	  for Asterisk under valgrind. The PBX pattern match test caused
	  valgrind to spew forth two invalid read errors. This patch
	  contains two changes that shut valgrind up and do not cause any
	  new memory leaks. Change 1: In
	  ast_context_remove_extension_callerid2, valgrind reported an
	  invalid read in the for loop close to the function's end.
	  Specifically, one of the the strcmp calls in the loop control was
	  reading invalid memory. This was because the caller of
	  ast_context_remove_extension_callerid2 (__ast_context destroy in
	  this case) passed as a parameter a shallow copy of an ast_exten's
	  exten field. This same ast_exten was what was destroyed inside
	  the for loop, thus any iterations of the for loop beyond the
	  destruction of the ast_exten would result in invalid reads. My
	  fix for this is to make a copy of the ast_exten's exten field and
	  pass the copy to ast_context_remove_extension_callerid2. In
	  addition, I have also acted similarly with the ast_exten's
	  matchcid field. Since in this case a NULL is handled quite
	  differently than an empty string, I needed to be a bit more
	  careful with its handling. Change 2: In __ast_context_destroy, we
	  iterated over a hashtab and called
	  ast_context_remove_extension_callerid2 on each item.
	  Specifically, the hashtab over which we were iterating was an
	  ast_exten's peer_table. Inside of
	  ast_context_remove_extension_callerid2, we could possibly destroy
	  this ast_exten, which also caused the hashtab to be freed.
	  Attempting to call ast_hashtab_end_traversal on the hashtab
	  iterator caused an invalid read to occur when trying to read the
	  iterator->tab->do_locking field since iterator->tab had already
	  been freed. My handling of this problem is a bit less
	  straightforward. With each iteration over the hashtab's contents,
	  we set a variable called "end_traversal" based on the return of
	  ast_context_remove_extension_callerid2. If 0 is ever returned,
	  then we know that the extension was found and destroyed. Because
	  of this, we cannot call ast_hashtab_end_traversal because we will
	  be guaranteeing a read of invalid memory. In such a case, we
	  forego calling ast_hashtab_end_traversal and instead call
	  ast_free on the hashtab iterator. Review:
	  https://reviewboard.asterisk.org/r/585

2010-03-24 18:13 +0000 [r254277-254321]  Jeff Peeler <jpeeler@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Allow configuration of minsecs and nextaftercmd per mailbox.
	  Previously only configurable globally. A unit test has also been
	  written to provide protection against parse failures for
	  supported mailbox options. (closes issue #16864) Reported by:
	  kobaz Patches: voicemail2.patch uploaded by kobaz (license 834)
	  Review: https://reviewboard.asterisk.org/r/555/

	* /, res/res_monitor.c: Merged revisions 254235 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
	  | 72 lines Ensure that monitor recordings are written to the
	  correct location (again) This is an extension to 248860. As such
	  the dialplan test has been extended: ; non absolute path, not
	  combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
	  exten => 5040, n, dial(sip/5001) ; absolute path, not combined
	  exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
	  5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
	  monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
	  combined: changemonitor from non absolute to no path (leaves
	  tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
	  exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
	  dial(sip/5001) ; combined: changemonitor from no path to non
	  absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
	  exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
	  wasn't possible before exten => 5044, n, dial(sip/5001) ; non
	  absolute path, combined exten => 5045, 1,
	  monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
	  dial(sip/5001) ; absolute path, combined exten => 5046, 1,
	  monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
	  dial(sip/5001) ; no path, combined exten => 5047, 1,
	  monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
	  combined: changemonitor from non absolute to absolute (leaves
	  tmp/jeff) exten => 5048, 1,
	  monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
	  changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
	  dial(sip/5001) ; combined: changemonitor from absolute to non
	  absolute (leaves /tmp/jeff) exten => 5049, 1,
	  monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
	  changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
	  dial(sip/5001) ; combined: changemonitor from no path to absolute
	  exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
	  changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
	  dial(sip/5001) ; combined: changemonitor from absolute to no path
	  (leaves /tmp/jeff) exten => 5051, 1,
	  monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
	  changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
	  not combined: changemonitor from non absolute to no path (leaves
	  tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
	  exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
	  dial(sip/5001) ; not combined: changemonitor from no path to non
	  absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
	  5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
	  dial(sip/5001) ; not combined: changemonitor from non absolute to
	  absolute (leaves tmp/jeff) exten => 5054, 1,
	  monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
	  changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
	  dial(sip/5001) ; not combined: changemonitor from absolute to non
	  absolute (leaves /tmp/jeff) exten => 5055, 1,
	  monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
	  changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
	  dial(sip/5001) ; not combined: changemonitor from no path to
	  absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
	  5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
	  n, dial(sip/5001) ; not combined: changemonitor from absolute to
	  no path (leaves /tmp/jeff) exten => 5057, 1,
	  monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
	  changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
	  ........

2010-03-23 22:48 +0000 [r254162]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* main/asterisk.c: make 'core show settings' should show all
	  settable directories (closes issue #17086) Reported by: tzafrir
	  Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir
	  (license 46)

2010-03-23 22:35 +0000 [r254159]  Russell Bryant <russell@digium.com>

	* main/test.c: Put test output for a failure in a CDATA section in
	  the XML results.

2010-03-23 21:17 +0000 [r254050]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Exit native bridging early for greater timing
	  accuracy with warnings This changes native bridging to break one
	  millisecond early so that the more accurate timeval calculations
	  done in the generic bridge can be performed using the bridge
	  config. Currently the time between exiting native bridging
	  slightly late can sometimes cause a large enough discrepancy for
	  warnings to be missed. For the record, 1.4 does not attempt to
	  native bridge at all when warnings are enabled. (closes issue
	  #15815) Reported by: adomjan Review:
	  https://reviewboard.asterisk.org/r/577/

2010-03-23 20:52 +0000 [r254045]  Sean Bright <sean@malleable.com>

	* apps/app_queue.c: Remove unused structure member in app_queue.
	  (closes issue #15494) Reported by: makoto

2010-03-23 19:19 +0000 [r254001]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* tests/Makefile: Change the name of the category 'TEST' to match
	  the name of the subdir

2010-03-23 16:52 +0000 [r253958]  Terry Wilson <twilson@digium.com>

	* main/http.c: Don't act like an http write failed when it didn't
	  fwrite returns the number of items written, not the number of
	  bytes

2010-03-23 14:22 +0000 [r253917]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/Makefile, include/asterisk/logger.h, main/Makefile,
	  Makefile.moddir_rules, pbx/Makefile, res/Makefile, CHANGES,
	  channels/Makefile, include/asterisk/options.h, main/cli.c: Change
	  per-file debug and verbose levels to be per-module, the way users
	  expect them to work. 'core set debug' and 'core set verbose' can
	  optionally change the level for a specific filename; however,
	  this is actually for a specific source file name, not the module
	  that source file is included in. With examples like chan_sip,
	  chan_iax2, chan_misdn and others consisting of multiple source
	  files, this will not lead to the behavior that users expect. If
	  they want to set the debug level for chan_sip, they want it set
	  for all of chan_sip, and not to have to also set it for
	  reqresp_parser and other files that comprise the chan_sip module.
	  This patch changes this functionality to be module-name based
	  instead of file-name based. To make this work, some Makefile
	  modifications were required to ensure that the AST_MODULE
	  definition is present in each object file produced for each
	  module as well. Review: https://reviewboard.asterisk.org/r/574/

2010-03-22 20:32 +0000 [r253872]  Mark Michelson <mmichelson@digium.com>

	* main/asterisk.c: Initialize channels prior to loading "preload"
	  modules. We can have bad results when a module, upon being
	  loaded, attempts to reference the channels container if the
	  container hasn't yet been initialized. I saw this happen by
	  trying to preload pbx_config.so and having a hint defined which
	  referenced a non-existent SIP peer.

2010-03-22 19:52 +0000 [r253800]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/features.c: Merged revisions 253799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
	  2010) | 4 lines Unconditionally copy the caller's account code to
	  the called party. (related to issue #16331) ........

2010-03-22 19:05 +0000 [r253712-253758]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/dbsep.cgi: Update query should be an UPDATE, not
	  a SELECT.

	* contrib/scripts/dbsep.cgi: Return the list for later
	  manipulation. This fixes an issue with the update procedure.
	  Debugging with mmichelson.

	* contrib/scripts/dbsep.cgi, configs/dbsep.conf.sample: Accomodate
	  equal signs in DSNs and add documentation, based upon
	  mmichelson's feedback.

2010-03-20 16:50 +0000 [r253536-253579]  Russell Bryant <russell@digium.com>

	* funcs/func_strings.c: Fix memory corruption found by unit tests.
	  ast_str_reset() was being called on a potentially uninitialized
	  pointer. Valgrind is my hero, once again.

	* cel/cel_pgsql.c, main/tcptls.c, main/manager.c, main/features.c,
	  main/test.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c,
	  main/cel.c: Resolve more compiler warnings on FreeBSD.

	* apps/app_voicemail.c: Include sys/wait.h on FreeBSD to get the
	  WEXITSTATUS() macro.

	* apps/app_dial.c, apps/app_followme.c: Resolve compiler warnings
	  on FreeBSD.

	* pbx/pbx_dundi.c: Resolve a compiler warning on FreeBSD.

	* channels/chan_dahdi.c: Use SHRT_MAX instead of MAXSHORT. These
	  changes fix build issues I had with this module on FreeBSD.

2010-03-19 07:37 +0000 [r253490]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/astobj2.c: prevent segfault if bad magic number is
	  encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report
	  'bad magic number', but internal_ao2_ref continues on, causing
	  segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ
	  before internal_ao2_ref is called, A02_MAGIC is being destroyed
	  (or a wrong pointer) by the time internal_ao2_ref uses
	  INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ
	  encouters a bad magic number. (issue #17037) Reported by:
	  alecdavis Patches: bug17037.diff.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis

2010-03-18 18:23 +0000 [r253357-253378]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Update comment to reflect new timeout value.

	* main/asterisk.c: Increase CLI command output timeout for asterisk
	  -rx to 60 seconds. (closes issue #17049) Reported by: russell
	  Tested by: russell Review:
	  https://reviewboard.asterisk.org/r/573/

2010-03-18 17:52 +0000 [r253345]  Leif Madsen <lmadsen@digium.com>

	* apps/app_userevent.c: Change usage of pipe to comma in UserEvent
	  docs. Change the example usage of pipe as a separator to comma in
	  the UserEvent documentation. (closes issue #16961) Reported by:
	  jlpedrosa

2010-03-18 15:59 +0000 [r253261]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Prevent a crash when a buddy gets offline.
	  (closes issue #16760) Reported by: fiddur Patches: 248394.diff
	  uploaded by fiddur (license 678)i with modifications by me Tested
	  by: fiddur, phsultan

2010-03-18 15:46 +0000 [r253256]  Leif Madsen <lmadsen@digium.com>

	* /, doc/tex/localchannel.tex: Update to new Local channel
	  documentation. Add same changes as commit to 1.4, but convert to
	  TeX. (issue #16963) Reported by: kobaz Patches:
	  localchannel-2.txt uploaded by kobaz (license 834)

2010-03-18 15:45 +0000 [r253255]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Just in case of a race, send the signal
	  on interrupt.

2010-03-17 19:06 +0000 [r253205]  Leif Madsen <lmadsen@digium.com>

	* main/test.c: main/test.c reports erroneous CLI message. (closes
	  issue #17051) Reported by: Nick_Lewis

2010-03-17 14:16 +0000 [r253113]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_gosub.c: Switch to using intptr_t, as suggested by
	  Kevin Fleming on the -dev list

2010-03-17 00:40 +0000 [r253028-253032]  Leif Madsen <lmadsen@digium.com>

	* main/xmldoc.c: Fix a typo.

	* configs/say.conf.sample: Merged revisions 253018 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16
	  Mar 2010) | 6 lines Add french snipset to say.conf. Add the
	  french snipset to say.conf. (Closes issue #15799) ........

2010-03-17 00:23 +0000 [r252976-253004]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_gosub.c: Argh.

	* configure, include/asterisk/autoconfig.h.in, tests/test_gosub.c,
	  configure.ac: Fix bamboo compile error by calculating an integer
	  with the same size as a pointer.

	* tests/test_gosub.c (added), apps/app_stack.c: Mask out previous
	  arguments on each nested invocation of Gosub. (closes issue
	  #16758) Reported by: wdoekes Patches:
	  20100316__issue16758.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/561/

2010-03-16 19:36 +0000 [r252849]  Russell Bryant <russell@digium.com>

	* tests/test_time.c: Re-enable test_time on non-Linux.

2010-03-16 19:36 +0000 [r252848]  Sean Bright <sean@malleable.com>

	* res/res_clialiases.c: Include an extra newline after "Aliased CLI
	  command" to get back the prompt. The other issue mentioned in
	  this bug will be more difficult to resolve since we have no idea
	  (right now) of knowing if the command that is aliased has been
	  installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
	  seanbright

2010-03-16 19:34 +0000 [r252846]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_time.c, include/asterisk/localtime.h,
	  main/stdtime/localtime.c: Fix test_time on Mac OS X (and other
	  platforms without inotify) Reviewboard:
	  https://reviewboard.asterisk.org/r/554/

2010-03-16 19:01 +0000 [r252767]  Russell Bryant <russell@digium.com>

	* utils/Makefile, /: Merged revisions 252766 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
	  | 6 lines Don't treat warnings as errors for muted. muted
	  supports OS X, but uses functions marked as deprecated in 10.6.
	  However, the functions are still supported, so just ignore the
	  warnings for now and allow the build to proceed. ........

2010-03-16 18:48 +0000 [r252762]  Leif Madsen <lmadsen@digium.com>

	* configs/extensions.ael.sample: Merged revisions 252761 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
	  | 7 lines Additional extensions.ael global variable fixes. Fixing
	  up a couple more overlapping global variable namespaces shared
	  with extensions.conf.sample. Also noticed a few of the lines that
	  were commented out didn't have the closing semi-colon so I added
	  that as well. (issue #17035) ........

2010-03-16 18:40 +0000 [r252760]  Tilghman Lesher <tlesher@digium.com>

	* codecs/gsm/Makefile: OSARCH is not inherited to this directory

2010-03-16 18:36 +0000 [r252759]  Russell Bryant <russell@digium.com>

	* tests/test_time.c: Disable this test on non-Linux for now.

2010-03-15 22:48 +0000 [r252709]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_fax.c: Improve handling of values supplied to
	  FAXOPT(ecm). Previously, values that began with whitespace were
	  silently treated as 'no', and all non-'yes' values were also
	  treated as 'no'. Now the supplied value is specifically checked
	  for a 'yes' or 'no' (or equivalent) value, after skipping leading
	  whitespace. If the value is not valid, then a warning message is
	  generated.

2010-03-15 22:14 +0000 [r252627]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Tell the RTP engine API about the initial
	  read and write format. Peer reviewed out-of-band by file.

2010-03-15 21:55 +0000 [r252623]  Sean Bright <sean@malleable.com>

	* apps/app_meetme.c: Resolve a crash in SLATrunk when the specified
	  trunk doesn't exist. Reported by philipp64 in #asterisk-dev.

2010-03-15 21:51 +0000 [r252619]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
	  252617 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
	  | 2 lines Uh, yeah. Umask. I'm stupid. ........

2010-03-15 20:52 +0000 [r252534]  Leif Madsen <lmadsen@digium.com>

	* /, configs/extensions.ael.sample: Merged revisions 252533 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
	  | 7 lines Update extensions.ael file to not overlap
	  extensions.conf. Updated the extensions.ael file so the global
	  variables don't overlap those that we have in extensions.conf
	  (sample files). This way unexpected things won't happed hopefully
	  if both pbx_ael and res_config are loaded. (closes issue #17035)
	  Reported by: pprindeville ........

2010-03-15 16:27 +0000 [r252362-252488]  Tilghman Lesher <tlesher@digium.com>

	* codecs/gsm/Makefile: Make the Makefile logic more explicit and
	  move the Snow Leopard logic down to where it's not executed on
	  non-Darwin systems. (closes issue #17028) Reported by: pabelanger
	  Patches: issue17028_20100315.patch uploaded by seanbright
	  (license 71) 20100315__issue17028.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman, pabelanger

	* channels/chan_sip.c: THIS IS NOT PYTHON. Indentation doesn't
	  matter, only braces do. (closes issue #17025) Reported by:
	  smurfix Patches: sip.patch uploaded by smurfix (license 547)

	* /: Recorded merge of revisions 252366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010)
	  | 2 lines Typo ........

	* Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /,
	  main/asterisk.c: Merged revisions 252361 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
	  | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
	  https://reviewboard.asterisk.org/r/551/ ........

2010-03-14 17:43 +0000 [r252314]  Sean Bright <sean@malleable.com>

	* cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: Fix building
	  CDR and CEL SQLite3 modules. They added a sqlite3_log() function
	  which was conflicting with our function names. (closes issue
	  #17017) Reported by: alephlg

2010-03-14 14:42 +0000 [r252277]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
	  configs/chan_ooh323.conf.sample, addons/ooh323c/src/ooh245.h,
	  addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ootypes.h,
	  addons/ooh323c/src/ooq931.c: generate roundtrip delay requests
	  and responses added response to roundtrip delay requests from
	  opposite side added roundtrip delay request sending to opposite
	  side after answer, added options for sending request (interval
	  between request and count of unreplied requests before forced
	  call hangup) (closes issue #16976) Reported by: vmikhelson
	  Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454)
	  Tested by: vmikhelson, may213

2010-03-13 22:21 +0000 [r252229-252241]  Russell Bryant <russell@digium.com>

	* main/app.c: Resolve unit test failure that occurred on Mac OSX.
	  On Linux (glibc), regcomp() does not return an error for an empty
	  string. However, the version on OSX will return an error. The
	  test for channel group matching by regex now passes on the mac,
	  as well.

	* tests/test_time.c: Resolve compiler warning by paying attention
	  to system() return value. This resolves the last compile failure
	  on bamboo.

2010-03-12 23:18 +0000 [r252133]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_time.c (added): Test script to verify that timezone
	  cache is properly removed on zonefile alteration.

2010-03-12 22:04 +0000 [r252089]  Terry Wilson <twilson@digium.com>

	* main/channel.c, res/res_rtp_asterisk.c, addons/chan_ooh323.c,
	  main/rtp_engine.c, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c, configs/sip.conf.sample,
	  include/asterisk/frame.h, include/asterisk/rtp_engine.h,
	  channels/sip/include/sip.h, channels/chan_mgcp.c: Only change the
	  RTP ssrc when we see that it has changed This change basically
	  reverts the change reviewed in
	  https://reviewboard.asterisk.org/r/374/ and instead limits the
	  updating of the RTP synchronization source to only those times
	  when we detect that the other side of the conversation has
	  changed the ssrc. The problem is that SRCUPDATE control frames
	  are sent many times where we don't want a new ssrc, including
	  whenever Asterisk has to send DTMF in a normal bridge. This is
	  also not the first time that this mistake has been made. The
	  initial implementation of the ast_rtp_new_source function also
	  changed the ssrc--and then it was removed because of this same
	  issue. Then, we put it back in again to fix a different issue.
	  This patch attempts to only change the ssrc when we see that the
	  other side of the conversation has changed the ssrc. It also
	  renames some functions to make their purpose more clear. Review:
	  https://reviewboard.asterisk.org/r/540/

2010-03-12 21:57 +0000 [r252088]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: add missing mfcr2_skip_category setting

2010-03-12 19:43 +0000 [r251989]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Don't override a user option with the
	  global option. (closes issue #16849) Reported by: ip-rob Patches:
	  20100311__issue16849.diff.txt uploaded by tilghman (license 14)
	  Tested by: ip-rob

2010-03-12 19:40 +0000 [r251946-251987]  Richard Mudgett <rmudgett@digium.com>

	* /: Merged revisions 251986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010)
	  | 1 line Make chan_dahdi wakeup_sub() prototype not conditional.
	  ........

	* channels/chan_dahdi.c: Doxegen this chan_dahdi lock.

2010-03-11 21:07 +0000 [r251877-251884]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_exec.c: Because ExecIf needs to reprocess arguments,
	  it's best if we don't remove quotes during parsing. (closes issue
	  #16905) Reported by: ip-rob Patches:
	  20100303__issue16905.diff.txt uploaded by tilghman (license 14)
	  Tested by: ip-rob

	* tests/test_stringfields.c: Fix tests on 32-bit systems.

	* apps/app_system.c: If the argument to the system application is
	  quoted, ensure we remove the quotes before trying to execute.
	  (closes issue #16842) Reported by: ip-rob Patches:
	  20100310__issue16842.diff.txt uploaded by tilghman (license 14)
	  Tested by: ip-rob

2010-03-11 18:07 +0000 [r251821]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c: Minor tweaks and
	  comment updates to chan_dahdi.

2010-03-11 07:03 +0000 [r251779]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_directory.c: Add supporting code for app-directory pause
	  option. Since 1.6.1 CLI help reports that option p(n) 'initial
	  pause' is available. Supporting code was never implemented.
	  (closes issue #16751) Reported by: alecdavis Patches:
	  directory_pause.trunk.diff.txt uploaded by alecdavis (license
	  585) Tested by: alecdavis Review:
	  https://reviewboard.asterisk.org/r/481/

2010-03-10 23:15 +0000 [r251736]  Jeff Peeler <jpeeler@digium.com>

	* tests/test_stringfields.c (added), main/utils.c: Add new unit
	  test for stringfields. (Copied from reviewboard) Tests the
	  following: 1. Basic allocation and setting of string fields. 2.
	  Shrinking a string field and re-expanding it. 3. Growing the last
	  allocation in a string field pool. 4. Setting a string to a large
	  value such that a new string field pool must be allocated. In
	  each part, we make sure that the string field is accurate (has
	  the correct value in it), make sure that the 2 bytes before the
	  string field has the correct capacity for the field, and for
	  tests 2-4, we make sure that the string field is where we expect
	  it to be in memory. Also tested: 5. Shrinking a string field and
	  partially re-expanding it. 6. Setting strings in such a way as to
	  create three separate string field pools and then removing the
	  middle pool. There is a bug fix in the init function, which
	  ensures the embedded_pool is set to NULL which is important for
	  stack allocated structures. Review:
	  https://reviewboard.asterisk.org/r/185/

2010-03-10 20:54 +0000 [r251682]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c: Hmmm, apparently needed to be fixed in
	  trunk, too. (closes issue #16900) Reported by: bluecrow76
	  Patches: asterisk-1.6.2.4-func_strings.diff uploaded by
	  bluecrow76 (license 270)

2010-03-10 20:53 +0000 [r251680]  Leif Madsen <lmadsen@digium.com>

	* apps/app_record.c: Be less ambiguous in Record() app docs. For
	  some reason the documentation for the 'k' application in trunk
	  and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them
	  all to match. The wording in 1.6.2 and trunk was ambiguous, so
	  you could interpret the wording the mean that recording would
	  continue upon hangup indefinitely, or you could interpret it to
	  mean that the recorded data would not be discarded upon hangup.
	  This change makes it clear we mean the latter, and not the
	  former. Came from a discussion in #asterisk on IRC.

2010-03-10 20:51 +0000 [r251679]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Fix ParkAndAnnounce not respecting parking
	  options. The patch ensures that if a peer does not exist, parking
	  settings are read from the channel. A unit test has been written
	  to ensure proper operation for both standard parking and parking
	  using masquerades. (closes issue #16592) Reported by: mwyres
	  Patches: bug_16592.diff uploaded by snuffy (license 35) Review:
	  https://reviewboard.asterisk.org/r/539/

2010-03-10 20:30 +0000 [r251677]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_substitution.c, funcs/func_strings.c: It's amazing
	  what writing a test will find. (issue #16900) Reported by:
	  bluecrow76

2010-03-10 18:25 +0000 [r251631]  Jeff Peeler <jpeeler@digium.com>

	* main/abstract_jb.c: Fix jitterbuffer logging not creating
	  logfiles. Three changes made here: 1) Do not fail if a previous
	  log does not exist (in fact, this is probably expected). 2)
	  Ensure that the file descriptor to write to gets assigned
	  properly. I am at a loss as to why assigning safe_fd outside the
	  if fixes this, but it makes the if statement slightly less
	  complicated anyway. 3) Move up the failure message so that the
	  errno of the failure is not overwritten by fclose. (closes issue
	  #16917) Reported by: Artem

2010-03-10 16:55 +0000 [r251538-251585]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h, channels/sig_pri.c: Simplified
	  dahdi_request() channel selection failed reason/cause code. Also
	  avoid potential crash because cause could be NULL.

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Reduce the amount of database access for
	  HAVE_PRI_SERVICE_MESSAGES. Rework HAVE_PRI_SERVICE_MESSAGES to
	  not use the active values directly from the database. Database
	  access is likely expensive. Database access now only happens on
	  initialization, destruction, and when the B channel is taken in
	  or out of service. This change is not related to call waiting but
	  it would cause the search for a call waiting interface to be very
	  expensive and slow down D channel message servicing.

2010-03-09 20:30 +0000 [r251475]  Tilghman Lesher <tlesher@digium.com>

	* codecs/gsm/Makefile, Makefile.rules: Build system modifications
	  to ensure that Asterisk properly builds on Mac OS X 10.6. (closes
	  issue #16997) Reported by: jquinn Patches:
	  20100309__issue16997__2.diff.txt uploaded by tilghman (license
	  14) Tested by: tilghman, russell

2010-03-08 18:08 +0000 [r251310]  Leif Madsen <lmadsen@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 251309 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010)
	  | 13 lines Fix Debian init script to not use -c. When using the
	  init script as-is currently, it could cause issues on Debian such
	  as high CPU usage. This fix has worked for several people so I'm
	  implementing the change. (closes issue #16784) Reported by:
	  pabelanger Tested by: pabelanger, mnick, davidw, mutineer612
	  (closes issue #16887) Reported by: jlpedrosa Tested by:
	  jlpedrosa, mutineer612 ........

2010-03-08 05:15 +0000 [r251262-251263]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/stdtime/localtime.c: Remove portions that weren't meant to
	  be committed for the OS X compat fix

	* funcs/func_pitchshift.c, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
	  main/stdtime/localtime.c: Change needed to make Mac OS X 10.6
	  happy

2010-03-07 14:53 +0000 [r251221-251222]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Clean transmit_* for start/stop media
	  transmission Small patch changing skinny_set_rtp_peer to use
	  transmit_stopmediatransmission and to use new
	  transmit_startmediatransmission. Basic testing on 30VIP's by
	  wedhorn Basic testing on 7960 by me (closes issue #16956)
	  Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by
	  wedhorn (license 30) Tested by: wedhorn,mvanbaak

	* channels/chan_skinny.c: Cleanup transmit_callstate handling Broke
	  the various functions included in transmit_callstate to their own
	  functions. Transmit_callstate now just transmits callstate.
	  Generally left the functionality as it was, which highlight some
	  minor code issues (eg multiple transmit_callstate's). I did
	  however revise the hint code usage of the old transmit_callstate
	  as it it not appropriate to put a device on hook based on the
	  change of a hinted device. (closes issue #16939) Reported by:
	  wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license
	  30) Tested by: mvanbaak,wedhorn

2010-03-07 00:45 +0000 [r251181]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooq931.c: small log issue from bug 0016664

2010-03-06 14:16 +0000 [r251137]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a crash in SIP blind transfer handling
	  found by an automated external test. The first real test added to
	  the external test suite found a pretty nasty crash that occurred
	  in Asterisk trunk. The crash was due to a race condition between
	  the REFER handling and channel destruction in the channel thread.
	  After the transfer has been completed, we go back to the
	  transferrer channel and try to lock it so we can fire off a CEL
	  event. However, there was no guarantee that the channel was still
	  around at that point since it's racing against the channel
	  thread. Since ast_channel is a reference counted object, the fix
	  is simple. The code unlocks the transferrer channel before
	  finally completing the transfer with an async goto. At this point
	  the channel thread is going to start call tear down and the
	  channel will eventually be destroyed. To ensure that the channel
	  is valid when we want to fire off the CEL event, increase the
	  channel's reference count.

2010-03-05 21:51 +0000 [r251038-251087]  David Vossel <dvossel@digium.com>

	* funcs/func_pitchshift.c: fixes xml error in func_pitchshift

	* funcs/func_pitchshift.c (added), CHANGES: PITCH_SHIFT dialplan
	  function The PITCH_SHIFT function can be used on a channel to
	  independently modify the pitch of both rx and tx audio streams.
	  Now you can improve your conference calls by assigning a random
	  pitch effect to everyone entering a meetme room, or just make
	  your day more interesting by making your co-workers sound funny.
	  These are just some of the numerious practical uses for this
	  function. Enjoy! https://reviewboard.asterisk.org/r/526/

2010-03-05 19:32 +0000 [r251022]  Russell Bryant <russell@digium.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
	  pbx/pbx_gtkconsole.c (removed): Remove pbx_gtkconsole and related
	  gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/

2010-03-05 19:10 +0000 [r250979]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_followme.c: Fix app_followme playing wrong sound files.
	  Fixes regression introduced in 140167 that uses the wrong
	  variable names. (closes issue #16930) Reported by: ianc Patches:
	  fix_reload_followme.diff uploaded by ianc (license 998)

2010-03-05 05:03 +0000 [r250917]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix up some of chan_sip's usage of the RTP
	  engine API. The get_local_address() function for an RTP instance
	  was used when building an SDP, but the results were not honored.
	  The RTP engine activate() function was not being used once we
	  have determined that media will now flow.

2010-03-05 04:37 +0000 [r250913]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Missing quote in ODBC query. (closes issue
	  #16953) Reported by: elguero Patches:
	  app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license
	  37)

2010-03-05 02:07 +0000 [r250871]  Russell Bryant <russell@digium.com>

	* include/asterisk/rtp_engine.h: Fix up the ast_rtp_property enum.
	  The mis-placement of the latest entry meant that when it was set,
	  it was writing one index past the end of the properties array in
	  the ast_rtp_instance (which happened to be the local_address
	  field).

2010-03-05 01:05 +0000 [r250787]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 250786 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04
	  Mar 2010) | 9 lines Fix not being able to specify a URL in MOH
	  class directory. Don't attempt to chdir on a URL! (closes issue
	  #16875) Reported by: raarts Patches: moh-http.patch uploaded by
	  raarts (license 937) ........

2010-03-04 20:12 +0000 [r250730]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_channel.c: Adjust XML for func_channel to indicate
	  that rtpdest can take a "text" argument.

2010-03-03 21:28 +0000 [r250609-250614]  Leif Madsen <lmadsen@digium.com>

	* /: Recorded merge of revisions 250613 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010)
	  | 11 lines Update existing Local channel documentation. A
	  complete re-write of the Local channel documentation has been
	  performed, with the existing information from localchannel.txt
	  and localchannel.tex merged in. (issue #16637) Reported by: kobaz
	  Patches: localchannel.tex uploaded by lmadsen (license 10)
	  localchannel.txt uploaded by lmadsen (license 10) Tested by:
	  lmadsen, jsmith, mmichelson ........

	* doc/tex/localchannel.tex: Update existing Local channel
	  documentation. A complete re-write of the Local channel
	  documentation has been performed, with the existing information
	  from localchannel.txt and localchannel.tex merged in. (closes
	  issue #16637) Reported by: kobaz Patches: localchannel.tex
	  uploaded by lmadsen (license 10) localchannel.txt uploaded by
	  lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson

2010-03-03 19:38 +0000 [r250565]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, channels/chan_dahdi.c, main/dial.c,
	  channels/chan_local.c, include/asterisk/channel.h,
	  apps/app_queue.c: Removed cdrflags from ast_channel structure.
	  Only chan_dahdi set a value in cdrflags. Everyone else just
	  copied it around the system. Noone cared about any value it may
	  have contained.

2010-03-03 19:06 +0000 [r250481]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  250480 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
	  | 15 lines Make sure to clear red alarm after polarity reversal.
	  From the issue: The automatic overnight line tests (or manual
	  ones) used on UK (BT) lines causes a red alarm on a dahdi /
	  TDM400P connected channel. This is because the line uses voltage
	  tests (battery loss) and polarity reversal. The polarity reversal
	  causes chan_dahdi to initiate v23 CallerID processing but during
	  this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
	  is never cleared. (closes issue #14163) Reported by: jedi98
	  Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
	  653) Tested by: mattbrown, Chainsaw, mikeeccleston ........

2010-03-03 19:02 +0000 [r250395-250478]  David Vossel <dvossel@digium.com>

	* main/test.c: Changes 0ms to <1ms in cli END results during 'test
	  execute'

	* /, channels/chan_iax2.c: Merged revisions 250394 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03
	  Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets
	  When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
	  call store_by_transfercallno() to link the chan_iax2_pvt struct
	  into iax_transfercallno_pvts. If a duplicate TXREQ packet is
	  received for the same call, the pvt struct will be linked into
	  iax_transfercallno_pvts multiple times. This patch fixes this.
	  Thanks rain for debugging this and providing a patch! (closes
	  issue #16904) Reported by: rain Patches:
	  iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
	  by: rain, dvossel ........

2010-03-03 17:37 +0000 [r250392]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
	  Add new config option to control AMI alarm event reporting in
	  chan_dahdi. New config parameter "reportalarms" added in
	  chan_dahdi.conf which supports the following possible values:
	  "channels": report each channel alarms (current behavior, default
	  for backward compatibility) "spans": report an "SpanAlarm" event
	  when the span of any configured channel is alarmed "all": report
	  channel and span alarms (aggregated behavior) "none": do not
	  report any alarms (closes issue #16709) Reported by: nahuelgreco
	  Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco
	  (license 162)

2010-03-03 16:43 +0000 [r250303-250346]  Tilghman Lesher <tlesher@digium.com>

	* main/editline/configure: One more fix to editline

	* main/editline/configure, main/editline/Makefile.in,
	  main/editline/sys.h, main/editline/configure.in: Eliminate
	  remaining libedit warnings (shown in bamboo)

2010-03-03 15:39 +0000 [r250302]  Matthew Nicholson <mnicholson@digium.com>

	* res/res_fax.c, apps/app_fax.c, CHANGES, res/res_fax_spandsp.c:
	  Updated CHANGES file to mention res_fax and res_fax_spandsp. Also
	  fixed MODULEINFO depends and conflicts for app_fax, res_fax, and
	  res_fax_spandsp.

2010-03-03 00:18 +0000 [r250235-250246]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes signed to unsigned int comparision
	  issue for FaxMaxDatagram value.

	* main/test.c: fixes assumption that test failed if it did not pass
	  when generating results

	* tests/test_utils.c: base64 unit test

2010-03-02 23:22 +0000 [r250190-250213]  Matthew Nicholson <mnicholson@digium.com>

	* configs/res_fax.conf.sample (added), include/asterisk/res_fax.h
	  (added): Merge missed files from res_fax/res_fax_spandsp merge.

	* res/res_fax.c (added), res/res_fax.exports (added),
	  include/asterisk/frame.h, res/res_fax_spandsp.c (added): Merge
	  res_fax and res_fax_spandsp.

2010-03-02 21:58 +0000 [r250141]  David Vossel <dvossel@digium.com>

	* apps/app_directed_pickup.c, CHANGES: adds 'p' option to
	  PickupChan The 'p' option allows the PickupChan app to pickup a
	  ringing phone by looking for the first match to a partial channel
	  name rather than requiring a full match. (closes issue #16613)
	  Reported by: syspert Patches: pickipbycallid.patch uploaded by
	  syspert (license 938) pickupbycallerid_v2.patch uploaded by
	  dvossel (license 671) Tested by: dvossel, syspert

2010-03-02 21:09 +0000 [r249950-250051]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/imapstorage.tex: Update IMAP documentation. Update the
	  IMAP documentation to make it clear that storing voicemails in
	  the same folder as a large number of emails could potentially
	  cause significant slow downs when writing or retrieving
	  voicemails. (issue #16704) Reported by: TimeHider Tested by:
	  lmadsen, TimeHider

	* /, configs/cdr.conf.sample: Merged revisions 250043 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02
	  Mar 2010) | 7 lines Update documentation to clarify purpose of
	  unanswered option. (closes issue #16267) Reported by: elsto
	  Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
	  10) Tested by: davidw, elsto ........

	* /: Recorded merge of revisions 250041 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010)
	  | 4 lines Update documentation to not imply we support overriding
	  options. (issue #16855) Reported by: davidw ........

	* doc/tex/configuration.tex: Update documentation to not imply we
	  support overriding options. (closes issue #16855) Reported by:
	  davidw

	* apps/app_directory.c: Fix literal values wrapped in
	  documentation. (closes issue #16145) Reported by: tilghman

2010-03-02 19:39 +0000 [r249947]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_echo.c: revert ability to exit echo app caused a
	  regression, as only supported VOICE, not VIDEO etc. (issue
	  #16880)

2010-03-02 19:24 +0000 [r249912-249925]  Leif Madsen <lmadsen@digium.com>

	* main/features.c: Add missing description of the PARKINGLOT
	  variable in XML documentation. (closes issue #16743) Reported by:
	  snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35)

	* pbx/pbx_dundi.c: Convert some DUNDI functions to XML
	  documentation. (closes issue #16798) Reported by: snuffy Patches:
	  xml_dundi.diff uploaded by snuffy (license 35)

2010-03-02 19:08 +0000 [r249893]  David Vossel <dvossel@digium.com>

	* channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
	  configs/console.conf.sample, channels/chan_local.c,
	  channels/chan_sip.c, configs/oss.conf.sample,
	  configs/usbradio.conf.sample, configs/misdn.conf.sample,
	  channels/chan_console.c, channels/chan_gtalk.c,
	  channels/chan_oss.c, channels/misdn_config.c,
	  include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
	  channels/chan_jingle.c, channels/chan_usbradio.c,
	  channels/chan_dahdi.c, channels/chan_skinny.c,
	  configs/mgcp.conf.sample, main/abstract_jb.c,
	  channels/chan_h323.c, channels/chan_alsa.c,
	  configs/sip.conf.sample, channels/chan_mgcp.c: fixes adaptive
	  jitterbuffer configuration When configuring the adaptive
	  jitterbuffer, the target_extra value not only could not be set
	  from the configuration, but was not even being set to its proper
	  default. This value is required in order for the adaptive
	  jitterbuffer to work correctly. To resolve this a config option
	  has been added to expose this value to the conf files, and a
	  default value is provided when no config specific value is
	  present.

2010-03-02 19:02 +0000 [r249892]  Leif Madsen <lmadsen@digium.com>

	* apps/app_osplookup.c, apps/app_confbridge.c, res/res_jabber.c:
	  Fix several XML documentation validate errors.

2010-03-02 18:31 +0000 [r249889-249891]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: fix build by checking result of symlink in
	  test_voicemail_vmsayname

	* CHANGES, apps/app_voicemail.c: Add new application VMSayName for
	  use with voicemail. VMSayName that will play the recorded name of
	  the voicemail user if it exists, otherwise will play the mailbox
	  number. A unit test has been written to verify correct
	  functionality called test_voicemail_vmsayname. (closes issue
	  #14973) Reported by: ghjm Review:
	  https://reviewboard.asterisk.org/r/530/

2010-03-02 07:38 +0000 [r249759-249801]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_echo.c: fixes ability to exit echo app when called from
	  a ISDN channel, null frames prevent '#' exit. Now only echo back
	  VOICE and DTMF frames (issue #16880) Reported by: alecdavis
	  Patches: echo_exit.diff.txt uploaded by alecdavis (license 585)
	  Tested by: alecdavis

	* channels/chan_dahdi.c: fix asterisk setting of pritimers from
	  chan_dahdi.conf regression since sig_pri split. (issue #16909)
	  Reported by: alecdavis Patches: pritimer.asterisk.diff.txt
	  uploaded by alecdavis (license 585) Tested by: alecdavis

2010-03-01 19:36 +0000 [r249672]  Sean Bright <sean@malleable.com>

	* /, apps/app_voicemail.c: Merged revisions 249671 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon,
	  01 Mar 2010) | 11 lines Fix crash in app_voicemail related to
	  message counting. We were passing a 'struct inprocess **' and
	  treating it like a 'struct inprocess *' causing a segfault.
	  (closes issue #16921) Reported by: whardier Patches:
	  20100301_issue16921.patch uploaded by seanbright (license 71)
	  Tested by: whardier ........

2010-03-01 19:33 +0000 [r249669-249670]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Cleanup display_*message functions. This
	  patch splits transmit_displaymessage into
	  transmit_clear_display_message and transmit_display_message which
	  better aligns with the skinny protocol. The new
	  transmit_display_message is not used in the current code, but
	  will be and so it is commented. Moved handle_datetime from this
	  function to onhook and offhook functions (display now properly
	  cleared at the end of a call on 30VIP). Removed skinny debug
	  messages from inline code as there's an ast_verb in
	  transmit_clear_display_message. Also, removed commentary that it
	  was a clear display as it is now apparent from the function name.
	  Split transmit_displaypromptmessage into display and clear.
	  (closes issue #16878) Reported by: wedhorn Patches:
	  skinny-clean02.diff uploaded by wedhorn (license 30)
	  skinny-clean03.diff uploaded by wedhorn (license 30)

	* channels/chan_skinny.c: fix endianes issues in chan_skinny
	  (closes issue #16826) Reported by: PipoCanaja Patches:
	  chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja
	  (license 994) Tested by: wedhorn

2010-03-01 18:36 +0000 [r249623]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Constify a bit of app_voicemail, to make
	  ODBC and IMAP compile once again.

2010-03-01 17:11 +0000 [r249538]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c, /: Merged revisions 249536 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01
	  Mar 2010) | 11 lines Modify queued frames from local channels to
	  not set the other side to up In this case, attended transfers
	  were broken due to ast_feature_request_and_dial detecting the
	  channel being set to up before the answer frame could be read and
	  therefore failing to mark the channel as ready. This fix is a
	  regression fix for 244785, which should continue to work properly
	  as well. (closes issue #16816) Reported by: jamhed Tested by:
	  jamhed, corruptor ........

2010-02-28 20:50 +0000 [r249491]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Fix unit test that Alec Davis broke.
	  (closes issue #16927) Reported by: alecdavis

2010-02-28 16:36 +0000 [r249449]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_voicemail.c: make unit test check for NULL folder, which
	  then defaults to INBOX previous test, gave false level of
	  assurance that code was healthy. (issue #16927) Reported by:
	  alecdavis Patches: based on app_voicemail_test.diff.txt uploaded
	  by alecdavis (license 585) Tested by: alecdavis

2010-02-28 07:10 +0000 [r249405]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, apps/app_voicemail.c: Properly document
	  voicemail API documents. Also fix a crash reported via the -dev
	  list.

2010-02-27 22:49 +0000 [r249320]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sig_pri.c: overlap receiving: automatically send CALL
	  PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
	  user has determined that sufficient call information has been
	  received the user shall stop T302 and send CALL PROCEEDING to the
	  network. Previously timeouts were possible if the dialplan took a
	  long time to issue any response back to the network. Verified
	  that our local TELCO also does the same. (issue #16789) Reported
	  by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
	  by alecdavis (license 585) Tested by: alecdavis

2010-02-27 14:08 +0000 [r249235]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 249234 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
	  Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
	  ........

2010-02-26 18:41 +0000 [r249187]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Cleanups to fix bugs in the VM count API
	  functions. - Urgent voicemails were not attached, because the
	  attachment code looked in the wrong folder. - Urgent voicemails
	  were sometimes counted twice when displaying the count of new
	  messages. - Backends were inconsistent as to which voicemails
	  each API counted. - Unit tests added to verify behavior in the
	  future. (closes issue #15654) Reported by: tomo1657 Patches:
	  20100225__issue15654.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman (closes issue #16448) Reported by: hevad
	  Review: https://reviewboard.asterisk.org/r/525/

2010-02-26 18:41 +0000 [r249186]  David Vossel <dvossel@digium.com>

	* main/test.c: adds Time field to "test show results" cli command

2010-02-26 17:13 +0000 [r249101-249105]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Send a manager event when the manager
	  BridgeAction command is used. (closes issue #16769) Reported by:
	  syspert Patches: bridgeaction.patch uploaded by syspert (license
	  938)

	* /, channels/chan_sip.c: Merged revisions 249100 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
	  2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
	  (closes issue #16792) Reported by: vrban Patches: t38_606.patch
	  uploaded by vrban (license 756) ........

2010-02-26 08:45 +0000 [r249009-249058]  Russell Bryant <russell@digium.com>

	* cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
	  cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
	  cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
	  cdr/cdr_tds.c, cdr/cdr_csv.c: formatting tweaks and
	  constification

	* main/cdr.c: Trim trailing whitespace (to help reduce diff against
	  cdr-q branch)

	* include/asterisk/cdr.h: Trim trailing whitespace, convert lists
	  of defines to enums

	* cdr/cdr_sqlite.c: trivial formatting tweak (working on reducing
	  diff against trunk for cdr-q)

	* cdr/cdr_sqlite3_custom.c: remove include

	* cdr/cdr_csv.c: constification, remove include

	* cdr/cdr_tds.c: Remove unnecessary includes, formatting tweak

	* cdr/cdr_pgsql.c: constification and remove unnecessary include

2010-02-25 23:09 +0000 [r248952]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_monitor.c: Merged revisions 248860 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
	  | 18 lines Ensure that monitor recordings are written to the
	  correct location (again) This is an extension to 248757. As such
	  the dialplan test has been extended: exten => 5040, 1,
	  monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
	  dial(sip/5001) exten => 5041, 1,
	  monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
	  dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
	  exten => 5042, n, dial(sip/5001) exten => 5043, 1,
	  monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
	  changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
	  exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
	  changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
	  design and emits a warning exten => 5044, n, dial(sip/5001)
	  ........

2010-02-25 22:41 +0000 [r248946]  Mark Michelson <mmichelson@digium.com>

	* main/acl.c: Fix incorrect ACL behavior when CIDR notation of "/0"
	  is used. AST-2010-003

2010-02-25 21:22 +0000 [r248861]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 248859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
	  | 15 lines Some platforms clear /var/run at boot, which makes
	  connecting a remote console... difficult. Previously, we only
	  created the default /var/run/asterisk directory at install time.
	  While we could create it in the init script, that would not work
	  for those who start asterisk manually from the command line. So
	  the safest thing to do is to create it as part of the Asterisk
	  boot process. This also changes the ownership of the directory,
	  because the pid and ctl files are created after we setuid/setgid.
	  (closes issue #16802) Reported by: Brian Patches:
	  20100224__issue16802.diff.txt uploaded by tilghman (license 14)
	  Tested by: tzafrir ........

2010-02-25 18:37 +0000 [r248793]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_monitor.c: Merged revisions 248757 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
	  | 15 lines Ensure that monitor recordings are written to the
	  correct location. Recordings should be placed in the monitor
	  directory when a non-absolute path is used. Exact dialplan used
	  for testing: exten => 5040, 1,
	  monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
	  dial(sip/5001) exten => 5041, 1,
	  monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
	  dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
	  exten => 5042, n, dial(sip/5001) ABE-2101 ........

2010-02-24 22:44 +0000 [r248584-248667]  Tilghman Lesher <tlesher@digium.com>

	* channels/Makefile: Also kill the .i files, or else the build
	  process will not recreate them, when we change flags. Fixes a
	  weird symbol problem mmichelson was having in a group branch, but
	  also applies to trunk.

	* /, main/logger.c, include/asterisk/term.h, main/term.c: Merged
	  revisions 248582 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
	  | 7 lines Remove color code sequences from verbose messages that
	  go to logfiles. (closes issue #16786) Reported by: dodo Patches:
	  logger2.patch uploaded by dodo (license 989) Tested by: tilghman
	  ........

2010-02-24 06:39 +0000 [r248533-248534]  Russell Bryant <russell@digium.com>

	* funcs/func_strings.c: Remove unnecessary warning message, make a
	  couple of formatting tweaks

	* tests/test_strings.c: Add ASTERISK_FILE_VERSION macro.

2010-02-23 22:29 +0000 [r248489]  Mark Michelson <mmichelson@digium.com>

	* tests/test_strings.c (added): Unit test for ast_str API. Review:
	  https://reviewboard.asterisk.org/r/517

2010-02-23 16:34 +0000 [r248397]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 248396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
	  | 9 lines fixes invite with replaces deadlock (closes issue
	  #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
	  uploaded by dvossel (license 671) Tested by: pwalker, dvossel
	  ........

2010-02-22 20:19 +0000 [r248347]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Move the REF_DEBUG comment higher in the
	  include list. Uncommenting the REF_DEBUG definition where it was
	  in the source resulted in only a small part of the astobj2
	  references being logged to a file. Moving this up higher in the
	  include list causes all references to be logged as they should
	  be.

2010-02-22 06:45 +0000 [r248225-248226]  Russell Bryant <russell@digium.com>

	* include/asterisk/taskprocessor.h, main/taskprocessor.c: Minor
	  tweaks to comment blocks and includes. Fix the copyright lines,
	  tweak doxygen formatting, and remove some unnecessary includes.

	* tests/test_devicestate.c: Tweak copyright and author lines.

2010-02-21 12:09 +0000 [r248184]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Cleanup transmit_* functions, part 1
	  Break transmit_tone into transmit_start_tone and
	  transmit_stop_tone as per the skinny protocol. (closes issue
	  #16874) Reported by: wedhorn Patches: skinny-clean01.diff
	  uploaded by wedhorn (license 30)

2010-02-20 22:37 +0000 [r248108]  Olle Johansson <oej@edvina.net>

	* res/res_rtp_asterisk.c: Improve support for RTCP reports without
	  report blocks

2010-02-19 18:38 +0000 [r248003]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: mfcr2 issue 0016844 - Fix portability bit
	  fields and make mfcr2_immediate_accept work again, reported and
	  patched by korihor

2010-02-19 17:40 +0000 [r247915]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: handle_request_invite revise comment, fix
	  coding guideline issues I'm working with this code right now
	  trying to analyze a deadlock. This change is just to clean up a
	  few things before I make a more complex patch.

2010-02-19 17:33 +0000 [r247914]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 247910 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
	  (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
	  .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
	  19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
	  consistent with other channel technologies. The processing of
	  DTMF tones on the receiving side of an ISDN channel is
	  inconsistent with the way it is handled in other channels,
	  especially DAHDI analog. This causes DTMF tones sent from an ISDN
	  phone to be doubled at the connected party. We are using the
	  following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
	  Option one is necessary because the asterisk DSP DTMF detection
	  is better than mISDN's internal DSP. Not as many false positives.
	  Option two is necessary to transmit DTMF tones end to end when
	  mISDN channels are connected to SIP channels with out of band
	  DTMF for example. The symptom is that DTMF tones sent by an ISDN
	  phone are doubled on the way through asterisk when two mISDN
	  channels are connected with a Local channel in between or if it
	  is bridged to an analog channel. The doubling of DTMF tones is
	  because DTMF is passed inband to asterisk by the mISDN channel
	  and passed out of band once again after the release of the DTMF
	  tone. Passing it inband is wrong. Neither an analog channel nor
	  SIP channel passes DTMF inband if configured to inband DTMF.
	  Analog and SIP channels filter out the DTMF tones because they
	  use the voice frames returned by ast_dsp_process. But chan_misdn
	  passes the unfiltered input voice frames instead. To overcome one
	  aspect of the problem, the doubling of DTMF tones when two mISDN
	  channels are directly bridged, someone made an 'optimization',
	  where in that case the DTMF tone passed out-of-band to the peer
	  channel is not translated to an inband tone at the transmit side.
	  This optimization is bad because it does not work in general. For
	  example, analog channels or mISDN channels when bridged through
	  an intermediary local channel will generate DTMF tones from
	  out-of-band information. Also, of course, it must not be done
	  when there is no inband DTMF available. This patch fixes the
	  issue. Now chan_misdn will filter the received inband DTMF signal
	  the same as other channel types. Another change included: No need
	  to build an extra translation path because ast_process_dsp does
	  it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
	  ................

2010-02-18 23:13 +0000 [r247787-247841]  Tilghman Lesher <tlesher@digium.com>

	* res/res_speech.c: Revert an errant part of a previous cleanup, to
	  fix a memory corruption issue. (closes issue #16368) Reported by:
	  thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf
	  (license 955)

	* channels/chan_sip.c: If the peer record is from realtime, it
	  could be set to 0, due to MySQL not representing NULL well in
	  integer columns. NULL means the value is not specified for the
	  column, which normally means the driver uses whatever is the
	  default value. However, on MySQL, placing a NULL in either a
	  float or integer column results in a retrieval of the 0 value.
	  Hence, users get an errant error on load. This patch suppresses
	  that error and makes the value as if it was not there. Note that
	  this cannot be done in the realtime driver, because the lack of
	  difference between NULL and 0 can only be intepreted correctly by
	  the driver itself. If we did it in the realtime driver, then it
	  would be effectively impossible to set any realtime field to 0,
	  because it would act as if the field were unspecified and
	  possibly take on a different value. (closes issue #16683)
	  Reported by: wdoekes

2010-02-18 21:23 +0000 [r247736-247770]  David Vossel <dvossel@digium.com>

	* bridges/bridge_softmix.c: fixes confbridge crash when no timing
	  module is loaded. (closes issue #16471) Reported by: kjotte
	  Patches: M16471.diff uploaded by junky (license 177) Tested by:
	  kjotte, junky

	* apps/app_queue.c: fixes Queue with C option crash (closes issue
	  #16475) Reported by: okrief Patches: queue_crash.diff uploaded by
	  dvossel (license 671)

2010-02-18 19:39 +0000 [r247652]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/features.c: Merged revisions 247651 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
	  2010) | 6 lines Copy the calling party's account code to the
	  called party if they don't already have one. (closes issue
	  #16331) Reported by: bluefox Tested by: mnicholson ........

2010-02-18 18:31 +0000 [r247609]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix placing ISDN calls on hold preventing native
	  bridging from being reexamined after a transfer. Consider the
	  following scenario: /-- B A == * == Network \-- C Party B calls
	  party A (EuroISDN BRI phone) Party A puts B on hold using the
	  HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on
	  hold to talk with party B again. Party A transfers B to C by
	  hanging up. The call does not get the opportunity to get
	  re-transferred into the ISDN network by the native bridge because
	  native bridging is not being reexamined after the initial
	  transfer.

2010-02-18 16:54 +0000 [r247503-247509]  Leif Madsen <lmadsen@digium.com>

	* /, README-SERIOUSLY.bestpractices.txt: Merged revisions 247508
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010)
	  | 1 line Add additional link to best practices document per
	  jsmith. ........

	* /, README-SERIOUSLY.bestpractices.txt (added): Merged revisions
	  247502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
	  | 10 lines Add best practices documentation. (issue #16808)
	  Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
	  Tested by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/507/ ........

2010-02-18 16:34 +0000 [r247500]  Philippe Sultan <philippe.sultan@gmail.com>

	* CHANGES, res/res_jabber.c: Add a new manager event for our
	  buddies status. The new JabberStatus event gives a concise view
	  of the status change to the AMI clients. Thanks fiddur! (closes
	  issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded
	  by fiddur (license 678) Tested by: fiddur, phsultan

2010-02-18 04:20 +0000 [r247423]  Russell Bryant <russell@digium.com>

	* Makefile, /, sounds/Makefile: Merged revisions 247422 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
	  | 10 lines Tweak argument handling for wget in the sounds
	  Makefile. 1) Fix the check to see if we are using wget to not be
	  full of fail. The configure script populates this variable with
	  the absolute path to wget if it is found, so it didn't work. 2)
	  Allow some extra arguments to be passed in for wget. This is just
	  a simple change to allow our Bamboo build script to tell wget to
	  be quiet and not fill up our logs with download status output.
	  ........

2010-02-17 22:44 +0000 [r247335-247381]  Mark Michelson <mmichelson@digium.com>

	* main/test.c: Fix a couple of bugs in test tab completion. 1. Add
	  missing unlock of lists. 2. Swap order of arguments to
	  test_cat_cmp in complete_test_name.

	* main/test.c: Tab completion for test categories and names for
	  "test show registered" and "test execute" CLI commands.

	* main/strings.c, include/asterisk/strings.h: Fix two problems in
	  ast_str functions found while writing a unit test. 1. The
	  documentation for ast_str_set and ast_str_append state that the
	  max_len parameter may be -1 in order to limit the size of the
	  ast_str to its current allocated size. The problem was that the
	  max_len parameter in all cases was a size_t, which is unsigned.
	  Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the
	  max_len parameter to be ssize_t fixed this issue. 2. Once issue 1
	  was fixed, there was an off-by-one error in the case where we
	  attempted to write a string larger than the current allotted size
	  to a string when -1 was passed as the max_len parameter. When
	  trying to write more than the allotted size, the ast_str's
	  __AST_STR_USED was set to 1 higher than it should have been.
	  Thanks to Tilghman for quickly spotting the offending line of
	  code. Oh, and the unit test that I referenced in the top line of
	  this commit will be added to reviewboard shortly. Sit tight...

2010-02-17 19:51 +0000 [r247295]  Jeff Peeler <jpeeler@digium.com>

	* funcs/func_groupcount.c, tests/test_app.c (added), main/app.c,
	  CHANGES: Add support for GROUP_MATCH_COUNT regex matching on
	  category Current support for regex matching was previously only
	  available on the group. Also, error reporting for regex failures
	  has been added. In addition to this feature enhancement a unit
	  test has been written to check the regular expression logic to
	  ensure the count operation is working as expected. (closes issue
	  #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by
	  kobaz (license 834) Review:
	  https://reviewboard.asterisk.org/r/503/

2010-02-17 19:23 +0000 [r247248-247282]  David Vossel <dvossel@digium.com>

	* tests/test_devicestate.c: modified device2extension_test's
	  category

	* tests/test_devicestate.c (added): unit test for combined device
	  state mapping and device to exten state mapping Review:
	  https://reviewboard.asterisk.org/r/516/

	* main/features.c, CHANGES, configs/features.conf.sample: addition
	  of dynamic parkinglots feature This feature allows for
	  parkinglots to be created dynamically within the dialplan. Thanks
	  to all who were involved with getting this patch written and
	  tested! (closes issue #15135) Reported by: IgorG Patches:
	  features.dynamic_park.v3.diff uploaded by IgorG (license 20)
	  2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
	  dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested
	  by: eliel, IgorG, acunningham, mvanbaak, zktech Review:
	  https://reviewboard.asterisk.org/r/352/

2010-02-17 16:24 +0000 [r247169]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 247168 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
	  2010) | 3 lines Make sure that when autofill is disabled that
	  callers not in the front of the queue cannot place calls.
	  ........

2010-02-17 07:01 +0000 [r247124-247125]  Tilghman Lesher <tlesher@digium.com>

	* main/loader.c: RTP documentation states that you can pass NULL as
	  the module, so make sure that's really the case.

	* channels/sip/include/dialog.h (added), channels/chan_sip.c,
	  channels/sip/include/config_parser.h,
	  channels/sip/include/globals.h (added),
	  channels/sip/dialplan_functions.c (added), channels/Makefile,
	  channels/sip/include/sip_utils.h,
	  channels/sip/include/dialplan_functions.h (added): Make all of
	  the various rtpqos parameters in this branch available from the
	  CHANNEL function. Also includes a test for retrieving rtpqos
	  parameters, including a NULL RTP driver. Additionally, some
	  further separation of the SIP internal API into headers was
	  necessary. (closes issue #16652) Reported by: kkm Patches:
	  20100204__issue16652.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/501/

2010-02-16 23:44 +0000 [r247076]  Mark Michelson <mmichelson@digium.com>

	* main/strings.c: Add va_end calls to __ast_str_helper. According
	  to the man page for stdarg(3), "Each invocation of va_copy() must
	  be matched by a corresponding invocation of va_end() in the same
	  function." There were several cases in __ast_str_helper where
	  va_copy was not matched with a corresponding call to va_end.

2010-02-16 22:58 +0000 [r247035]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c: generate
	  connected line info update from info in h.323 packets Tested by:
	  benngard

2010-02-16 21:15 +0000 [r246985]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/strings.h: Add some clarifying documentation to
	  the ast_str_set and ast_str_append functions.

2010-02-16 21:03 +0000 [r246980-246981]  David Vossel <dvossel@digium.com>

	* main/tcptls.c: swap openssl with OpenSSL in warning message.
	  (issue #16673)

	* main/tcptls.c: warning message if openssl support is missing
	  while attempting tls connection (closes issue #16673) Reported
	  by: michaesc Patches: tls_error_msg.diff uploaded by dvossel
	  (license 671)

2010-02-16 18:29 +0000 [r246942]  Mark Michelson <mmichelson@digium.com>

	* tests/test_pbx.c (added): Add unit test for dialplan pattern
	  matching. This test works by reading input from arrays to build a
	  sample dialplan. From there, patterns are attempted to be matched
	  against said dialplan, with the expected match given. We then
	  search in our example dialplan to see if we find a match and if
	  what we find matches what we expected it to match. (closes issue
	  #16809) Reported by: lmadsen Tested by: mmichelson Review:
	  https://reviewboard.asterisk.org/r/504/

2010-02-16 17:07 +0000 [r246899]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes sample rate conversion issue with Monitor
	  application When using ast_seekstream with the read/write streams
	  of a monitor, the number of samples we are seeking must be of the
	  same rate as the stream or the jump calculation will be
	  incorrect. This patch adds logic to correctly convert the number
	  of samples to jump to the sample rate the read/write stream is
	  using. For example, if the call is G722 (16khz) and the
	  read/write stream is recording a 8khz wav, seeking 320 samples of
	  16khz audio is not the same as seeking 320 samples of 8khz audio
	  when performing the ast_seekstream on the stream. ABE-2044

2010-02-16 15:36 +0000 [r246710-246863]  Tilghman Lesher <tlesher@digium.com>

	* build_tools/cflags.xml, build_tools/cflags-devmode.xml: Revert
	  changes for now, pending discussion

	* build_tools/cflags-devmode.xml: Add a few more targets for
	  DEBUG_THREADLOCALS

	* build_tools/cflags.xml, channels/chan_usbradio.c,
	  build_tools/cflags-devmode.xml, main/strings.c,
	  apps/app_voicemail.c: Change the blanket rules to delete
	  .lastclean on all CFLAGS menuselect targets to be more
	  particular. This change builds upon the recent change to
	  menuselect to add 'touch_on_change' as an attribute of both
	  categories and members. This should allow only the most invasive
	  defines to cause a complete rebuild, while defines which only
	  affect a subset of modules will only cause a rebuild of that
	  smaller set.

	* channels/chan_sip.c: Allow Timer B to be set on the peer, and
	  ensure SIP rules are followed (or warn) in comparison to Timer
	  T1. (closes issue #16643) Reported by: nahuelgreco Patches:
	  20100204__issue16643.diff.txt uploaded by tilghman (license 14)
	  Tested by: oej

	* Makefile, /: Merged revisions 246709 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
	  | 5 lines Make the menuselect instructions correct by allowing
	  'make menuselect' to actually solve dependency problems.
	  (Previously, it would fail out again with the same message about
	  running 'make menuselect', which was NOT at all helpful.)
	  ........

2010-02-15 22:08 +0000 [r246669]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Restore triedtopribridge flag code removed
	  in -r211197. Ooops. Failed to note that we were inside a for loop
	  and pri_channel_bridge() needs to be executed only once.

2010-02-15 21:37 +0000 [r246667]  Tilghman Lesher <tlesher@digium.com>

	* utils/utils.xml: Instead of just automatically filtering out in
	  the Makefile, give an indication of dependencies in menuselect.

2010-02-15 15:45 +0000 [r246627]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/reqresp_parser.c,
	  channels/sip/include/sip_utils.h,
	  channels/sip/include/reqresp_parser.h: chan_sip parse code
	  refactoring plus two new unit tests Code Refactoring Changes -
	  read_to_parts() moved to reqresp_parser.c and has been renamed as
	  get_name_and_number() - get_in_brackets() moved to
	  reqresp_parser.c - find_closing_quotes() added to sip_utils.h
	  Logic Changes - get_name_and_number() now uses parse_uri() and
	  get_calleridname() for parsing. Before this change only names
	  within quotes were found, when names not within quotes are
	  possible. New Unit Tests -sip_get_name_and_number_test
	  -sip_get_in_brackets_test (closes issue #16707) Reported by:
	  Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license
	  671) Review: https://reviewboard.asterisk.org/r/499/

2010-02-12 23:32 +0000 [r246420-246546]  David Vossel <dvossel@digium.com>

	* main/channel.c, /: Merged revisions 246545 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
	  | 16 lines lock channel during datastore removal On channel
	  destruction the channel's datastores are removed and destroyed.
	  Since there are public API calls to find and remove datastores on
	  a channel, a lock should be held whenever datastores are removed
	  and destroyed. This resolves a crash caused by a race condition
	  in app_chanspy.c. (closes issue #16678) Reported by:
	  tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
	  tim ringenbach (license 540) Tested by: dvossel ........

	* channels/chan_sip.c: fixes areas where port should be removed
	  from domain during parsing A patch was committed recently that
	  converted duplicate uri parsing code to use the parse_uri
	  function. There were two instances where this conversion did not
	  mimic previous behavior exactly because the port was not being
	  parsed off the end of the domain. In order to do this, a dummy
	  pointer argument needs to be passed into parse_uri so it will
	  know it must parse out the port from the domain. If a port output
	  paramenter is not present, the domain is returned with the port
	  still attached.

2010-02-12 08:30 +0000 [r246382]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP
	  lookup application.

2010-02-11 21:57 +0000 [r246299-246338]  David Vossel <dvossel@digium.com>

	* tests/test_heap.c, tests/test_event.c,
	  channels/sip/reqresp_parser.c, channels/sip/config_parser.c:
	  fixes some test description formatting inconsistencies so log
	  file looks nice

	* tests/test_astobj2.c (added), main/astobj2.c: astobj2 unit test
	  and bug fix A bug was discovered during the creation of the
	  astobj2 unit test. When OBJ_MULTIPLE | OBJ_UNLINK is used, the
	  objects being returned had a ref count issue. This patch resolves
	  that. Review: https://reviewboard.asterisk.org/r/496/

2010-02-10 23:19 +0000 [r246260]  Russell Bryant <russell@digium.com>

	* include/asterisk/event.h, tests/test_event.c (added),
	  main/event.c: Add a test module for the event API, test_event.c.
	  This module includes a single test so far that creates events
	  using two different methods and does some verification on the
	  result to make sure the correct data can be retrieved from the
	  event that was created. One bug was found in the event API while
	  developing this test, which makes me happy. :-) Review:
	  https://reviewboard.asterisk.org/r/495/

2010-02-10 23:13 +0000 [r246249]  David Vossel <dvossel@digium.com>

	* channels/sip/reqresp_parser.c,
	  channels/sip/include/reqresp_parser.h: additional parse_uri test
	  and documentation

2010-02-10 21:55 +0000 [r246200-246208]  Tilghman Lesher <tlesher@digium.com>

	* res/res_pktccops.exports (added): res_pktccops needs to be able
	  to export a symbol for chan_mgcp (closes issue #16782) Reported
	  by: nahuelgreco Patches: res_pktccops.exports uploaded by
	  nahuelgreco (license 162)

	* funcs/func_strings.c: Fussy compiler on another machine...

	* funcs/func_strings.c: Fix weird issue with unit tests on
	  optimized build - turned out to be a signing issue.

2010-02-10 17:49 +0000 [r246116]  David Vossel <dvossel@digium.com>

	* /, apps/app_queue.c: Merged revisions 246115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
	  | 8 lines fixes random deadlock in app_queue with use_weight
	  during reload (closes issue #16677) Reported by: tim_ringenbach
	  Patches: app_queue_use_weight_deadlock.diff uploaded by tim
	  ringenbach (license 540) ........

2010-02-10 16:47 +0000 [r246070]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_local.c: Change channel state on local channels for
	  busy,answer,ring. Previously local channels channel state never
	  changed. This became problematic when the state of the other side
	  of the local channel was lost, for example during a masquerade.
	  Changing the state of the local channel allows for the scenario
	  to be detected when the channel state is set to ringing, but the
	  peer isn't ringing. The specific problem scenario is described in
	  164201. Although this was noted on one of the issues, here is the
	  tested dialplan verified to work: exten =>
	  9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
	  *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
	  exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
	  *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
	  not exten =>
	  0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
	  issue #14992) Reported by: davidw

2010-02-10 16:01 +0000 [r245945-246030]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  res/res_agi.c: Solaris doesn't like outputting a NULL to a %s in
	  format strings. Detect all platforms that don't like that,
	  either, and ensure that when documentation is missing, we pass a
	  non-NULL pointer when outputting the corresponding documentation.
	  (closes issue #16689) Reported by: bklang Patches:
	  20100209__issue16689__with_tests.diff.txt uploaded by tilghman
	  (license 14) Review: https://reviewboard.asterisk.org/r/497/

	* funcs/func_strings.c: Enable warnings on atypical conditions for
	  the FILTER function (suggested by mmichelson on the -dev list).

	* /, funcs/func_strings.c, configs/extensions.conf.sample: Merged
	  revisions 245944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
	  | 2 lines Include examples of FILTER usage in extension patterns
	  where a "." may be a risk. ........

2010-02-09 23:32 +0000 [r245864]  Russell Bryant <russell@digium.com>

	* include/asterisk/test.h, tests/test_sha1.c (removed),
	  include/asterisk/utils.h, tests/test_substitution.c,
	  tests/test_heap.c, tests/test_ast_format_str_reduce.c,
	  tests/test_skel.c, tests/test_utils.c, funcs/func_math.c,
	  channels/sip/reqresp_parser.c, main/test.c, tests/test_md5.c
	  (removed), channels/sip/config_parser.c, tests/test_sched.c:
	  Various updates to the unit test API. 1) It occurred to me that
	  the difference in usage between the error ast_str and the
	  ast_test_update_status() usage has turned out to be a bit
	  ambiguous in practice. In a lot of cases, the same message was
	  being sent to both. In other cases, it was only sent to one or
	  the other. My opinion now is that in every case, I think it makes
	  sense to do both; we should output it to the CLI as well as save
	  it off for logging purposes. This change results in most of the
	  changes in this diff, since it required changes to all existing
	  unit tests. It also allowed for some simplifications of unit test
	  API implementation code. 2) Update ast_test_status_update() to
	  include the file, function, and line number for the code
	  providing the update. 3) There are some formatting tweaks here
	  and there. Hopefully they aren't too distracting for code review
	  purposes. Reviewboard's diff viewer seems to do a pretty good job
	  of pointing out when something is a whitespace change. 4) I moved
	  the md5_test and sha1_test into the test_utils module. It seemed
	  like a better approach since these tests are so tiny. 5) I
	  changed the number of nodes used in heap_test_2 from 1 million to
	  100 thousand. The only reason for this was to reduce the time it
	  took for this test to run. 6) Remove an unused function prototype
	  that was at the bottom of utils.h. 7) Simplify test_insert()
	  using the LIST_INSERT_SORTALPHA() macro. The one minor difference
	  in behavior is that it no longer checks for a test registered
	  with the same name. 8) Expand the code in test_alloc() to provide
	  specific error messages for each failure case, to clearly inform
	  developers if they forget to set the name, summary, description,
	  etc. 9) Tweak the output of the "test show registered" CLI
	  command. I swapped the name and category to have the category
	  first. It seemed more natural since that is the sort key. 10)
	  Don't output the status ast_str in the "test show results" CLI
	  command. This is going to tend to be pretty verbose, so just
	  leave that for the detailed test logs (test generate results).
	  Review: https://reviewboard.asterisk.org/r/493/

2010-02-09 23:18 +0000 [r245793-245804]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes a merging error for the iaxs and
	  iaxsl off by one fix

	* /, channels/chan_iax2.c: Merged revisions 245792 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09
	  Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue.
	  2^15 = 32768 which is the maximum allowed iax2 callnumber.
	  Creating the iaxs and iaxsl array of size 32768 means the maximum
	  callnumber is actually out of bounds. This causes a nasty crash.
	  (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
	  uploaded by dvossel (license 671) ........

2010-02-09 18:06 +0000 [r245729]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_fax.c: Ensure frames are only freed once. (closes issue
	  #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt
	  uploaded by tilghman (license 14) Tested by: kenny, bloodoff,
	  misaksen

2010-02-09 17:40 +0000 [r245727]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: This commit removes an extra newline in T.38
	  generated SDP packets. This bug was caused by the fix introduced
	  in r243860. (closes issue #16766) Reported by: raivisr Patches:
	  t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
	  Tested by: raivisr

2010-02-09 16:24 +0000 [r245680]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Don't offer MMR or JBIG transcoding during T.38
	  negotiation. After further discussion with Steve Underwood, we
	  should not (yet) be offering to receive MMR or JBIG transcoded
	  streams from T.38 endpoints. A future spandsp release will
	  support those features, and then they can be enabled during
	  negotiation

2010-02-08 23:43 +0000 [r245597-245624]  Russell Bryant <russell@digium.com>

	* main/event.c: Fix return value of get_ie_str() and
	  get_ie_str_hash() for non-existent IE. I found this bug while
	  developing a unit test for event allocation. Testing is awesome.

	* tests/test_utils.c: UNREGISTER instead of REGISTER in
	  unload_module().

	* main/pbx.c: Use memmove() instead of memcpy() for a case where
	  the buffers overlap. Once again, valgrind is freaking awesome.
	  That is all.

	* channels/Makefile: Remove object files from the channels/sip/
	  directory on make clean.

2010-02-08 22:31 +0000 [r245578]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, channels/Makefile: Actually use _ASTLDFLAGS in the
	  main/ and channels/ Makefiles. They were previously passed
	  correctly, but they simply weren't used. This caused issues with
	  various platforms whose builds needed to pass special linker
	  flags via the configure script. (closes issue #16596) Reported
	  by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded
	  by pprindeville (license 347) Tested by: tilghman

2010-02-08 20:41 +0000 [r245497]  Jason Parker <jparker@digium.com>

	* /, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 245496
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
	  4 lines Remove reference of documentation in source directory.
	  People don't always build Asterisk from source (distro packages,
	  anybody?). ........

2010-02-08 04:51 +0000 [r245268-245385]  Russell Bryant <russell@digium.com>

	* contrib/scripts/install_prereq: Add the libvpb-dev package as a
	  dependency.

	* pbx/pbx_gtkconsole.c: Add a todo for pbx_gtkconsole for updating
	  to gtk2. This module needs to be converted to gtk2, or we will
	  eventually have to just remove it from the tree. gtk1 isn't even
	  packaged anymore in the distro I'm using. I suspect nobody uses
	  this and that nobody would notice if we removed it.

	* contrib/scripts/install_prereq: Add more packages required for
	  building Asterisk modules.

	* channels/chan_usbradio.c: Make chan_usbradio compile.

	* tests/test_sha1.c (added): Add a SHA1 test module. Review:
	  https://reviewboard.asterisk.org/r/492/

	* tests/test_md5.c: Remove unnecessary include, ast_md5_hash()
	  comes from utils.h.

	* tests/test_md5.c (added): Add an MD5 test module. Review:
	  https://reviewboard.asterisk.org/r/491/

	* tests/test_ast_format_str_reduce.c: Fix a couple of spelling
	  errors, and add format module dependencies.

	* channels/sip/include/config_parser.h, channels/sip/include/sip.h,
	  channels/sip/include/sip_utils.h,
	  channels/sip/include/reqresp_parser.h: Tweak formatting and add
	  minor updates to some comments.

	* main/test.c: Remove an extra space.

2010-02-07 19:51 +0000 [r245230]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Remove parsing of constantssrc from
	  reload_config. This config option is already handled by the
	  function handle_common_options and it is unnecessary to parse the
	  value again.

2010-02-06 14:43 +0000 [r245192]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Remove useless sip
	  options related to hash table size. First off, these options
	  weren't actually doing anything. By the time the options were
	  parsed, the peer and dialog containers had already been allocated
	  with their default values. Second, hash table size is something
	  that doesn't really make sense to change in a config file. If a
	  user is that interested in changing the hashtable size, he can
	  modify the source itself. I have removed the parsing of the
	  hash_peer, hash_user, and hash_dialog options. I have removed the
	  hash_user_size variable altogether since it is not used at all. I
	  also changed hash_peer_size and hash_dialog_size to be constant,
	  and have changed the symbols to be in all caps as constants
	  typically are. I have also removed the entire section in
	  sip.conf.sample regarding configurable hashtable sizes.

2010-02-05 21:21 +0000 [r245147]  David Vossel <dvossel@digium.com>

	* include/asterisk/astobj2.h, main/astobj2.c: fixes astobj2
	  unlinking of multiple objects when OBJ_MULTIPLE was disabled When
	  OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a
	  bucket were being unlinked instead of just the first match. This
	  fixes that. Review: https://reviewboard.asterisk.org/r/490/

2010-02-05 19:26 +0000 [r245090]  Jeff Peeler <jpeeler@digium.com>

	* /, LICENSE, contrib/firmware (removed): Merged revisions 245044
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
	  2010) | 5 lines Remove contrib/firmware directory as it is empty
	  Remove explicit license for IAXy firmware as it is no longer
	  included in the tree ........

2010-02-05 19:07 +0000 [r245046]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_ast_format_str_reduce.c, main/file.c: Merge tests that
	  verify the same thing. (Oops.)

2010-02-05 18:12 +0000 [r245006]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: adds total call numbers available to 'iax2
	  show callnumber usage' cli output

2010-02-05 17:20 +0000 [r244945]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  res/res_calendar_caldav.c: Fix crash on 32-bit for users not
	  using https (closes issue #16778) Reported by: pitel Patches:
	  diff.txt uploaded by twilson (license 396) Tested by: twilson,
	  pitel

2010-02-05 17:05 +0000 [r244927]  Sean Bright <sean@malleable.com>

	* /, main/asterisk.c: Merged revisions 244926 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
	  2010) | 1 line Update main copyright date. ........

2010-02-05 16:59 +0000 [r244769-244924]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/include/config_parser.h,
	  channels/sip/config_parser.c: fixes issue with sip registry not
	  having correct default expiry default expiry was not being set
	  correctly for a registry object. Thanks to ebroad for reporting
	  the issue and testing the patch.

	* main/astobj2.c: fixes memory leak in astobj2 test
	  ao2_iterator_destroy was not being used on the iterator during
	  the test. This resulted in the container never actually being
	  destroyed.

	* channels/chan_sip.c: parse_moved_contact tries to parse
	  contact_name twice parse_moved_contact attempts to remove a
	  quoted string twice, and the first try wasn't even being done
	  correctly.

2010-02-04 22:43 +0000 [r244728-244768]  Tilghman Lesher <tlesher@digium.com>

	* main/file.c: Try to make ast_format_str_reduce fail...

	* include/asterisk/manager.h: Oops

	* include/asterisk/manager.h: Define a small set of constant return
	  values

2010-02-04 15:36 +0000 [r244688]  David Vossel <dvossel@digium.com>

	* main/test.c: fix truncated format string in 'test show
	  registered' When using the 'test show registered' cli command the
	  'Test Results' category was truncating the last few characters
	  making it look like 'Test Resul'. I also expanded other parts of
	  the format to better represent how long function names and
	  categories will likely be.

2010-02-04 00:12 +0000 [r244647]  Richard Mudgett <rmudgett@digium.com>

	* channels/sip: Add ignore *.i files property to the new
	  channels/sip directory.

2010-02-03 20:48 +0000 [r244598]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c, CHANGES: Add some additional option support for
	  non-default parking lots. The options are: parkedcallparking,
	  parkedcallhangup, parkedcallrecording, and parkedcalltransfers.
	  Previously these options were only available for the default
	  parking lot. (closes issue #16641) Reported by: bluecrow76
	  Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76
	  (license 270)

2010-02-03 20:33 +0000 [r244597]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, channels/sip/include/config_parser.h
	  (added), channels/sip/reqresp_parser.c (added), channels/sip
	  (added), channels/Makefile, channels/sip/config_parser.c (added),
	  channels/sip/include (added), channels/sip/include/sip.h (added),
	  channels/sip/include/sip_utils.h (added),
	  channels/sip/include/reqresp_parser.h (added): -----Changes -----
	  New files - channels/sip/sip.h – A new header for shared #define,
	  enum, and struct definitions. - channels/sip/include/sip_utils.h
	  – sip util functions shared among the all the sip APIs -
	  channels/sip/include/config_parser.h – sip config-parser API -
	  channels/sip/config_parser.c – Contains sip.conf parsing helper
	  functions with unit tests. -
	  channels/sip/include/reqresp_parser.h – sip request response
	  parser API - channels/sip/reqresp_parser.c – Contains sip request
	  and response parsing helper functions with unit tests. New Unit
	  Tests - sip_parse_uri_test - sip_parse_host_test -
	  sip_parse_register_line_test Code Refactoring - All reusable
	  #define, enum, and struct definitions were moved out of
	  chan_sip.c into sip.h. During this process formatting changes
	  were made to comments in both sip.h and chan_sip.c in order to
	  better adhere to the coding guidelines. - The beginnings of three
	  new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h
	  using existing chan_sip.c functions. - parse_uri() and
	  get_calleridname() were moved from chan_sip.c to request-parser.c
	  along with unit tests for both functions. - sip_parse_host() and
	  sip_parse_register_line() were moved from chan_sip.c to
	  config-parser.c along with unit tests for both functions. Changes
	  to parse_uri() -removal of the options parameter. It was never
	  used and did not behave correctly. -additional check for
	  [?header] field. When this field was present, the transport type
	  was not being set correctly. ----- Overview ----- This patch is
	  introduced with the hope that unit tests for all our sip parsing
	  functions will be written soon. chan_sip is a huge file, and with
	  the addition of each unit test chan_sip is going to grow larger
	  and harder to maintain. I'm proposing we begin refactoring
	  chan_sip, starting with the parsing functions. With each parsing
	  function we move into a separate helper file, a unit test should
	  accompany it. I've attempted to lay down the ground work for this
	  change by creating two new parser helper files (config-parser.c
	  and reqresp-parser.c) and moving all shared structs, enums, and
	  defines from chan_sip.c into a shared sip.h file. We can't verify
	  everything in Asterisk using unit tests, but string parsing is
	  one area where unit tests make the most sense. By beginning to
	  restructure the code in this way, chan_sip not only becomes less
	  bloated, but Asterisk as a whole will become more stable. Review:
	  https://reviewboard.asterisk.org/r/477/

2010-02-03 19:26 +0000 [r244547]  Mark Michelson <mmichelson@digium.com>

	* main/sched.c: Initialize counters in ast_sched_report so that
	  resulting data is not bogus.

2010-02-03 18:34 +0000 [r244505]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: The chanvar= setting should inherit the
	  entire list of variables, not just the first one. (closes issue
	  #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded
	  by raarts (license 937) Tested by: raarts

2010-02-02 22:27 +0000 [r244443]  David Vossel <dvossel@digium.com>

	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
	  fixes crash during T.38 negotiation caused by invalid or missing
	  FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
	  by: krn (closes issue #16724) Reported by: barthpbx (closes issue
	  #16517) Reported by: bklang (closes issue #16485) Reported by:
	  elsto

2010-02-02 20:32 +0000 [r244071-244393]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, CHANGES: Properly respect GOSUB_RESULT as to
	  what to do with the master channel. Previously, we would parse
	  GOSUB_RESULT, but not actually do anything with it. Also, allow
	  GOSUB_RETVAL to be inherited back across a peer/master channel.
	  (closes issue #16687) Reported by: bklang Patches:
	  app_dial-preserve-gosub_retval.patch uploaded by bklang (license
	  919) (with modifications) (closes issue #16686) Reported by:
	  bklang Patches: app_dial-respect-gosub_result.patch uploaded by
	  bklang (license 919) (with modifications)

	* funcs/func_math.c: Correct some off-by-one errors, especially
	  when expressions don't contain expected spaces. Also include the
	  tests provided by the reporter, as regression tests. (closes
	  issue #16667) Reported by: wdoekes Patches:
	  astsvn-func_match-off-by-one.diff uploaded by wdoekes (license
	  717)

	* /, apps/app_voicemail.c: Merged revisions 244242 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01
	  Feb 2010) | 11 lines Backup and restore original textfile, for
	  prosthesis (gerund of prepend). Also, fix menuselect such that
	  changing voicemail build options correctly causes rebuild.
	  (closes issue #16415) Reported by: tomo1657 Patches:
	  prepention.patch uploaded by tomo1657 (license 484) (with
	  modifications by me to backport to 1.4) ........

	* main/channel.c, channels/chan_local.c, /: Merged revisions 244070
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010)
	  | 16 lines Revert previous chan_local fix (r236981) and fix
	  instead by destroying expired frames in the queue. (closes issue
	  #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt
	  uploaded by tilghman (license 14)
	  20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
	  (license 14) Tested by: kobaz, atis (closes issue #16581)
	  Reported by: ZX81 (closes issue #16681) Reported by: alexr1
	  ........

2010-01-28 22:37 +0000 [r243986]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: Optimization to manager events. When potentially
	  sending manager events, return immediately if there are no
	  sessions or hooks. Also, avoid locking the hooks list if it is
	  empty. (issue #16455) Reported by: atis Patches:
	  manager_hooks_trunk.patch uploaded by atis (license 242)

2010-01-28 20:00 +0000 [r243943]  Tilghman Lesher <tlesher@digium.com>

	* channels/iax2-parser.c: Informational message, not an error.

2010-01-28 18:35 +0000 [r243780-243860]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Add a missing line terminator for T.38 SDP.

	* /, channels/chan_sip.c: Merged revisions 243779 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010)
	  | 2 lines Fix a bogus third argument to ast_copy_string().
	  ........

2010-01-27 20:37 +0000 [r243551-243693]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_queue.c: Merged revisions 243691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010)
	  | 5 lines Revert 243570, I should have looked at this closer.
	  Will reopen the issue, but am leaving the review closed as the
	  change was pointless. (issue #16488) ........

	* CHANGES: expand code based appreviation of AST_CONFIG_DIR to
	  configuration directory

	* /, apps/app_queue.c: Merged revisions 243570 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010)
	  | 9 lines Extend announcement URL used with Queue from 80 chars
	  to PATH_MAX. (closes issue #16488) Reported by: syspert Patches:
	  soundfilelen.pacth-2 uploaded by syspert (license 938) Review:
	  https://reviewboard.asterisk.org/r/475/ ........

	* Makefile, CHANGES, include/asterisk/options.h, main/asterisk.c,
	  main/loader.c: Add new option to asterisk.conf (lockconfdir) to
	  protect conf dir during reloads (closes issue #16358) Reported
	  by: raarts Patches: lockconfdir.diff uploaded by raarts (license
	  937) modified by me

2010-01-27 18:08 +0000 [r243487]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 243486 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan
	  2010) | 3 lines Use a safe list traversal while checking for
	  duplicate vars in pbx_builtin_setvar_helper. ........

2010-01-27 17:32 +0000 [r243482]  Russell Bryant <russell@digium.com>

	* funcs/func_channel.c, channels/chan_iax2.c: Fix the ability to
	  specify an OSP token for an outbound IAX2 call. When this patch
	  was originally submitted, the code allowed for the token to be
	  set via a channel variable. I decided that a cleaner approach
	  would be to integrate it into the CHANNEL() function.
	  Unfortunately, that is not a suitable approach. It's not possible
	  to get the value set on the channel soon enough using that
	  method. So, go back to the simple channel variable method.
	  (closes issue #16711) Reported by: homesick Patches: iax-svn.diff
	  uploaded by homesick (license 91)

2010-01-26 23:56 +0000 [r243391]  David Vossel <dvossel@digium.com>

	* /, main/features.c: Merged revisions 243390 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010)
	  | 9 lines fixes bug with channel receiving wrong privileges after
	  call parking (closes issue #16429) Reported by: Yasuhiro Konishi
	  Patches: features.c.diff uploaded by Yasuhiro Konishi (license
	  947) Tested by: dvossel ........

2010-01-26 20:49 +0000 [r243346]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_senddtmf.c: Code clean up in app_senddtmf Pushes code
	  clean up done in app_externalivr back into app_senddtmf Review:
	  https://reviewboard.asterisk.org/r/473/

2010-01-26 18:20 +0000 [r243244-243266]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 243258 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010)
	  | 2 lines Remove unnecessary code in ast_read as issue 16058 has
	  been fully solved now. ........

	* main/frame.c: Fix crash resulting from frames with invalid data
	  pointers. In ast_frdup the frame data union does not get set to
	  point to malloced memory if the datalen is zero, so make sure to
	  handle the same case in ast_frisolate appropriately. (closes
	  issue #16058) Reported by: atis Patches: bug16058-fix.patch
	  uploaded by jpeeler (license 325) Tested by: atis

2010-01-26 17:40 +0000 [r243200-243242]  David Vossel <dvossel@digium.com>

	* main/test.c: modify 'test show registered' cli output format In
	  order to improve readability, the output from 'test show
	  registered' has been modified to truncate fields to fit within
	  the format output if they are over a certain length.

	* include/asterisk/utils.h, channels/chan_sip.c, tests/test_utils.c
	  (added), main/test.c, main/utils.c: RFC compliant uri and
	  display-name encode/decode 1. URI Encoding This patch changes
	  ast_uri_encode()'s behavior when doreserved is enabled.
	  Previously when doreserved was enabled only a small set of
	  reserved characters were encoded. This set was comprised
	  primarily of the reserved characters defined in RFC3261 section
	  25.1, but contained other characters as well. Rather than only
	  escaping the reserved set, doreserved now escapes all characters
	  not within the unreserved set as defined by RFC 3261 and RFC
	  2396. Also, the 'doreserved' variable has been renamed to
	  'do_special_char' in attempts to avoid confusion. When doreserve
	  is not enabled, the previous logic of only encoding the
	  characters <= 0X1F and > 0X7f remains, except for the '%'
	  character, which must always be encoded as it signifies a HEX
	  escaped character during the decode process. 2. URI Decoding:
	  Break up URI before decode. In chan_sip.c ast_uri_decode is
	  called on the entire URI instead of it's individual parts after
	  it is parsed. This is not good as ast_uri_decode can introduce
	  special characters back into the URI which can mess up parsing.
	  This patch resolves this by not decoding a URI until parsing is
	  completely done. There are many instances where we check to see
	  if pedantic checking is enabled before we decode a URI. In these
	  cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual
	  parsed segments of the URI rather than constantly putting if
	  (pedantic) { decode() } checks everywhere in the code. In the
	  areas where ast_uri_decode is not dependent upon pedantic
	  checking this macro is not used, but decoding is still moved to
	  each individual part of the URI. The only behavior that should
	  change from this patch is the time at which decoding occurs.
	  Since I had to look over every place URI parsing occurs to create
	  this patch, I found several places where we use duplicate code
	  for parsing. To consolidate the code, those areas have updated to
	  use the parse_uri() function where possible. 3. SIP display-name
	  decoding according to RFC3261 section 25. To properly decode the
	  display-name portion of a FROM header, chan_sip's
	  get_calleridname() function required a complete re-write. More
	  information about this change can be found in the comments at the
	  beginning of this function. 4. Unit Tests. Unit tests for
	  ast_uri_encode, ast_uri_decode, and get_calleridname() have been
	  written. This involved the addition of the test_utils.c file for
	  testing the utils api. (closes issue #16299) Reported by: wdoekes
	  Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes
	  (license 717) get_calleridname_rewrite.diff uploaded by dvossel
	  (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review:
	  https://reviewboard.asterisk.org/r/469/

2010-01-26 15:46 +0000 [r243118-243158]  Russell Bryant <russell@digium.com>

	* tests/test_substitution.c: Log the variable name being tested.

	* tests/test_substitution.c: Update test_substitution to show
	  failures in the test log.

	* funcs/func_aes.c: Update func_aes to its pre-ast_str_substitution
	  state. This change makes the AES tests in test_substitution.c
	  pass. We still need to work through what's going wrong in the
	  ast_str version.

2010-01-26 01:56 +0000 [r242967-243077]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_substitution.c: Fixing last errors in the conversion,
	  though it appears that the AES_* functions are still broken.

	* tests/test_substitution.c: Using a dummy channel causes CDR()
	  testing to fail.

	* tests/test_substitution.c: Wish I had gotten to the review before
	  this got submitted, because there's failures we need to address.

	* /, main/Makefile, res/Makefile: Merged revisions 242969 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010)
	  | 2 lines Err, and use the new menuselect define, too. ........

	* build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
	  configure, configure.ac: Merged revisions 242966 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25
	  Jan 2010) | 2 lines Only rebuild parsers by an option in
	  menuselect ........

2010-01-25 21:32 +0000 [r242954-242965]  Russell Bryant <russell@digium.com>

	* tests/test_substitution.c, tests/test_heap.c,
	  tests/test_ast_format_str_reduce.c, tests/test_skel.c,
	  tests/test_sched.c: Make unit test modules depend on
	  TEST_FRAMEWORK instead of off by default.

	* tests/test_substitution.c: Convert test_substitution module to
	  the unit test API. Review:
	  https://reviewboard.asterisk.org/r/474/

2010-01-25 21:20 +0000 [r242933]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooCalls.c: small corrections in call clearing

2010-01-25 21:13 +0000 [r242904-242919]  Olle Johansson <oej@edvina.net>

	* main/pbx.c, main/manager.c, include/asterisk/pbx.h: Change api
	  for pbx_builtin_setvar to actually return error code if a
	  function can't be written to. This patch removes code that was
	  duplicated from pbx.c to manager.c in order to prevent API change
	  in released versions of Asterisk. There are propably also other
	  places that would benefit from reading the return code and react
	  if a function returns error codes on writing a value into it.

	* main/manager.c, /: Merged revisions 242850 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2
	  lines Report error when writing to functions returns error in AMI
	  setvar action ........

2010-01-25 20:18 +0000 [r242857]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, main/Makefile, configure.ac, res/Makefile: Merged
	  revisions 242852 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010)
	  | 2 lines Restore FreeBSD to able-to-compile-ish-mode ........

2010-01-25 18:01 +0000 [r242812]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: Fix INTERNAL_OBJ error on stop when
	  calendars.conf missing Initialize the calendars container before
	  calling load_config and return FAILURE on allocation failure.
	  Also, use the AST_MODULE_LOAD_* values for return values. Thanks
	  to rmudgett for pointing out the error and the need to use the
	  defined values for return

2010-01-25 05:45 +0000 [r242719-242729]  Tilghman Lesher <tlesher@digium.com>

	* /, main/Makefile, res/Makefile: Merged revisions 242728 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010)
	  | 2 lines Buildbot pointed out an error (thanks, buildbot!)
	  ........

	* /, res/Makefile: Merged revisions 242723 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010)
	  | 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for
	  the commands. ........

	* /, main/Makefile: Merged revisions 242683 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010)
	  | 2 lines Make the build of the Asterisk expression parser match
	  that of the AEL parser. ........

2010-01-24 22:42 +0000 [r242645]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooStackCmds.h,
	  addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooCmdChannel.c,
	  addons/ooh323c/src/ooStackCmds.c: AST_CONTROL_CONNECTED_LINE
	  frame type processing added to setup DisplayIE field incorrect
	  q.931 message order filtered on incoming calls (first msg must be
	  setup, next must be not setup)

2010-01-24 21:49 +0000 [r242607]  Sean Bright <sean@malleable.com>

	* res/res_phoneprov.c: Instead of crashing, allocate our header
	  ast_str before we try to use it. (closes issue #16680) Reported
	  by: lmadsen Patches: issue16680_20100122.patch uploaded by
	  seanbright (license 71) Tested by: lmadsen

2010-01-24 06:40 +0000 [r242521]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  pbx/Makefile, res/Makefile, makeopts.in: Merged revisions 242520
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010)
	  | 8 lines Only rebuild bison and flex source files on demand, if
	  bison and flex are detected by the configure script. Changed
	  after discussion on the -dev list about possible unnecessary
	  build failures, due to checkouts/untars causing these special
	  source files to possibly be newer than their resulting C files.
	  This should additionally ensure that nobody need learn about
	  extra Makefile arguments to ensure the proper files get rebuilt
	  when changes are made to these special source files. ........

2010-01-22 21:45 +0000 [r242424]  Tilghman Lesher <tlesher@digium.com>

	* /, res/Makefile: Merged revisions 242423 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010)
	  | 7 lines Rebuild from flex, bison sources when necessary. (issue
	  #14629) Reported by: Marquis Patches:
	  20100121__issue14629.diff.txt uploaded by tilghman (license 14)
	  ........

2010-01-22 16:20 +0000 [r242357]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c: Add send DTMF feature to ExternalIVR app
	  Implemented a new command 'D' that allows client IVRs to send
	  DTMF digits to the channel. (closes issue #16615) Reported by:
	  thedavidfactor Review: https://reviewboard.asterisk.org/r/465/

2010-01-22 15:09 +0000 [r242317]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_sched.c: The irony of not compile-testing a test
	  program before committing is killing me.

2010-01-22 09:28 +0000 [r242227]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 242226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3
	  lines Initialize notify_types to NULL ........

2010-01-22 04:57 +0000 [r242184-242186]  Russell Bryant <russell@digium.com>

	* main/test.c: Update the doxygenification of some comments.

	* tests/test_sched.c: Convert scheduler API entry order test to the
	  test API. Review: https://reviewboard.asterisk.org/r/470/

	* tests/test_skel.c: Add test API usage example to test_skel.c.
	  Review: https://reviewboard.asterisk.org/r/471/

2010-01-21 22:37 +0000 [r242092]  Mark Michelson <mmichelson@digium.com>

	* main/acl.c: Add missing argument to ast_calloc calls.

2010-01-21 21:05 +0000 [r242043]  Olle Johansson <oej@edvina.net>

	* main/acl.c: Make sure we initialize the ast_ha structure with
	  ast_calloc

2010-01-21 15:27 +0000 [r241938]  Sean Bright <sean@malleable.com>

	* /, configure, configure.ac: Merged revisions 241932 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu,
	  21 Jan 2010) | 5 lines Fix configure check for PTHREAD_ONCE_INIT
	  when manually adding -Wall to CFLAGS. (closes issue #16666)
	  Reported by: romain_proformatique ........

2010-01-21 15:14 +0000 [r241896]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_vpb.cc: Formats are inconsistent between even
	  32-bit and 64-bit Linux. Use casts to ensure both compile.

2010-01-21 14:10 +0000 [r241855-241856]  Russell Bryant <russell@digium.com>

	* main/test.c: Point to a useful reference on the XML output
	  format.

	* main/test.c: Modify test results XML format to match the JUnit
	  format. When this code was developed, we came up with our own XML
	  format for the test output. I have since started looking at
	  integration with other tools, namely continuous integration
	  frameworks, and this format seems to be supported across a number
	  of applications. With these changes in place, I was able to get
	  Atlassian Bamboo to interpret the test results.

2010-01-21 05:54 +0000 [r241766]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_math.c: Merged revisions 241765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010)
	  | 2 lines Guard against division by zero. ........

2010-01-20 21:14 +0000 [r241627-241714]  David Vossel <dvossel@digium.com>

	* res/res_rtp_asterisk.c: rtp timestamp to timeval calculation fix
	  The rtp timestamp to timeval calculation was only accurate for
	  8kHz audio. This patch corrects this. Review:
	  https://reviewboard.asterisk.org/r/468/ SWP-648

	* Makefile, /: Merged revisions 241626 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010)
	  | 6 lines fixes parsing error in Makefile. Some echo lines were
	  missing "; . Thanks to jparker for pointing out the problem.
	  ........

2010-01-20 17:49 +0000 [r241581]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/cdr.c: Add Calling and Called Subaddress to CDR record
	  Requires 'callingsubaddr' and 'calledsubaddr' fields in backend
	  cdr. (closes issue #16600) Reported by: alecdavis Patches:
	  cdr_subaddr.diff.txt uploaded by alecdavis (license 585) Tested
	  by: alecdavis Review: https://reviewboard.asterisk.org/r/460/

2010-01-20 13:01 +0000 [r241503]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_vpb.cc: Fix up compile breakage from
	  ast_tvdiff_ms() API change.

2010-01-20 08:18 +0000 [r241416]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/pbx.c, channels/sig_pri.c: Update CDR variables as pbx
	  starts Allows CDR variables added in cdr.c:set_one_cid to become
	  visable during the call, by executing ast_cdr_update() early in
	  __ast_pbx run. Reverts sig_pri changes in trunk that are specific
	  to isdn technology only. (closes issue #16638) Reported by:
	  alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis

2010-01-19 22:59 +0000 [r241366]  Jeff Peeler <jpeeler@digium.com>

	* main/pbx.c: Initialize data on the stack so that Park doesn't
	  interpret random arguments. passdata was only being set in
	  pbx_substitue_variables when arguments were passed. (closes issue
	  #16406) (closes issue #16586) Reported by: DLNoah Patches:
	  bug16586v2.patch uploaded by jpeeler (license 325) Tested by:
	  DLNoah

2010-01-19 22:41 +0000 [r241364]  Tilghman Lesher <tlesher@digium.com>

	* doc/janitor-projects.txt, apps/app_sendtext.c: Enable SendText to
	  send strings in encoded format. See
	  http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html

2010-01-19 18:51 +0000 [r241314-241315]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_agent.c: small correction from 241314

	* /, channels/chan_agent.c: Merged revisions 241227 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19
	  Jan 2010) | 13 lines Fix deadlock in agent_read by removing call
	  to agent_logoff. One must always lock the agents list lock before
	  the agent private. agent_read locks the private immediately, so
	  locking the agents list lock is not an option (which is what
	  agent_logoff requires). Because agent_read already has access to
	  the agent private all that is necessary is to do the required
	  hanging up that agent_logoff performed. (closes issue #16321)
	  Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler
	  (license 325) ........

2010-01-19 17:42 +0000 [r241230]  Jason Parker <jparker@digium.com>

	* Makefile: Allow parallel make (-j) to work properly. After some
	  back and forth with the reporter, we came up with the necessary
	  changes. (closes issue #16489) Reported by: Chainsaw Patches:
	  asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw
	  (license 723) Tested by: Chainsaw, qwell

2010-01-19 00:28 +0000 [r241188]  Tilghman Lesher <tlesher@digium.com>

	* main/srv.c, res/res_agi.c, CHANGES, include/asterisk/srv.h:
	  Create iterative method for querying SRV results, and use that
	  for finding AGI servers. (closes issue #14775) Reported by:
	  _brent_ Patches: 20091215__issue14775.diff.txt uploaded by
	  tilghman (license 14) hagi-5.patch uploaded by brent (license
	  388) Tested by: _brent_ Reviewboard:
	  https://reviewboard.asterisk.org/r/378/

2010-01-19 00:24 +0000 [r241187]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sig_pri.c: Update CDR variables before pbx starts
	  (overlap dial) Allows CDR variables added in cdr.c:set_one_cid to
	  become visable during the call. (issue #16638) Reported by:
	  alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis

2010-01-18 22:31 +0000 [r241143]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c,
	  main/features.c, pbx/pbx_dundi.c, main/enum.c,
	  include/asterisk/time.h, main/timing.c: Extend max call limit
	  duration from 24.8 days to 292+ million years. If the limit was
	  set past MAX_INT upon answering, the call was immediately hung up
	  due to overflow from the return of ast_tvdiff_ms (in
	  ast_check_hangup). The time calculation functions ast_tvdiff_sec
	  and ast_tvdiff_ms have been changed to return an int64_t to
	  prevent overflow. Also the reporter suggested adding a message
	  indicating the reason for the call hanging up. Given that the new
	  limit is so much higher, the message (which would only really be
	  useful in the overflow scenario) has been made a debug message
	  only. (closes issue #16006) Reported by: viraptor

2010-01-18 22:03 +0000 [r241098]  Jason Parker <jparker@digium.com>

	* main/rtp_engine.c: Fix an RTP instance allocation failure on
	  Solaris. (closes issue #16543) Reported by: crjw Patches:
	  rtp_sin_family.patch uploaded by crjw (license 963) Tested by:
	  crjw, qwell

2010-01-18 22:00 +0000 [r241097]  Alec L Davis <sivad.a@paradise.net.nz>

	* channels/sig_pri.c: Update CDR variables before pbx starts Allows
	  CDR variables added in cdr.c:set_one_cid to become visable during
	  the call. (closes issue #16638) Reported by: alecdavis Patches:
	  cdr_update.diff.txt uploaded by alecdavis (license 585)

2010-01-18 19:57 +0000 [r241016]  Sean Bright <sean@malleable.com>

	* /, main/config.c: Merged revisions 241015 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan
	  2010) | 12 lines Plug a memory leak when reading configs with
	  their comments. While reading through configuration files with
	  the intent of returning their full contents (comments
	  specifically) we allocated some memory and then forgot to free
	  it. This doesn't fix 16554 but clears up a leak I had in the lab.
	  (issue #16554) Reported by: mav3rick Patches:
	  issue16554_20100118.patch uploaded by seanbright (license 71)
	  Tested by: seanbright ........

2010-01-18 19:26 +0000 [r241012]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c, CHANGES: Make HASHes inheritable across
	  channel creation.

2010-01-18 18:00 +0000 [r240973-240974]  David Ruggles <thedavidfactor@gmail.com>

	* UPGRADE.txt: ExternalIVR information for UPGRADE.txt added a
	  paragraph about the fixes and changes to the ExternalIVR
	  application.

	* doc/externalivr.txt: Updated ExternalIVR documentation Rewrote a
	  large portion of the existing documentation and added information
	  about the TCP/IP socket interface

2010-01-18 17:45 +0000 [r240971]  David Vossel <dvossel@digium.com>

	* Makefile, CHANGES: transmit_silence_during_record replaced by
	  transmit_silence In asterisk.conf, transmit_silence_during_record
	  has been removed in favor of using only the transmit_silence
	  option. The transmit_silence_during_record option remains a valid
	  option in asterisk.conf, but has been removed from the sample
	  config and noted in CHANGES.

2010-01-18 17:41 +0000 [r240969]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c: Add notification of interrupted file Add
	  file information to data element of T event so the file
	  information is sent to the client when it is interrupted.
	  Previously only notification of pending files that were dropped
	  was sent (closes issue #16147) Reported by: thedavidfactor Tested
	  by: thedavidfactor Review:
	  https://reviewboard.asterisk.org/r/449/

2010-01-18 16:45 +0000 [r240842-240887]  David Vossel <dvossel@digium.com>

	* Makefile: updated transmit_silence option documentation in
	  asterisk.conf This patch updates the transmit_silence option to
	  better document why the option exists, and what it affects.
	  Thanks to russell for providing the verbage for this update.

	* apps/app_queue.c: fixes spelling error. s/memeber/member

2010-01-17 19:45 +0000 [r240717]  Sean Bright <sean@malleable.com>

	* main/pbx.c: Avoid a crash on Solaris when running 'core show
	  functions.' (closes issue #16309) Reported by: asgaroth

2010-01-16 00:54 +0000 [r240667]  Sean Bright <sean@malleable.com>

	* res/res_musiconhold.c: Get MoH building on OpenSolaris.

2010-01-15 23:50 +0000 [r240629]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, main/asterisk.c: Err, oops, it was already the way I
	  intended.

2010-01-15 23:09 +0000 [r240548-240552]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxygen/commits.h: Note where empty lines should
	  reside in commit messages.

	* Makefile, /: Merged revisions 240547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r240547 | russell | 2010-01-15 17:06:11 -0600 (Fri, 15 Jan 2010)
	  | 2 lines Fix a spelling error in the asterisk.conf sample.
	  ........

2010-01-15 22:07 +0000 [r240505]  Sean Bright <sean@malleable.com>

	* res/res_timing_timerfd.c: Clarify error message in
	  res_timing_timerfd.

2010-01-15 21:42 +0000 [r240421-240500]  Tilghman Lesher <tlesher@digium.com>

	* utils/astcanary.c: Oops, missed an include

	* utils/astcanary.c, main/asterisk.c: The previous attempt at using
	  a pipe to guarantee astcanary shutdown did not work. We're
	  revisiting the previous patch, albeit with a method that
	  overcomes the prior criticism that it was not POSIX-compliant.
	  (closes issue #16602) Reported by: frawd Patches:
	  20100114__issue16602.diff.txt uploaded by tilghman (license 14)
	  Tested by: frawd

	* apps/app_directed_pickup.c, main/features.c,
	  include/asterisk/manager.h: Add pickup event to AMI. Also, fix
	  AMI documentation. (closes issue #16431) Reported by: syspert
	  Patches: 20100112__issue16431.diff.txt uploaded by tilghman
	  (license 14)

2010-01-15 20:58 +0000 [r240420]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: Make sure to set owner_line, ownder_func, and
	  owner_file in ast_calloc_with_stringfields. Asterisk would crash
	  on startup if MALLOC_DEBUG were set in menuselect. This is
	  because the manager action UpdateConfig had to resize its string
	  field allocation to set the description. When the resize
	  occurred, ast_copy_string would crash because we were attempting
	  to copy a string from a NULL pointer. Setting the strings
	  initially makes the code much less crashy.

2010-01-15 20:58 +0000 [r240415-240419]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Make sure that the limit is N, not N - 1.

	* /, apps/app_voicemail.c: Merged revisions 240414 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15
	  Jan 2010) | 15 lines Disallow leaving more than maxmsg
	  voicemails. This is a possibility because our previous method
	  assumed that no messages are left in parallel, which is not a
	  safe assumption. Due to the vmu structure duplication, it was
	  necessary to track in-process messages via a separate structure.
	  If at some point, we switch vmu to an ao2-reference-counted
	  structure, which would eliminate the prior noted duplication of
	  structures, then we could incorporate this new in-process
	  structure directly into vmu. (closes issue #16271) Reported by:
	  sohosys Patches: 20100108__issue16271.diff.txt uploaded by
	  tilghman (license 14) 20100108__issue16271__trunk.diff.txt
	  uploaded by tilghman (license 14)
	  20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
	  (license 14) Tested by: jsutton ........

2010-01-15 20:41 +0000 [r240411]  Russell Bryant <russell@digium.com>

	* main/event.c: Ensure payload type is properly checked when
	  comparing against cached events. (closes issue #16607) Reported
	  by: ddv2005 Patches: event.patch uploaded by ddv2005 (license
	  769)

2010-01-15 18:21 +0000 [r240368]  Sean Bright <sean@malleable.com>

	* main/pbx.c, main/manager.c, res/res_smdi.c, apps/app_meetme.c,
	  channels/chan_sip.c, cel/cel_tds.c, main/features.c,
	  res/res_phoneprov.c, cdr/cdr_tds.c, apps/app_jack.c: Convert a
	  few places to use ast_calloc_with_stringfields where applicable.

2010-01-15 16:51 +0000 [r240329]  Russell Bryant <russell@digium.com>

	* configure: Update configure script for an OSP toolkit related
	  change.

2010-01-15 16:28 +0000 [r240328]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/sip.conf.sample: Clarify RTP NAT handling a bit.

2010-01-14 23:13 +0000 [r240226-240271]  Sean Bright <sean@malleable.com>

	* res/res_config_ldap.c: Plug a memory leak in res_config_ldap.
	  (closes issue #16257) Reported by: nito Patches:
	  issue16257_20100111.diff uploaded by seanbright (license 71)

	* res/res_timing_timerfd.c: If we aren't running on a machine that
	  support CLOCK_MONOTONIC, don't load. Group developed and tested
	  by seanbright, Corydon76, Kobaz, and Amorsen.

2010-01-14 18:03 +0000 [r240179]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Fix broken call pickup The problem was the
	  OUTGOING flag was not getting set properly on the channel,
	  resulting in pickup failing as ast_read thought the call was
	  inbound. Refer to 170393 for a more verbose description as this
	  is the same exact change. (closes issue #16539) Reported by:
	  syspert Patches: bug16539.patch uploaded by jpeeler (license 325)
	  Tested by: syspert

2010-01-14 17:34 +0000 [r240129-240175]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Similarly, ensure that matchcid is duplicated
	  correctly when merging contexts.

	* main/pbx.c: Ensure that the callerid is NULL when the parent is
	  effectively NULL. This applies only to pattern-match hints, which
	  create exact-match hints on the fly.

2010-01-14 16:14 +0000 [r240078]  Matthew Nicholson <mnicholson@digium.com>

	* main/udptl.c: This change fixes a few bugs in the way the far max
	  IFP was calculated that were introduced in r231692. (closes issue
	  #16497) Reported by: globalnetinc Patches:
	  udptl-max-ifp-fix1.diff uploaded by mnicholson (license 96)
	  Tested by: globalnetinc

2010-01-14 14:38 +0000 [r240039]  Leif Madsen <lmadsen@digium.com>

	* doc/building_queues.txt (added): Add documentation about how to
	  build queues. Add a how-to set of documentation about building
	  queues with Asterisk. This documentation is based on Asterisk
	  1.6.2 but should work on most versions with minor modifications.
	  (closes issue #16237) Reported by: lmadsen Patches: Building
	  Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by:
	  pdhales, lmadsen, cmdrwalrus

2010-01-13 23:22 +0000 [r239920-239997]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Oops, another tag error

	* main/pbx.c: Oops, missed a closing tag

	* main/pbx.c, include/asterisk/pbx.h: Add the TESTTIME() dialplan
	  function, which permits testing GotoIfTime. Specifically, by
	  setting TESTTIME() to a particular date and time, you can test
	  whether a dialplan correctly branches as was intended. This was
	  developed after recent questions on the -users list on how to
	  test their holiday dialplan logic. (closes issue #16464) Reported
	  by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by
	  tilghman (license 14) Review:
	  https://reviewboard.asterisk.org/r/458/

	* main/ast_expr2f.c, main/ast_expr2.fl: Flex uses fwrite
	  incorrectly, which breaks the build. Providing a workaround.

2010-01-13 19:48 +0000 [r239839]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 239838 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010)
	  | 11 lines Fix regression for timed out parked call returning to
	  caller This issue seems to have been exposed by the fix in 160390
	  whereby using a masquerade prevented a crash. The new channel
	  used in the masquerade was not copying the macro information from
	  the old channel. (closes issue #15459) Reported by: djrodman
	  Patches: patch_15459.txt uploaded by mnick (license ) ........

2010-01-13 19:31 +0000 [r239834]  Leif Madsen <lmadsen@digium.com>

	* configs/extensions.conf.sample: Add more examples to
	  extensions.conf showing how to use various functionality and
	  provide commonly useful features. (closes issue #16090) Reported
	  by: pprindeville Patches: extensions.conf-bugid16090.patch#3
	  uploaded by pprindeville (license 347) Tested by: tzafrir,
	  pprindeville, lmadsen

2010-01-13 18:16 +0000 [r239797]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Code
	  previously added to ast_expr2f.c warranted a change in the source
	  file ast_expr2.fl. Also, made a Makefile change to ensure that
	  the expression parser C source files get regenerated correctly,
	  when we need that to happen.

2010-01-13 16:31 +0000 [r239712]  David Vossel <dvossel@digium.com>

	* Makefile, main/channel.c, apps/app_waitforring.c,
	  apps/app_waitforsilence.c: add silence gen to wait apps
	  asterisk.conf's 'transmit_silence' option existed before this
	  patch, but was limited to only generating silence while recording
	  and sending DTMF. Now enabling the transmit_silence option
	  generates silence during wait times as well. To achieve this,
	  ast_safe_sleep has been modified to generate silence anytime no
	  other generators are present and transmit_silence is enabled.
	  Wait apps not using ast_safe_sleep now generate silence when
	  transmit_silence is enabled as well. (closes issue #16524)
	  Reported by: kobaz (closes issue #16523) Reported by: kobaz
	  Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/456/

2010-01-13 10:45 +0000 [r239663-239665]  Olle Johansson <oej@edvina.net>

	* main/poll.c: MAX() moved to utils.h

	* channels/chan_sip.c: SIP Show channelstats fix - use float
	  division to show proper stats (closes issue #15819) Reported by:
	  klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt
	  uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This
	  patch is for trunk only and will be blocked in 1.6.2

2010-01-13 07:02 +0000 [r239624-239625]  TransNexus OSP Development <support@transnexus.com>

	* doc/tex/channelvariables.tex: Updated channel variable list of
	  osplookup application.

	* apps/app_osplookup.c: Updated XML doc for OSP.

2010-01-12 19:58 +0000 [r239571]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Blank callerid and NULL callerid should not compare
	  equal. The second is the default state for matching CID in the
	  dialplan (no matching) while the first matches one particular
	  CallerID. This is a regression. (fixes AST-314, SWP-611)

2010-01-12 18:55 +0000 [r239525]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/cdr.c: add Dialed Number Identifier (DNID) field to cdr
	  records. reviewboard link:
	  https://reviewboard.asterisk.org/r/455/ Reported by: alecdavis
	  Tested by: alecdavis Patch cdr_dnid.diff2.txt uploaded by
	  alecdavis (license 585)

2010-01-12 18:22 +0000 [r239520]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Note that direct T.38 is not supported.
	  (closes issue #16411) Reported by: stanusr Patches:
	  __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen
	  (license 10)

2010-01-12 17:09 +0000 [r239473]  Sean Bright <sean@malleable.com>

	* res/res_config_ldap.c: Fix crash in res_config_ldap. We need to
	  allocate enough room for 2 pointers, not 2 characters. (closes
	  issue #16397) Reported by: bklang Patches: res_config_ldap.patch
	  uploaded by applsplatz (license 949) Tested by: applsplatz

2010-01-12 16:14 +0000 [r239427]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes text support in sdp answer The code
	  that handled setting 'm=text' in the sdp was not executing in the
	  correct order. The check to see if text was needed came after the
	  check to add 'm=text' to the sdp, this resulted in 'm=text'
	  always being set to 0 because it looked like text was never
	  required. (closes issue #16457) Reported by: peterj Patches:
	  textportinsdp.diff uploaded by peterj (license 951)
	  issue16457.diff uploaded by dvossel (license 671) Tested by:
	  peterj

2010-01-12 07:48 +0000 [r239389]  Olle Johansson <oej@edvina.net>

	* include/asterisk/astmm.h: Adding Tilghman's documentation from
	  asterisk-dev to the actual file.

2010-01-12 03:21 +0000 [r239152-239308]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/safe_asterisk: Merged revisions 239307 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r239307 | tilghman | 2010-01-11 21:18:36 -0600 (Mon, 11 Jan 2010)
	  | 8 lines Portability and other fixes for the safe_asterisk
	  script (closes issue #16416) Reported by: bklang Patches:
	  safe_asterisk-compat-1.patch uploaded by bklang (license 919)
	  20100106__issue16416__trunk.diff.txt uploaded by tilghman
	  (license 14) Tested by: bklang ........

	* contrib/init.d/rc.mandriva.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.slackware.asterisk,
	  contrib/init.d/rc.archlinux.asterisk,
	  contrib/init.d/rc.suse.asterisk: Add LSB headers to init scripts.
	  (closes issue #14864) Reported by: lathama Patches:
	  lsb-init-info-debian.diff uploaded by pkempgen (license 169)

	* res/res_pktccops.c: Socket level option is SOL_SOCKET, not
	  SO_SOCKET. (issue #16580)

	* Makefile, contrib/init.d/rc.mandriva.asterisk,
	  contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.suse.asterisk: Permit more options in the
	  Makefile as to startup options (closes issue #16454) Reported by:
	  syspert Patches: 20091228__issue16454__3.diff.txt uploaded by
	  tilghman (license 14) Tested by: syspert

	* Makefile: Including bundle1.o breaks Tiger and Leopard (issue
	  #16449)

	* addons/cdr_mysql.c, configs/cdr_mysql.conf.sample: Permit dates
	  and times to be stored in timezones other than the default
	  (typically, UTC) (closes issue #16401) Reported by: lordmortis

2010-01-11 16:41 +0000 [r239111-239114]  Sean Bright <sean@malleable.com>

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  res/res_calendar_caldav.c, res/res_clialiases.c: Pass NULL for
	  the ao2_callback function pointer instead of duplicating cb_true.

	* main/astobj2.c: Fix ao2_callback when both OBJ_MULTIPLE and
	  OBJ_NODATA are passed. There is an issue which only affects trunk
	  and the new ao2_callback OBJ_MULTIPLE implementation. When both
	  OBJ_MULTIPLE and OBJ_NODATA are passed, only the first object is
	  visited, regardless of what is returned by the specified
	  callback. This causes a problem when we are clearing a container,
	  i.e.: ao2_callback(container, OBJ_UNLINK | OBJ_NODATA |
	  OBJ_MULTIPLE, NULL, NULL); Only unlinks the first object. This
	  patch resolves this. (closes issue #16564) Reported by: pj
	  Patches: issue16564_20100111.diff uploaded by seanbright (license
	  71) Tested by: pj, seanbright Review:
	  https://reviewboard.asterisk.org/r/457/

	* main/test.c: Fix spelling of 'category.'

2010-01-10 19:37 +0000 [r239074]  Tilghman Lesher <tlesher@digium.com>

	* addons/chan_ooh323.c, main/frame.c, channels/chan_iax2.c:
	  According to POSIX, the capital L modifier applies only to
	  floating point types. Fixes a crash on Solaris. (closes issue
	  #16572) Reported by: crjw Patches: frame_changes.patch uploaded
	  by crjw (license 963) Plus several others found and fixed by me

2010-01-10 17:53 +0000 [r239037]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooq931.h, addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooq931.c: add docallbacks flag in q931decode
	  function because when we decode received q931 packet we must do
	  callbacks and when we print sended q931 packet we must not.

2010-01-10 06:56 +0000 [r239000]  Tilghman Lesher <tlesher@digium.com>

	* Makefile, main/asterisk.c: It's been long enough -- make the
	  behavior introduced in 1.6 the default.

2010-01-09 01:08 +0000 [r238916]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 238915 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010)
	  | 6 lines -1 is interpreted as an error, intead of the maximum
	  mask. (closes issue #16241) Reported by: vnovy Patches:
	  manager.c.patch uploaded by vnovy (license 922) ........

2010-01-08 23:30 +0000 [r238835]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 238834 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r238834 | jpeeler | 2010-01-08 17:28:37 -0600 (Fri, 08 Jan 2010)
	  | 4 lines Stop a crash when no peer is passed to masq_park_call.
	  (distantly related to issue #16406) ........

2010-01-08 22:54 +0000 [r238754-238795]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Add the class actually used in the
	  MusicOnHold start event. (closes issue #16499) Reported by:
	  syspert Patches: mohclass.patch uploaded by syspert (license 938)

	* res/res_agi.c: Initialize variables that we attempt to free
	  later. (closes issue #16302) Reported by: yahsyn Patches:
	  20091124__issue16302.diff.txt uploaded by tilghman (license 14)
	  Tested by: yahsyn

2010-01-08 21:04 +0000 [r238716]  Matthew Nicholson <mnicholson@digium.com>

	* tests/test_ast_format_str_reduce.c (added): Added a test for
	  ast_format_reduce_str(). (related to issue #16560)

2010-01-08 19:39 +0000 [r238635]  David Vossel <dvossel@digium.com>

	* include/asterisk/audiohook.h, main/audiohook.c: fixes
	  AUDIOHOOK_INHERIT regression During the process of removing an
	  audiohook from one channel and attaching it to another the
	  audiohook's status is updated to DONE and then back to whatever
	  it was previously. Typically updating the status after setting it
	  to DONE is not a good idea because DONE can trigger unrecoverable
	  audiohook destruction events... because of this a conditional
	  check was added to audiohook_update_status to explicitly prevent
	  the audiohook from ever changing after being set to DONE. It was
	  this check that prevented audiohook inherit from work properly
	  though. Now ast_audiohook_move_by_source is treated as a special
	  exception, as the audiohook must be returned to its previous
	  status after attaching it to the new channel. This is only a safe
	  operation because the audiohook's lock is held the entire time,
	  otherwise this could cause trouble. (closes issue #16522)
	  Reported by: corruptor

2010-01-08 19:32 +0000 [r238630]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/file.c: Merged revisions 238629 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan
	  2010) | 5 lines Properly calculate the remaining space in the
	  output string when reducing format strings. (closes issue #16560)
	  Reported by: goldwein ........

2010-01-08 17:18 +0000 [r238583]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Stop trying to find a parking space after
	  traversing the parkinglot one time. (closes issue #16428)
	  Reported by: Yasuhiro Konishi

2010-01-07 21:24 +0000 [r238527]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Fix using the wrong pointer type in
	  do_idle_thread().

2010-01-07 20:42 +0000 [r238361-238492]  David Vossel <dvossel@digium.com>

	* main/channel.c: fixes ast_transfer stall until hangup if called
	  with a channel that doesn't support transfers ast_transfer sets
	  res to 0 if there is no technology transfer function, but then
	  tests for it to be negative before deciding to do an early exit.
	  As a result, it will will wait for an AST_CONTROL_TRANSFER
	  message that will never come. (closes issue #16424) Reported by:
	  davidw Patches: Issue_16424_trunk_234134.patch uploaded by davidw
	  (license 780)

	* /, channels/chan_iax2.c: Merged revisions 238411 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07
	  Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in
	  chan_iax A signed short was used to represent a callnumber. This
	  is makes it possible to attempt to access the iaxs array with a
	  negative index. (closes issue #16565) Reported by: jensvb
	  ........

	* channels/chan_sip.c: Change in sip show channels display format
	  allowing more digits for CID (closes issue #16459) Reported by:
	  Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
	  (license 953)

	* apps/app_queue.c: cli 'queue show' formatting fix. queue name was
	  truncated over 12 characters (closes issue #16078) Reported by:
	  RoadKill Patches: quequename_limit.patch uploaded by ppyy
	  (license 906) Tested by: dvossel

2010-01-07 09:14 +0000 [r238313]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* configs/sip.conf.sample: Document the usefulness of explicit
	  udp:// in the register string

2010-01-06 21:45 +0000 [r238231]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_cdr.c: Merged revisions 238230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
	  | 4 lines Revise documentation on disposition values to the
	  actual values used. (closes issue #16289) Reported by: wdoekes
	  ........

2010-01-06 20:37 +0000 [r238134-238181]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c: Fix misreverting from 177158. (closes issue
	  #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by
	  dimas (license 88) Tested by: shanermn

	* main/features.c: Fix channel name comparison for bridge
	  application. The channel name comparison was not comparing the
	  whole string and therefore if one channel name was a substring of
	  the other, the bridge would fail. (closes issue #16528) Reported
	  by: telecos82 Patches: res_features_r236843.diff uploaded by
	  telecos82 (license 687)

2010-01-06 16:36 +0000 [r238091]  David Vossel <dvossel@digium.com>

	* include/asterisk/test.h: fixes test.c compile issue when
	  TEST_FRAMEWORK is not enabled The ast_test_status_update()
	  function is defined in test.h. When TEST_FRAMEWORK is not enabled
	  a macro is defined as a no-op place holder for this function. The
	  macro did not contain the correct number of arguments. This
	  caused a compile error. Much thanks to wdoekes for reporting the
	  issue and supplying the patch!

2010-01-06 15:35 +0000 [r238014]  Sean Bright <sean@malleable.com>

	* addons/format_mp3.c: Fix reading samples from format_mp3 after
	  ast_seekstream/ast_tellstream. There is a bug when using
	  ast_seekstream/ast_tellstream with format_mp3 in that the file
	  read position is not reset before attempting to read samples. So
	  when we seek to determine the maximum size of the file (as in
	  res_agi's STREAM FILE) we weren't then resetting the file pointer
	  so that we could properly read samples. This patch addresses that
	  (in a similar manner to format_wav.c). (closes issue #15224)
	  Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff
	  uploaded by seanbright (license 71) Tested by: rbd, seanbright
	  Review: https://reviewboard.asterisk.org/r/453

2010-01-06 15:19 +0000 [r238010]  Russell Bryant <russell@digium.com>

	* /, apps/app_mp3.c: Merged revisions 238009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
	  | 7 lines Resolve a crash due to an ast_frame not being fully
	  initialized. (closes issue #16531) Reported by: john8675309
	  (closes SWP-615) ........

2010-01-06 06:53 +0000 [r237968]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Whoa, duplicate setting (dead code).

2010-01-05 23:08 +0000 [r237920]  David Vossel <dvossel@digium.com>

	* apps/app_queue.c: fixes holdtime playback issue in app_queue When
	  reporting hold time, the number of seconds should be mod 60.
	  Otherwise audio playback could be something like "2 minutes 123
	  seconds" rather than "2 minutes 3 seconds". Also, the "minute"
	  sound file is missing, so for the moment until that file can be
	  created the "minutes" file is used instead. (closes issue #16168)
	  Reported by: nickilo Patches: patch-unified-trunk-rev-222176
	  uploaded by nickilo (license ) Tested by: nickilo, wonderg

2010-01-05 20:56 +0000 [r237882]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Mismerged a bit.

2010-01-05 19:29 +0000 [r237839]  David Vossel <dvossel@digium.com>

	* main/pbx.c: fixes subscriptions being lost after 'module reload'
	  During a module reload if multiple extension configs are present,
	  such as both extensions.conf and extensions.ael, watchers for one
	  config's hints will be lost during the merging of the other
	  config. This happens because hint watchers are only preserved for
	  the current config being merged. The old context list is
	  destroyed after the merging takes place, meaning any watchers
	  that were not perserved will be removed. Now all hints are
	  preserved during merging regardless of what config file is being
	  merged. These hints are only restored if they are present within
	  the new context list. (closes issue #16093) Reported by: jlaroff

2010-01-05 18:57 +0000 [r237804]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h, channels/sig_pri.c: Removed unused
	  parameters from analog_available() and sig_pri_available().

2010-01-05 18:46 +0000 [r237802-237803]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, CHANGES: Add a missing part of the connected
	  line work into trunk. Part of the work done for connected line
	  was to add an optional argument to the 'f' option to allow for
	  the connected party information of the outgoing channel to be set
	  to the argument provided. This was overlooked during the merge of
	  the work to trunk and is being added back now. The CHANGES file
	  has also been updated to note this change.

	* CHANGES: Spell "aficionado" like someone who isn't stupid.

2010-01-05 17:26 +0000 [r237699-237749]  Russell Bryant <russell@digium.com>

	* main/utils.c: Fix build of utility apps that include utils.c.

	* /, main/utils.c: Merged revisions 237697 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
	  | 7 lines Change a NOTICE log message to DEBUG where it belongs.
	  (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
	  ........

2010-01-05 16:08 +0000 [r237656]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop
	  <channel> work again. (closes issue #16534) Reported by:
	  jlaguilar Fix as suggested by jlaguilar in the bugreport

2010-01-04 21:48 +0000 [r237406-237574]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c: Merged revisions 237573 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
	  | 6 lines Bounds checking for input string (closes issue #16407)
	  Reported by: qwell Patches: 20100104__issue16407.diff.txt
	  uploaded by tilghman (license 14) ........

	* main/pbx.c, /: Merged revisions 237493 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
	  | 8 lines Regression in issue #15421 - Pattern matching (closes
	  issue #16482) Reported by: wdoekes Patches:
	  astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
	  20091223__issue16482.diff.txt uploaded by tilghman (license 14)
	  Tested by: wdoekes, tilghman ........

	* main/config.c: Oops, didn't compile (thanks, kpfleming)

	* main/config.c: Further reduce the encoded blank values back to
	  blank in the realtime API. (closes issue #16533) Reported by:
	  sergee Patches: 200100104__issue16533.diff.txt uploaded by
	  tilghman (license 14) Tested by: sergee

	* main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
	  revisions 237405 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
	  | 16 lines Add a flag to disable the Background behavior, for AGI
	  users. This is in a section of code that relates to two other
	  issues, namely issue #14011 and issue #14940), one of which was
	  the behavior of Background when called with a context argument
	  that matched the current context. This fix broke FreePBX,
	  however, in a post-Dial situation. Needless to say, this is an
	  extremely difficult collision of several different issues. While
	  the use of an exception flag is ugly, fixing all of the issues
	  linked is rather difficult (although if someone would like to
	  propose a better solution, we're happy to entertain that
	  suggestion). (closes issue #16434) Reported by: rickead2000
	  Patches: 20091217__issue16434.diff.txt uploaded by tilghman
	  (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
	  tilghman (license 14) Tested by: rickead2000 ........

2010-01-04 16:39 +0000 [r237327]  David Vossel <dvossel@digium.com>

	* apps/app_queue.c: app_queue segfaults if realtime field uniqueid
	  is NULL (closes issue #16385) Reported by: haakon Patches:
	  app_queue.c.patch uploaded by haakon (license 880)
	  app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by:
	  haakon

2010-01-04 16:24 +0000 [r237323]  Jeff Peeler <jpeeler@digium.com>

	* res/res_agi.c: Fix timeout for AGI command speech recognize.
	  (closes issue #16297) Reported by: semond

2010-01-04 16:20 +0000 [r237319]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 237318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04
	  Jan 2010) | 3 lines It's also possible for the Local channel to
	  directly execute an Application. Reviewboard:
	  https://reviewboard.asterisk.org/r/452/ ........

2010-01-04 07:55 +0000 [r237284]  Olle Johansson <oej@edvina.net>

	* res/res_pktccops.c, channels/chan_mgcp.c: - Disable res_pktccops
	  by default - Add dependency in chan_mgcp that was missing - Add a
	  small amount of doc to the source code

2010-01-04 03:38 +0000 [r237250]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: 1. Added reporting operator names in
	  AuthReq. 2. Added retrieving operator names from AuthRsp and
	  exporting them.

2010-01-02 16:35 +0000 [r237213]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: global_contact_ha was renamed in trunk

2010-01-02 09:54 +0000 [r237136]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 237135 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
	  lines Release memory of the contact acl before unloading module
	  ........

2009-12-30 23:51 +0000 [r237098]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ooCalls.c: small q931 processing and
	  signalling corrections don't decode UUIE from Q931StatusMessage
	  clean call without callIdentifier data don't start tcs/msd
	  exchange procedure after call proceeding received (closes issue
	  #16365) Reported by: benngard2 Tested by: may213, benngard2

2009-12-30 22:30 +0000 [r237050]  Jason Parker <jparker@digium.com>

	* main/say.c, doc/lang/vietnamese.ods (added),
	  apps/app_voicemail.c: Add app_voicemail and say.c support for
	  Vietnamese. Also add an XXX comment that I'm baffled nobody has
	  ever complained about. We say "first message", and then we go
	  into language-specific stuff where we proceed to say..."first
	  message". (closes issue #15053) Reported by: dinhtrung Patches:
	  vietnamese.ods uploaded by dinhtrung (license 776)
	  app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes
	  issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded
	  by dinhtrung (license 776)

2009-12-30 21:59 +0000 [r236982]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 236981 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30
	  Dec 2009) | 9 lines Don't queue frames to channels that have no
	  means to process them. (closes issue #15609) Reported by: aragon
	  Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt
	  uploaded by tilghman (license 14) Tested by: aragon Review:
	  https://reviewboard.asterisk.org/r/452/ ........

2009-12-30 21:09 +0000 [r236893-236902]  Jeff Peeler <jpeeler@digium.com>

	* utils/ael_main.c: One more LOW_MEMORY compile fix.

	* channels/chan_sip.c, main/cli.c: Fix compiling with LOW_MEMORY.
	  Modified handle_verbose to be LOW_MEMORY aware, removed old RTP
	  related code in chan_sip. (closes issue #16381) Reported by:
	  michael_iedema Patches: ast_complete_source_filename.patch
	  uploaded by michael iedema (license 942) modified by me

2009-12-30 17:53 +0000 [r236802-236847]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_adaptive_odbc.c, cel/cel_adaptive_odbc.c: When the field
	  is blank, don't warn about the field being unable to be coerced,
	  just skip the column. (closes
	  http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
	  Reported by Nic Colledge on the -dev list, fixed by me.

	* channels/chan_sip.c: Shut down the SIP session timers more
	  gracefully, in order to prevent a possible crash. (closes issue
	  #16452) Reported by: corruptor Patches:
	  20091221__issue16452.diff.txt uploaded by tilghman (license 14)
	  Tested by: corruptor

2009-12-29 10:59 +0000 [r236756]  TransNexus OSP Development <support@transnexus.com>

	* configs/osp.conf.sample, apps/app_osplookup.c, configure.ac: 1.
	  Updated for OSP Toolkit 3.6.0. 2. Added service type ported
	  number query. 3. Formated code.

2009-12-28 22:09 +0000 [r236713]  Jason Parker <jparker@digium.com>

	* main/ast_expr2.y, main/ast_expr2.c: Allow "REMAINDER" to function
	  properly in expressions. (closes issue #16427) Reported by:
	  wdoekes Patches: ast16-reminder-remainder.patch uploaded by
	  wdoekes (license 717) Tested by: wdoekes

2009-12-28 17:37 +0000 [r236667]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Use recommended option, not deprecated
	  option. (closes issue #16515) Reported by: ManChicken

2009-12-28 15:22 +0000 [r236510-236613]  Sean Bright <sean@malleable.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/threadstorage.h: Merged revisions 236585 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
	  2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
	  requires extra braces. There was conditional code (based on build
	  platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
	  was removed since it is fixed in newer versions of
	  Solaris/OpenSolaris, but I am still running into it on Solaris 10
	  x86 so add a configure-time check for it. ........

	* /, apps/app_meetme.c: Merged revisions 236509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
	  2009) | 12 lines Avoid a crash with large numbers of MeetMe
	  conferences. Similar to changes made to Queue(), when we have
	  large numbers of conferences in meetme.conf (1000s) and we use
	  alloca()/strdupa(), we can blow out the stack and crash, so
	  instead just use a single fixed buffer. (closes issue #16509)
	  Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
	  by seanbright (license 71) Tested by: seanbright ........

2009-12-27 18:20 +0000 [r236434]  Tilghman Lesher <tlesher@digium.com>

	* contrib/init.d/rc.debian.asterisk, /: Merged revisions 236433 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009)
	  | 2 lines Turn on colors in the daemon, since there's many
	  requests for it on Ubuntu. ........

2009-12-26 15:27 +0000 [r236358]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 236357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
	  2009) | 1 line update to latest releases with zero uid/gid
	  ........

2009-12-23 19:17 +0000 [r236304-236312]  David Vossel <dvossel@digium.com>

	* CHANGES: Update CHANGES to reflect new QUEUE_MEMBER option,
	  "ready"

	* apps/app_queue.c: QUEUE_MEMBER(..., ready) counts only ready
	  agents, not free agents wrapping up The QUEUE_MEMBER dialplan
	  function can return total members, logged-in members and "free"
	  members count. A member is counted as "free" immediately after
	  his call ends, even though its wrap-up time, if specified in
	  queues.conf, has not yet expired, and the queue will not actually
	  route a call to it. This Patch introduces a new "ready" option
	  that only counts free agents no longer in the wrap up time
	  period. (closes issue #16240) Reported by: kkm Patches:
	  appqueue-memberfun-readyoption-trunk.diff uploaded by kkm
	  (license 888) Tested by: kkm, dvossel

	* CHANGES, apps/app_queue.c: update CHANGES to reflect new 'R'
	  app_queue option plus a minor optimization to the feature patch
	  (issue #16384)

	* apps/app_queue.c: new parameter 'R' to the Queue application The
	  'R' argument stops moh and indicates ringing once the agent is
	  ringing. This allows the person in the queue to know their call
	  is potentially about to be answered. (closes issue #16384)
	  Reported by: haakon Patches: new_app_queue.c.patch uploaded by
	  haakon (license 880) Tested by: haakon, loloski, dvossel

2009-12-23 18:25 +0000 [r236183-236300]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: AGI may be invoked from outside the dialplan
	  (closes issue #16510) Reported by: atis Patches:
	  20091223__issue16510.diff.txt uploaded by tilghman (license 14)
	  Tested by: atis

	* /, res/res_agi.c: Merged revisions 236184 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
	  | 4 lines If EXEC only gets a single argument, don't crash when
	  the second is used. (closes issue #16504) Reported by: bklang
	  ........

	* include/asterisk/test.h: Allow test_heap.c to compile when
	  AST_DEVMODE is true, but TEST_FRAMEWORK is false

	* apps/app_voicemail.c: Actually use tmp for something (brings
	  trunk back into sync with 1.6 branches).

2009-12-22 21:53 +0000 [r236027-236144]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes iax "can't compress subclass
	  4294967295" error (closes issue #16456) Reported by: dvossel
	  Tested by: dvossel

	* /, channels/chan_sip.c: Merged revisions 236062 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
	  | 11 lines fixes issue with p->method incorrectly set to ACK It
	  is possible for a second ACK to come in for a retransmitted
	  message. If an ack does not match an unacked message in our
	  queue, restore the previous p->method as this ACK is completely
	  ignored. (closes issue #16295) Reported by: omolenkamp Patches:
	  issue16295_v2.diff uploaded by dvossel (license 671) ........

	* CHANGES: update CHANGES to reflect the addition of the test
	  framework

	* include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
	  tests/test_heap.c, main/test.c (added),
	  include/asterisk/_private.h, main/asterisk.c: Unit Test Framework
	  API The Unit Test Framework is a new API that manages
	  registration and execution of unit tests in Asterisk with the
	  purpose of verifying the operation of C functions. The Framework
	  consists of a single test manager accompanied by a list of
	  registered test functions defined within the code. A test is
	  defined, registered, and unregistered from the framework using a
	  set of macros which allow the test code to only be compiled
	  within asterisk when the TEST_FRAMEWORK flag is enabled in
	  menuselect. This allows the test code to exist in the same file
	  as the C functions it intends to verify. Registered tests may be
	  viewed and executed via a set of new CLI commands. CLI commands
	  are also present for generating and exporting test results into
	  xml and txt formats. For more information and use cases please
	  refer to the documentation provided at the beginning of the
	  test.h file. Review: https://reviewboard.asterisk.org/r/447/

2009-12-21 19:54 +0000 [r235941]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_monitor.c: Merged revisions 235940 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
	  | 13 lines Change Monitor to not assume file to write to does not
	  contain pathing. 227944 changed the fname_base argument to always
	  append the configured monitor path. This change was necessary to
	  properly compare files for uniqueness. If a full path is given
	  though, nothing needs to be appended and that is handled
	  correctly now. (closes issue #16377) (closes issue #16376)
	  Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
	  uploaded by dant (license 670) ........

2009-12-21 18:51 +0000 [r235904]  Kevin P. Fleming <kpfleming@digium.com>

	* contrib/upstart/asterisk.upstart-0.3.9, include/asterisk/cel.h,
	  main/say.c, include/asterisk/channel.h,
	  include/asterisk/manager.h, channels/sig_pri.c,
	  include/asterisk/logger.h, include/asterisk/http.h,
	  include/asterisk/callerid.h, include/asterisk/syslog.h,
	  channels/chan_dahdi.c, include/asterisk/app.h,
	  include/asterisk/doxyref.h, include/asterisk/event.h,
	  channels/sig_analog.c, channels/chan_misdn.c,
	  contrib/upstart/asterisk.user.conf,
	  include/asterisk/rtp_engine.h,
	  include/asterisk/security_events.h,
	  include/asterisk/stringfields.h: Change all refererences to 1.6.3
	  to be 1.8, since that will be the next feature release

2009-12-21 17:00 +0000 [r235822]  Tilghman Lesher <tlesher@digium.com>

	* /, main/features.c: Merged revisions 235821 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
	  | 8 lines Send parking lot announcement to the channel which
	  parked the call, not the park-ee. (closes issue #16234) Reported
	  by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
	  by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
	  uploaded by tilghman (license 14) Tested by: yeshuawatso ........

2009-12-20 08:22 +0000 [r235740-235774]  Alec L Davis <sivad.a@paradise.net.nz>

	* main/dsp.c: restarts busydetector (if enabled) when DTMF is
	  received after call is bridged. (closes issue 0016389) Reported
	  by: alecdavis Tested by: alecdavis Patch
	  dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)

	* apps/app_dial.c, CHANGES: app_dial optional parameter to option
	  'r' to allow play indication from indications.conf (closes issue
	  #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch
	  app_dial.play_ring_indications.diff7.txt uploaded by alecdavis
	  (license 585)

2009-12-18 22:51 +0000 [r235660]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /, include/asterisk/cdr.h: Merged revisions
	  235635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
	  | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
	  simple in that it reorders the disposition defines so that the
	  fix for issue 12946 works properly (the default CDR disposition
	  was changed to AST_CDR_NOANSWER). Also, the
	  AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
	  CDR records are written. The side effects of CDR changes are
	  scary, so I'm documenting the test cases performed to attempt to
	  catch any regressions. The following tests were all performed
	  using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
	  B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
	  blind transfers to C Hangup C (Both SIP and features) A calls B A
	  attended transfers to C Hangup C A calls B A attended transfers
	  to C (SIP) C blind transfers to A (features) Hangup A All of the
	  test scenario CDRs matched. The following tests were performed
	  just with the patch to ensure proper operation (with
	  unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
	  =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
	  (closes issue #16180) Reported by: aatef Patches: bug16180.patch
	  uploaded by jpeeler (license 325) ........

2009-12-18 22:40 +0000 [r235573-235656]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, configure.ac: Merged revisions 235652 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
	  Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
	  ........

	* /, configure, configure.ac: Merged revisions 235572 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
	  Dec 2009) | 2 lines Point to the typical missing package, not the
	  cryptic "termcap support". ........

2009-12-17 23:21 +0000 [r235521]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Remove some old code for going to the 'fax'
	  extension when a T.38 switchover occurs. This would have already
	  happened when we detected the CNG tone so this was basically a
	  noop.

2009-12-17 17:19 +0000 [r235422]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 235421 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009)
	  | 8 lines Use context from which Macro is executed, not macro
	  context, if applicable. Also, ensure that the extension COULD
	  match, not just that it won't match more. (closes issue #16113)
	  Reported by: OrNix Patches: 20091216__issue16113.diff.txt
	  uploaded by tilghman (license 14) Tested by: OrNix ........

2009-12-17 00:52 +0000 [r235342-235382]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Fix call forwarding
	  for analog phones. (closes issue #16440) Reported by: mmichelson

	* configs/jabber.conf.sample, include/asterisk/jabber.h, CHANGES,
	  res/res_jabber.c: Add auth_policy option to jabber.conf for auto
	  user registration. The option is global and currently the
	  acceptable values as noted in the sample config are accept or
	  deny. (closes issue #15228) Reported by: lp0

2009-12-16 05:24 +0000 [r235298]  Jared Smith <jaredsmith@jaredsmith.net>

	* /, configs/sip.conf.sample: Merged revisions 235181 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15
	  Dec 2009) | 4 lines Add a line showing that we can use CIDR
	  notation. patch by jsmith, after discussion with jtodd ........

2009-12-16 00:31 +0000 [r235265]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, CHANGES: Enhance AMI redirect to allow channels
	  to be redirected to different places. New parameters
	  ExtraContext, ExtraExtension, and ExtraPriority have been added
	  to redirect the second channel to a different location.
	  Previously, it was only possible to redirect both channels to the
	  same place. (closes issue #15853) Reported by: haakon Patches:
	  trunk-manager.c.patch uploaded by haakon (license 880) Tested by:
	  jpeeler

2009-12-15 23:51 +0000 [r235229]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/strings.h: Is it Friday yet?

2009-12-15 23:41 +0000 [r235226]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c: Change match criteria existence in
	  ast_channel_cmp_cb to use ast_strlen_zero. (closes issue #16161)
	  Reported by: may213 Patches: core-show-channel.patch uploaded by
	  may213 (license 454)

2009-12-15 18:43 +0000 [r235132]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: reverse minor sip registration regression A
	  registration regression caused by a code tweak in (issue #14331)
	  and a bug fix in (issue #15539) caused some sip registration
	  config entries to be constructed incorrectly. Origially issue
	  #14331 contained the code tweak as well as a bug fix, but since
	  the issue was reported as a tweak the bug fix portion was moved
	  into issue #15539. Both the tweak and the bug fix contained minor
	  incorrect logic that resulted in some SIP registrations to fail.
	  (issue #14331) (issue #15539)

2009-12-15 15:33 +0000 [r235053]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 235052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009)
	  | 4 lines Mandatory argument checking (closes issue #16446)
	  Reported by: nicchap ........

2009-12-15 14:35 +0000 [r235010]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: spandsp does in fact support V.17 modulation at
	  14.4 kilobits per second, so we should generate T38MaxBitRate of
	  14400 (even though that doesn't really affect the FAX
	  transmission much at all)

2009-12-15 07:18 +0000 [r234855-234976]  Alec L Davis <sivad.a@paradise.net.nz>

	* apps/app_directory.c: Support option 'n', as applications like
	  Playback, Background etc. Suggested on asterisk-dev as trivial
	  application change. Reported by: alecdavis Tested by: alecdavis

	* main/dsp.c: Whitespace.

	* main/dsp.c: restarts busydetector (if enabled) when DTMF is
	  received. (closes issue #16389) Reported by: alecdavis Tested by:
	  alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis
	  (license 585)

	* apps/app_directory.c: fixes escape to extensions 'o' and 'a', for
	  digits '0' and '*' (closes issue #16437) Reported by: alecdavis
	  Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by
	  alecdavis (license 585)

	* apps/app_directory.c: ast_stream_and_wait(chan,dir-usingkeypad)
	  didn't capture the dialled DTMF. (closes issue #16409) Reported
	  by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt
	  uploaded by alecdavis (license 585)

2009-12-14 23:16 +0000 [r234820]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Allow greetings-only mailboxes for Voicemail. (closes issue
	  #15132) Reported by: floletarmo Patches: voicemail_changes.patch
	  uploaded by floletarmo (license 784) (with some additional
	  changes by me)

2009-12-14 21:32 +0000 [r234776]  Jason Parker <jparker@digium.com>

	* apps/app_readexten.c: Allow tonelist as argument to ReadExten.
	  ReadExten already supported playing a tonezone from
	  indications.conf. It now has the ability to use a tonelist like
	  440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert
	  Patches: app_readexten.c.patch uploaded by jcovert (license 551)
	  Tested by: qwell Patch modified by me, to maintain backwards
	  compatibility.

2009-12-14 21:13 +0000 [r234700]  Tilghman Lesher <tlesher@digium.com>

	* /, build_tools/make_version_c, build_tools/make_version_h: Merged
	  revisions 234699 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009)
	  | 5 lines Deal with the situation where .flavor exists but
	  .version does not. Also make the script slightly more portable,
	  in keeping with autoconf syntax. (closes issue #14737) Reported
	  by: davidw ........

2009-12-14 17:19 +0000 [r234631]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/imapstorage.tex, /: Update IMAP build documentation.
	  Update the IMAP build documentation to show how to build on
	  64-bit platforms. (issue #16433) Reported by: shrift Tested by:
	  lmadsen

2009-12-14 16:08 +0000 [r234572]  Sean Bright <sean@malleable.com>

	* main/timing.c: The default rate for 'timing test' is actually
	  50/sec, not 100/sec as advertised.

2009-12-14 10:46 +0000 [r234526]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 234492 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8
	  lines Stop sending 183's after call hangup. There where still
	  cases where the 183 keep-alive mechanism would not stop sending
	  183's even though the Asterisk server had sent a final reply to
	  the invite. EDVX-28 ........

2009-12-13 09:41 +0000 [r234458]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Trim leading/trailing spaces from the filename, to
	  deal with common user error.

2009-12-11 23:17 +0000 [r234380]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_meetme.c: Merged revisions 234379 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009)
	  | 11 lines Fix talking detection status after conference user is
	  muted. This patch ensures that when a conference user is muted
	  that the accompanying AMI Meetme talking off event is sent. Also,
	  the meetme list output is updated to show the muted user as
	  unmonitored. (closes issue #16247) Reported by: dimas Patches:
	  v3-16247.patch uploaded by dimas (license 88) ........

2009-12-10 21:01 +0000 [r234256]  Jason Parker <jparker@digium.com>

	* Makefile, /: Merged revisions 234255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) |
	  9 lines Fix unselecting of menuselect options via GLOBAL_MAKEOPTS
	  and USER_MAKEOPTS. (closes issue #16296) Reported by: abelbeck
	  Patches: issue16296-20091210.diff uploaded by qwell (license 4)
	  (abelbeck described a fix, which I expanded upon) Tested by:
	  abelbeck, qwell, lmadsen ........

2009-12-10 18:56 +0000 [r234210]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Missed a case that emits a WARNING where
	  none is warranted.

2009-12-10 17:31 +0000 [r234173]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_meetme.c, apps/app_page.c, main/app.c, CHANGES: Add
	  audio announcement option to app_page As described in the CHANGES
	  file: * MeetMe has a new option 'G' to play an announcement
	  before joining a conference. * Page has a new option 'A(x)' which
	  will playback an announcement simultaneously to all paged phones
	  (and optionally excluding the caller's one using the new option
	  'n') before the call is bridged. To add the new option to meetme,
	  the conference flag options had to be extended to 64 bits.
	  (closes issue #14365) Reported by: dferrer Patches:
	  page_announce.patch uploaded by dferrer (license 525) modified by
	  me Review: https://reviewboard.asterisk.org/r/188/

2009-12-10 16:24 +0000 [r234129]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 234095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009)
	  | 9 lines When we receive no response at all to our INVITE, allow
	  the channel to be destroyed. (closes issue #15627) Reported by:
	  falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded
	  by tilghman (license 14) 20091209__issue15627__1.4.diff.txt
	  uploaded by tilghman (license 14) Tested by: falves11 Review:
	  https://reviewboard.asterisk.org/r/446/ (closes issue #15716)
	  Reported by: dant (closes issue #16270) Reported by: corruptor
	  (closes issue #15356) Reported by: falves11 (issue #16382)
	  Reported by: lftsy ........

2009-12-09 23:35 +0000 [r233967-234055]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt, CHANGES: Move an entry from CHANGES to UPGRADE.txt.

	* UPGRADE.txt, CHANGES: Move an entry from CHANGES that should be
	  in UPGRADE.txt.

	* CHANGES: Provide a real description of LOCAL_PEEK().

	* CHANGES: Remove a feature from CHANGES that was listed twice for
	  1.6.2.

	* CHANGES: Fix up the faxdetect entry in CHANGES. This feature was
	  listed as a 1.6.2 feature, even though it's in all 1.6.X
	  versions. The description of the feature was also no longer
	  accurate.

	* CHANGES: Remove an entry from CHANGES that is already in
	  UPGRADE.txt (where it should be).

2009-12-08 18:40 +0000 [r233718-233732]  Tilghman Lesher <tlesher@digium.com>

	* addons/res_config_mysql.c: Typo pointed out on #asterisk-dev (by
	  atis_work)

	* res/res_musiconhold.c: Find another ref leak and change how we
	  manage module references. (closes issue #16388, closes issue
	  #16279, closes issue #16390) Reported by: parisioa Patches:
	  20091208__issue16388.diff.txt uploaded by tilghman (license 14)
	  Tested by: parisioa, tilghman Review:
	  https://reviewboard.asterisk.org/r/442/

2009-12-08 18:00 +0000 [r233692]  Russell Bryant <russell@digium.com>

	* formats/format_sln.c, formats/format_wav.c,
	  formats/format_ogg_vorbis.c, formats/format_sln16.c,
	  formats/format_wav_gsm.c, formats/format_siren7.c,
	  formats/format_ilbc.c, formats/format_vox.c,
	  formats/format_pcm.c, formats/format_h263.c,
	  formats/format_g723.c, formats/format_h264.c,
	  formats/format_g726.c, formats/format_siren14.c,
	  formats/format_jpeg.c, formats/format_gsm.c,
	  formats/format_g729.c: Set a module load priority for format
	  modules. A recent change to app_voicemail made it such that the
	  module now assumes that all format modules are available while
	  processing voicemail configuration. However, when autoloading
	  modules, it was possible that app_voicemail was loaded before the
	  format modules. Since format modules don't depend on anything,
	  set a module load priority on them to ensure that they get loaded
	  first when autoloading. This fix applies to trunk, 1.6.1, and
	  1.6.2. The fix for 1.4 and 1.6.0 will require a different
	  approach since the module load priority functionality is not
	  present in the module API. (issue #16412) Reported by: jiddings

2009-12-07 23:28 +0000 [r233611]  David Vossel <dvossel@digium.com>

	* main/utils.c: fixes incorrect logic in ast_uri_encode issue
	  #16299

2009-12-07 23:10 +0000 [r233577]  Atis Lezdins <atis@iq-labs.net>

	* contrib/valgrind.supp: Fix compatibility with valgrind 3.3 and
	  older. (noticed in issue #16388) Reported by: parisioa Patches:
	  valgrind.supp uloaded by atis (license 242) Tested by: atis,
	  parisioa

2009-12-07 19:48 +0000 [r233545]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c: Fix TCP Client interface Fix a couple of
	  very minor bugs that prevent the socket client from working. The
	  wrong set of properties were used in one place and the size of
	  the address variable isn't set if the host name is an ip address.
	  Also includes a fix for a bug that was introduced previously.
	  (closes issue #16121) Reported by: thedavidfactor Tested by:
	  thedavidfactor Review: https://reviewboard.asterisk.org/r/439/

2009-12-07 18:08 +0000 [r233472]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 233471 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
	  | 9 lines fixes missing Contact header angle brackets (closes
	  issue #16298) Reported by: mgernoth Patches:
	  reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
	  by: dvossel ........

2009-12-07 17:59 +0000 [r233468]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/jabber.h, CHANGES, res/res_jabber.c: Add
	  applications JabberJoin, JabberLeave, JabberSendGroup for XMPP
	  groupchat (closes issue #14352) Reported by: fiddur Patches:
	  trunk-14352-2.diff uploaded by phsultan (license 73) Tested by:
	  fiddur

2009-12-07 16:14 +0000 [r233394]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Do not reject SDP packets describing only
	  non audio streams. (closes issue #16387) Reported by: zalex1953
	  Patches: media-level-c-fix1.diff uploaded by mnicholson (license
	  96) Tested by: mnicholson, zalex1953

2009-12-06 07:01 +0000 [r233358]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/compat.h, main/strcompat.c, main/app.c: Move
	  implementation of closefrom(3) from app.c to strcompat.c

2009-12-04 21:54 +0000 [r233280]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, /: Merged revisions 233279 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04
	  Dec 2009) | 7 lines clarify requirecalltoken option in
	  iax.sample.conf (closes issue #16223) Reported by: bklang
	  Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
	  (license 919) ........

2009-12-04 21:06 +0000 [r233239]  Tilghman Lesher <tlesher@digium.com>

	* main/translate.c: Using the builtin function breaks OpenBSD 4.2
	  (closes issue #16395) Reported by: jtodd

2009-12-04 20:21 +0000 [r233121-233235]  David Vossel <dvossel@digium.com>

	* CHANGES: update CHANGES file for .m3u support in Mp3Player
	  application

	* apps/app_mp3.c: .m3u support for Mp3Player app (closes issue
	  #14823) Reported by: macli Patches: app_mp3.diff1 uploaded by
	  macli (license ) Tested by: macli, dvossel

	* CHANGES: update CHANGES for new queue option,
	  penaltymemberslimit.

	* apps/app_queue.c: changes penaltymemberslimit to use scanf for
	  config value parsing

	* configs/queues.conf.sample, apps/app_queue.c: new queue option,
	  penaltymemberslimit, disregards penalty on too few queue members
	  when enabled (closes issue #14559) Reported by: fiddur Patches:
	  trunk-199584-1.diff uploaded by fiddur (license 678) Tested by:
	  fiddur, dvossel

	* /, apps/app_voicemail.c: Merged revisions 233116 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04
	  Dec 2009) | 6 lines document and rename strip_control() in
	  app_voicemail (closes issue #16291) Reported by: wdoekes ........

2009-12-04 17:18 +0000 [r233100]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 233092 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
	  | 7 lines Only do frame payload check for HOLD frames. This code
	  was added for helping to debug the source of invalid HOLD frames.
	  However, a side effect of this is that it will incorrectly report
	  errors for frames that have an integer payload. Make the check
	  for this block specific to the HOLD frame case. ........

2009-12-04 17:15 +0000 [r233093]  Matthias Nick <mnick@digium.com>

	* pbx/pbx_config.c: Parse global variables or expressions in hint
	  extensions Parse global variables or expressions in hint
	  extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
	  (closes issue #16166) Reported by: rmudgett Tested by: mnick,
	  rmudgett

2009-12-04 16:55 +0000 [r233059-233089]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Let's unlock the lines list after the
	  AST_LIST_TRAVERSE instead of inside it.

	* channels/chan_skinny.c: Only assign line and device in
	  handle_transfer_button when we have a subchannel. (closes issue
	  #16040) Reported by: ebroad

2009-12-04 16:08 +0000 [r233050]  Tilghman Lesher <tlesher@digium.com>

	* addons/res_config_mysql.c: Update the mysql driver to always
	  return NULL columns, as this is needed for the realtime API to
	  work correctly. (closes issue #16138) Reported by: sohosys
	  Patches: 20091029__issue16138.diff.txt uploaded by tilghman
	  (license 14) Tested by: sohosys

2009-12-04 15:38 +0000 [r233046]  Matthias Nick <mnick@digium.com>

	* /, main/dsp.c: Merged revisions 233014 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
	  11 lines Warning message gets displayed only once Added
	  additional field 'int display_inband_dtmf_warning', which when
	  set to '1' displays the warning ('Inband DTMF is not supported on
	  codec %s. Use RFC2833'), and when set to '0' doesn't display the
	  warning. Otherwise you would get hundreds of warnings every
	  second. (closes issue #15769) Reported by: falves11 Patches:
	  patch_15769_14.txt uploaded by mnick (license 874) Tested by:
	  mnick, falves11 ........

2009-12-04 05:26 +0000 [r232854-232982]  Tilghman Lesher <tlesher@digium.com>

	* res/res_pktccops.c: Buildbot complained

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  res/res_pktccops.c: OS X does not define MSG_NOSIGNAL, but it
	  does have a socket option SO_NOSIGPIPE. (closes issue #16178)
	  Reported by: oej

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
	  pagerdateformat, to allow shorter dates for SMS messages. (closes
	  issue #16263) Reported by: andrew Patches: pagerdate.patch
	  uploaded by andrew (license 240) (with a slight modification by
	  me)

	* /, apps/app_voicemail.c: Merged revisions 232820 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03
	  Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change
	  the use of language codes so that language registers as a prefix,
	  rather than an exact match. (closes issue #16272) Reported by:
	  patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by
	  tilghman (license 14) ........

2009-12-03 20:26 +0000 [r232853]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooh245.c: jitterbuffer setup correction
	  correction of double pointer references from previous rev

2009-12-03 08:47 +0000 [r232738-232771]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Replaced two deprecated functions of OSP
	  Toolkit.

	* apps/app_osplookup.c: Added custom info support.

2009-12-03 00:38 +0000 [r232700]  Jeff Peeler <jpeeler@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Extend voicemail to allow IMAP folders to be specified per
	  mailbox. Previously only possible per context, new option called
	  imapfolder. (closes issue #14298) Reported by: jablko Patches:
	  patch-200906202 uploaded by jablko (license 675)

2009-12-03 00:09 +0000 [r232660-232661]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Remove debugging line

	* include/asterisk/astobj2.h, res/res_musiconhold.c: Fix multiple
	  issues with musiconhold, which led to classes not getting
	  destroyed properly. * Classes are now tracked past removal from
	  the core container, and module removal is actively prevented
	  until all references are freed. * A hanging reference stored in
	  the channel has been removed. This could have caused a mismatch
	  and the music state not properly cleared, if two or more reloads
	  occurred between MOH being stopped and MOH being restarted. * In
	  certain circumstances, duplicate classes were possible. * A race
	  existed at reload time between a process being killed and the
	  thread responsible for reading from the related pipe respawning
	  that process. * Several reference counts have also been
	  corrected. At least one could have caused deleted classes to
	  stick around forever, consuming resources. This originally
	  manifested as MOH external processes that were not killed at
	  reload time. (closes issue #16279, closes issue #16207) Reported
	  by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt
	  uploaded by tilghman (license 14) Tested by: parisioa, tilghman

2009-12-02 23:27 +0000 [r232657]  David Vossel <dvossel@digium.com>

	* UPGRADE.txt, CHANGES: update CHANGES and UPGRADE.txt for early
	  media behavior change between 1.6.1 and 1.6.2 (closes issue
	  #16212) Reported by: miki

2009-12-02 22:17 +0000 [r232587]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c: Prevent double closing of FDs by EIVR
	  This caused a problem when asterisk was under heavy load and
	  running both AGI and EIVR applications. EIVR would close an FD at
	  which point it would be considered freed and be used by a new AGI
	  instance the second close would then close the FD now in use by
	  AGI. (closes issue #16305) Reported by: diLLec Tested by:
	  thedavidfactor, diLLec Review:
	  https://reviewboard.asterisk.org/r/436/

2009-12-02 22:02 +0000 [r232582]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c, /: Merged revisions 232581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
	  | 7 lines Send ack (response/message) after receiving manager
	  action userevent (closes issue #16264) Reported by: dimas
	  Patches: event-ack.patch uploaded by dimas (license 88) ........

2009-12-02 21:37 +0000 [r232580]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c: Fix support for multiline SMS messages in
	  chan_mobile. (closes issue #16278) Reported by: Artem Patches:
	  multiline-sms-fix2.diff uploaded by mnicholson (license 96)
	  Tested by: Artem

2009-12-02 21:32 +0000 [r232576]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: Make manager response to "Action: events" finish
	  with empty line (closes issue #16275) Reported by: vnovy Patches:
	  manager.c.diff uploaded by vnovy (license 922)

2009-12-02 21:13 +0000 [r232544]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c: Do something with the service indicator so
	  that asterisk does not attempt to use a chan_mobile endpoint that
	  does not have service. (closes issue #16132) Reported by: nikkk
	  Patches: service-indicator2.diff uploaded by mnicholson (license
	  96) Tested by: nikkk

2009-12-02 20:10 +0000 [r232442-232510]  Joshua Colp <jcolp@digium.com>

	* CHANGES, main/asterisk.c, doc/asterisk.sgml: Add an 'X' option to
	  the asterisk application which enables #exec for configuration
	  files. This option can be used to enable #exec support in the
	  asterisk.conf configuration file. (closes issue #16260) Reported
	  by: atis Patches: exec_includes.patch uploaded by atis (license
	  242)

	* apps/app_record.c, CHANGES: Add an option to Record which enables
	  a mode where any DTMF digit will terminate recording. (closes
	  issue #15436) Reported by: Vince Patches: app_record.diff
	  uploaded by Vince (license 823) Tested by: dbrooks

2009-12-02 17:18 +0000 [r232365]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Do not change the exten string field or
	  rebuild the contact header on an inbound sip_pvt if the outbound
	  call is redirected.

2009-12-02 17:06 +0000 [r232356]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_amd.c: Merged revisions 232355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
	  lines Fix a bug where if you hung up very quickly after calling
	  AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
	  (closes issue #16239) Reported by: CGMChris ........

2009-12-02 17:00 +0000 [r232351]  David Vossel <dvossel@digium.com>

	* /, main/acl.c: Merged revisions 232350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
	  | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
	  strace. (closes issue #16290) Reported by: wdoekes ........

2009-12-02 16:40 +0000 [r232345]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Add support for handling the 415 Unsupported
	  media type response like we do for a 488 Not acceptable here
	  response. (closes issue #16186) Reported by: atis Patches:
	  sip_t38_response_415.patch uploaded by atis (license 242)

2009-12-02 15:42 +0000 [r232269]  David Vossel <dvossel@digium.com>

	* funcs/func_groupcount.c, /: Merged revisions 232268 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02
	  Dec 2009) | 9 lines fixes segfault in func_groupcount closes
	  issue #16337) Reported by: Parantido Patches: issue_16337.diff
	  uploaded by dvossel (license 671) Tested by: Parantido, dvossel
	  ........

2009-12-02 14:54 +0000 [r232230]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where a scheduled item ID would
	  get retained on registrations in a certain scenario causing code
	  to execute during reload that should not. (issue AST-263)

2009-12-02 03:26 +0000 [r232164]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, main/strcompat.c, configure.ac: So
	  apparently, some platforms don't have ffsll(3). The manpage lies;
	  it says that the function is in POSIX, but that's only for
	  ffs(3), not ffsll(3).

2009-12-02 00:45 +0000 [r232091]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 232090 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01
	  Dec 2009) | 10 lines Do not modify the gain settings on data
	  calls. (The digital flag actually represents a data call.)
	  (closes issue #15972) Reported by: udosw Patches:
	  transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
	  Tested by: alecdavis ........

2009-12-01 23:56 +0000 [r232008-232017]  Russell Bryant <russell@digium.com>

	* main/translate.c: Use __builtin_ffsll() from gcc instead of
	  ffssll() to fix a FreeBSD build error.

	* funcs/func_lock.c: Fix a build error on FreeBSD.

	* /, main/file.c: Merged revisions 232007 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
	  | 2 lines Fix a warning pointed out by buildbot. ........

2009-12-01 21:54 +0000 [r231927]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 231911 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
	  | 12 lines Fix crash with invalid frame data The crash was
	  happening as a result of a frame containing an invalid data
	  pointer, but was set with data length of zero. The few times the
	  issue was reproduced it _seemed_ that the frame was queued
	  properly, that is the data pointer was set to NULL. I never could
	  reproduce the crash so as a last resort the crash has been fixed,
	  but a check in __ast_read has been added to give as much
	  information about the source of problematic frames in the future.
	  (closes issue #16058) Reported by: atis ........

2009-12-01 21:20 +0000 [r231867]  David Vossel <dvossel@digium.com>

	* main/pbx.c, /: Merged revisions 231853 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
	  | 3 lines WaitExten m option with no parameters generates frame
	  with zero datalen but non-null data ptr ........

2009-12-01 20:27 +0000 [r231814-231850]  Tilghman Lesher <tlesher@digium.com>

	* res/res_rtp_asterisk.c, channels/chan_unistim.c,
	  main/rtp_engine.c, addons/chan_ooh323.c, channels/chan_sip.c,
	  res/res_adsi.c, addons/chan_ooh323.h,
	  include/asterisk/callerid.h, channels/chan_phone.c,
	  channels/chan_dahdi.c, channels/chan_skinny.c, main/callerid.c,
	  channels/chan_h323.c, addons/ooh323cDriver.c,
	  include/asterisk/rtp_engine.h, addons/ooh323cDriver.h: More
	  32->64 bit codec conversions. In the process of swapping ULAW to
	  a place in the extended codec space, we found several unhandled
	  cases, where a 32-bit integer was still being used to handle a
	  codec field. Most of these have been fixed with this commit,
	  although there is at least one case (codec_dahdi) which depends
	  upon outside headers to be altered before a conversion can be
	  made. (Fixes AST-278, SWP-459)

	* include/asterisk/mod_format.h: Formats need to be able to
	  represent all 64 codec bits.

2009-12-01 15:47 +0000 [r231741]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/file.c: Merged revisions 231740 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
	  2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
	  and return an error if no know formats are found. ........

2009-11-30 21:47 +0000 [r231692]  Kevin P. Fleming <kpfleming@digium.com>

	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
	  Another round of UDPTL stack fixes/improvements: 1) Allow users
	  of UDPTL stack to associate a character-string tag with a UDPTL
	  session, so that log/error/debug messages generated by the UDPTL
	  stack can be 'connected' to the endpoint that caused them to be
	  generated. 2) Improve comments (and process) of calculating the
	  far end's maximum IFP size when redundancy mode is in use for
	  error correction. 3) When an IFP larger than the calculated 'far
	  max IFP' size is presented for writing, truncate it rather than
	  putting in the buffer and allowing the buffer to overflow; this
	  will cause the ends to retrain to a lower bit rate that produces
	  IFPs of an appropriate size if possible, and if not possible, the
	  FAX transfer will fail completely. In these cases, it is due to
	  the one endpoint supplying a T38FaxMaxDatagram value that is
	  improperly calculated and is too low to be of use; we have
	  configuration options available to override this behavior. 4)
	  Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
	  longer needed.

2009-11-30 21:31 +0000 [r231616-231688]  Matthew Nicholson <mnicholson@digium.com>

	* include/asterisk/file.h, /, main/file.c, main/app.c,
	  apps/app_voicemail.c: Merged revisions 231614 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
	  2009) | 8 lines Remove duplicate entries from voicemail format
	  lists. This prevents app_voicemail from entering an infinite loop
	  when the same format is specified twice in the format list.
	  (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
	  Review: https://reviewboard.asterisk.org/r/429/ ........

	* include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
	  Reverted 231616

	* include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
	  Merged revisions 231614 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
	  2009) | 8 lines Remove duplicate entries from voicemail format
	  lists. This prevents app_voicemail from entering an infinite loop
	  when the same format is specified twice in the format list.
	  (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
	  Review: https://reviewboard.asterisk.org/r/429/ ........

2009-11-30 20:44 +0000 [r231602]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When receiving SDP that matches the version
	  of the last one do not treat it as a fatal error. (closes issue
	  #16238) Reported by: seandarcy

2009-11-30 18:55 +0000 [r231491-231556]  David Vossel <dvossel@digium.com>

	* apps/app_queue.c: app_queue crashes randomly, often during
	  call-transfers This patch adds a ref to the queue_ent object's
	  parent call_queue in queue_exec() so the call_queue won't be
	  destroyed while the the queue_ent still holds a pointer to it.
	  (closes issue 0015686) Tested by: dvossel, aragon

	* res/res_rtp_asterisk.c, /: Merged revisions 231441 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30
	  Nov 2009) | 11 lines fixes crash caused by RTP comfort noise
	  payload greater than 24 bytes AST-2009-010 (closes issue #16242)
	  Reported by: amorsen Patches: issue16242.diff uploaded by oej
	  (license 306) Tested by: amorsen, oej, dvossel ........

2009-11-30 16:53 +0000 [r231439]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.dynamics (added), Makefile.rules: Export dynamic
	  (weak-linked) symbols correctly. (closes issue #15193) Reported
	  by: eliel Patches: 20091111__issue15193.diff.txt uploaded by
	  tilghman (license 14)

2009-11-30 16:29 +0000 [r231436]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where an immediate masquerade
	  would cause a queued unhold frame to get lost. Now we just
	  indicate unhold directly after the masquerade is complete. (issue
	  ABE-2011)

2009-11-27 08:47 +0000 [r231401]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: 1. Modified exported variable names. 2.
	  Added destination port support. 3. Added new protocols. 4. Added
	  QoS.

2009-11-26 02:09 +0000 [r231299-231369]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES, main/asterisk.c: Reorder option flags.
	  Change guidelines so that example code is consistent with
	  guidelines

	* main/channel.c, /: Merged revisions 231298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
	  | 2 lines After a frame duplication failure, unlock the channel
	  before returning. ........

2009-11-25 15:42 +0000 [r231189]  Matthew Nicholson <mnicholson@digium.com>

	* pbx/pbx_lua.c: Load pbx_lua with global symbols to allow linking
	  with other lua libraries. Found by Maxim Litnitskiy.

2009-11-24 20:31 +0000 [r231134]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Found a few places where queue refcounts were
	  counted incorrectly. Also add debug statements. (closes issue
	  #15982, closes issue #15984) Reported by: atis Patches:
	  20091111__issue15982.diff.txt uploaded by tilghman (license 14)
	  Tested by: atis

2009-11-24 18:50 +0000 [r231058-231095]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Fix erroneous hangup extension execution
	  ast_spawn_extension behaves differently from 1.4 in that hangups
	  and extensions that do not exist do not return an error, whereas
	  in 1.6 it does. This is now taken into account so that the
	  AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue
	  #16106) Reported by: ajohnson Tested by: ajohnson

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Fix problem on digital channels due to digital flag not getting
	  set Changed areas in sig_pri to set the digital flag using a
	  callback that will also set the corresponding flag in chan_dahdi.
	  Modified dahdi_request slightly so that if a bearer is marked as
	  digital, that information is available when creating the new
	  channel. (closes issue #16151) Reported by: alecdavis Patch based
	  on bug_16151.diff.txt uploaded by alecdavis (license 585)

2009-11-24 13:52 +0000 [r231025]  Matthew Nicholson <mnicholson@digium.com>

	* CHANGES: Updated CHANGES file to describe the new 'd' option to
	  app_followme added in r230964 (related to issue #14155) Reported
	  by: junky

2009-11-24 04:58 +0000 [r230994]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, funcs/func_strings.c, CHANGES: Add
	  REPLACE & PASSTHRU functions, overhaul of func_strings, fix API
	  docs for the ast_get_encoded_* functions. * Add REPLACE function,
	  which searches a given variable for a set of characters and
	  replaces each with a given character. * Add PASSTHRU function,
	  which passes a literal string back, like a NoOp for functions.
	  Intent is to be able to specify a literal string to another
	  function that takes a variable name as an argument. * Let the
	  array manipulation functions work with dialplan functions, in
	  addition to variables. This allows the array manipulation
	  functions to modify ASTDB and ODBC backends, assuming the
	  func_odbc configuration has both read and write functions.
	  (closes issue #15223) Reported by: ajohnson Patches:
	  20091112__issue15223.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen, tilghman

2009-11-23 22:37 +0000 [r230964]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_followme.c: Add an option to app_followme to disable the
	  "please hold" announcement. (closes issue #14155) Reported by:
	  junky Patches: M14555-trunk.diff uploaded by junky (license 177)
	  (modified) Tested by: junky

2009-11-23 15:45 +0000 [r230881]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Change fax
	  detection in chan_sip so it behaves as one would expect.
	  Internally the way T.38 is negotiated has changed and the option
	  no longer reflects a behavior that is valid. It will now look for
	  a CNG tone on received calls and if present send the call to the
	  'fax' extension. It is then up to the application or channel to
	  request the switch over to T.38.

2009-11-23 15:34 +0000 [r230773-230877]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 230839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
	  2009) | 1 line Correct fix for issue #16268... the reporter's
	  original patch was very close to correct. ........

	* /, channels/chan_sip.c: Merged revisions 230772 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
	  2009) | 5 lines Ensure that SDP parsing does not ignore the last
	  line of the SDP. (closes issue #16268) Reported by: sgimeno
	  ........

2009-11-20 22:35 +0000 [r230726]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes iax2 show cache locking error, thanks
	  alecdavis! (closes issue #16094) Reported by: alecdavis Patches:
	  bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
	  alecdavis, dvossel

2009-11-20 21:47 +0000 [r230697]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/unaligned.h: Revert code in error and include
	  the gcc suggested workaround for the original problem, while gcc
	  investigates.

2009-11-20 21:01 +0000 [r230628]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/features.c: Merged revisions 230627 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
	  2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
	  if it exists. This is necessary for the recordagentcalls option
	  in chan_agent to store the recorded file name in the bridge CDR.
	  (closes issue #14590) Reported by: msetim Patches:
	  queue_agent_userfield.patch uploaded by Laureano (license 265)
	  Tested by: Laureano, mnicholson ........

2009-11-20 17:28 +0000 [r230584]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt, apps/app_externalivr.c: Fix/Implement error
	  events for non-existing files also include a better cmd define
	  for S command Review: https://reviewboard.asterisk.org/r/430/

2009-11-20 17:26 +0000 [r230509-230583]  David Vossel <dvossel@digium.com>

	* include/asterisk/audiohook.h, main/audiohook.c: audiohook signal
	  trigger on every status change (issue #14618) Review:
	  https://reviewboard.asterisk.org/r/434/

	* /, apps/app_mixmonitor.c: Merged revisions 230508 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19
	  Nov 2009) | 10 lines fixes MixMonitor thread not exiting when
	  StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
	  Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
	  671) Tested by: dvossel, AlexMS Review:
	  https://reviewboard.asterisk.org/r/424/ ........

2009-11-19 14:53 +0000 [r230438]  David Ruggles <thedavidfactor@gmail.com>

	* apps/app_externalivr.c: Basic cleanup of ExternalIVR: cleaned up
	  argument parsing; implemented good coding practices where
	  applicable; replaced most notice level logging with verbose
	  logging; replaced warning messages that terminated with error
	  messages; fixed memory leak identified by russellb

2009-11-16 16:40 +0000 [r230343-230381]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Fix another buglet in T.38 session teardown at
	  the end of FAX sessions.

	* apps/app_fax.c: Ensure that only one end of a T.38 session
	  initiates teardown at completion.

2009-11-16 01:49 +0000 [r230314]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: 1. Added SIP Diversion support. 2. Fixed
	  compile warning for UUID.

2009-11-15 17:23 +0000 [r230247]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 230246 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15
	  Nov 2009) | 6 lines Correct mistaken option name in error
	  message. The configuration option for allowing hosts to make
	  non-token-based calls is 'calltokenoptional', not
	  'calltokenignore'. (reported on asterisk-users) ........

2009-11-15 07:53 +0000 [r230217]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/channel.h: Increase maximum length of language
	  buffers (closes issue #16217) Reported by: dsessions

2009-11-13 22:00 +0000 [r230145]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 230144 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
	  lines Respect the maddr parameter in the Via header. (closes
	  issue #14446) Reported by: frawd Patches: via_maddr.patch
	  uploaded by frawd (license 610) Tested by: frawd ........

2009-11-13 20:42 +0000 [r230111]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_dial.c, channels/chan_sip.c, apps/app_meetme.c,
	  apps/app_fax.c, configs/manager.conf.sample,
	  res/res_musiconhold.c, include/asterisk/manager.h,
	  channels/chan_iax2.c, apps/app_queue.c, CHANGES,
	  res/res_monitor.c, main/cdr.c, main/channel.c, main/manager.c,
	  main/features.c, apps/app_minivm.c, apps/app_chanspy.c,
	  apps/app_voicemail.c: Display a list of channel variables in each
	  channel-oriented event. (Closes AST-33) Reviewboard:
	  https://reviewboard.asterisk.org/r/368/

2009-11-13 19:44 +0000 [r229912-230039]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 230038 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov
	  2009) | 9 lines Fix a crash caused by two threads thinking they
	  should both free the chan_local private structure when only one
	  should. (closes issue #15314) Reported by: sroberts Patches:
	  Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
	  780) Tested by: davidw, lottc ........

	* UPGRADE.txt, apps/app_chanisavail.c, CHANGES: Store the cause
	  code that is returned when trying to create a channel in
	  ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of
	  overwriting the device state in AVAILSTATUS. (closes issue
	  #14426) Reported by: macli

	* /: Merged revisions 229965 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
	  lines Document a limitation in the AVAILSTATUS variable from
	  ChanIsAvail and provide a workaround for it that does not change
	  existing behavior. (closes issue #14426) Reported by: macli
	  ........

	* channels/chan_sip.c: Fix T.38 negotiation regression introduced
	  with the SDP parser changes.

2009-11-13 10:53 +0000 [r229819-229871]  Olle Johansson <oej@edvina.net>

	* main/loader.c: Fixing trunk in a way so that it compiles again.
	  Thanks, Philippe :-)

	* addons/cdr_mysql.c: If CDR logging is disabled, it's considered a
	  FAILURE

	* configs/modules.conf.sample, CHANGES, main/asterisk.c,
	  main/loader.c: Add the capability to require a module to be
	  loaded, or else Asterisk exits. Review:
	  https://reviewboard.asterisk.org/r/426/

2009-11-13 03:16 +0000 [r229788]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Added full number portability parameter
	  support.

2009-11-12 23:43 +0000 [r229750-229754]  Jason Parker <jparker@digium.com>

	* configs/alsa.conf.sample: Update sample config for ALSA mute and
	  noaudiocapture

	* channels/chan_alsa.c: Add mute functionality. Add config option
	  to not try to open capture device. Adds "console {mute|unmute}"
	  CLI command. Adds mute and noaudiocapture config options (will
	  update sample configs shortly). (closes issue #14673) Reported
	  by: Nick_Lewis Patches: chan_alsa.c-oneway3.patch uploaded by
	  Nick Lewis (license 657) Tested by: qwell

	* channels/chan_oss.c: Fix mute toggling on OSS channels.

2009-11-12 16:44 +0000 [r229670]  David Vossel <dvossel@digium.com>

	* funcs/func_audiohookinherit.c, /: Merged revisions 229669 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
	  | 6 lines fixes merging error, datastore was being freed in the
	  wrong function. (closes issue #16219) Reported by: aragon
	  ........

2009-11-12 13:54 +0000 [r229639]  Leif Madsen <lmadsen@digium.com>

	* configs/sip.conf.sample: Update sip.conf.sample. Just updating a
	  spelling error and some capitalization in a documentation update
	  that Olle added. May the Swenglish be with you.

2009-11-12 10:24 +0000 [r229606-229607]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Clarification

	* configs/sip.conf.sample: Clarify some security issues early in
	  the sample configuration

2009-11-11 20:47 +0000 [r229568]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt: Remove non-functional feature from
	  ExternalIVR documentation Remove non-functional socket
	  implementation of ExternalIVR from documentation (closes issue
	  #16225) Reported by: thedavidfactor Patches:
	  externalivr.txt.20091111.1542.patch uploaded by thedavidfactor
	  (license 903)

2009-11-11 19:48 +0000 [r229460-229499]  David Brooks <dbrooks@digium.com>

	* main/pbx.c, /: Merged revisions 229498 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
	  | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
	  crash if NULL is passed to ast_log. This simple patch simply uses
	  S_OR to get around this. (closes issue #15392) Reported by:
	  yrashk ........

	* apps/app_softhangup.c: Flags not initialized in app_softhangup.c,
	  causing undefined behavior Trivial patch [kobaz] to initialize an
	  ast_flags = {0} (closes issue #16129) Reported by: kobaz

2009-11-11 14:30 +0000 [r229431]  Leif Madsen <lmadsen@digium.com>

	* CHANGES: Update CHANGES file. Updating the CHANGES file after
	  noticing an email on the asterisk-dev mailing list from Russell.
	  (issue #15874)

2009-11-10 22:14 +0000 [r229361]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 229360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
	  | 12 lines If two pattern classes start with the same digit and
	  have the same number of characters, they will compare equal. The
	  example given in the issue report is that of [234] and [246],
	  which have these characteristics, yet they are clearly not
	  equivalent. The code still uses these two characteristics, yet
	  when the two scores compare equal, an additional check will be
	  done to compare all characters within the class to verify
	  equality. (closes issue #15421) Reported by: jsmith Patches:
	  20091109__issue15421__2.diff.txt uploaded by tilghman (license
	  14) Tested by: jsmith, thedavidfactor ........

2009-11-10 22:01 +0000 [r229356]  David Ruggles <thedavidfactor@gmail.com>

	* doc/externalivr.txt: Merged revisions 229355 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
	  2009) | 9 lines Fix ExternalIVR Documentation Remove
	  documentation for event that doesn't function (closes issue
	  #16220) Reported by: thedavidfactor Patches:
	  externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
	  (license 903) ........

2009-11-10 21:22 +0000 [r229351]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: When GOSUB is invoked within an AGI, it may not
	  exit correctly. (closes issue #16216) Reported by: atis Patches:
	  20091110__atis_work.diff.txt uploaded by tilghman (license 14)
	  Tested by: atis

2009-11-10 20:06 +0000 [r229282]  Joshua Colp <jcolp@digium.com>

	* /, codecs/codec_g726.c: Merged revisions 229281 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
	  lines Remove broken support for direct transcoding between G.726
	  RFC3551 and G.726 AAL2. On some systems the translation core
	  would actually consider g726aal2 -> g726 -> signed linear to be a
	  quicker path then g726aal2 -> signed linear which exposed this
	  problem. (closes issue #15504) Reported by: globalnetinc ........

2009-11-10 17:33 +0000 [r229228]  David Ruggles <thedavidfactor@gmail.com>

	* /, doc/externalivr.txt: Merged revisions 229191 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
	  2009) | 11 lines Document ExternalIVR event tag collision
	  ExternalIVR uses the D tag for two different event types. This
	  documents that behavior and how to differentiate between the two
	  cases. Also includes a minor spelling fix and clarification
	  (closes issue #16211) Reported by: thedavidfactor Patches:
	  externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
	  (license 903) ........

2009-11-10 17:16 +0000 [r229168]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 229167 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10
	  Nov 2009) | 9 lines don't crash on log message in solaris
	  AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
	  bklang ........

2009-11-10 15:53 +0000 [r229102]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Reverted revision 201717. (closes issue
	  0016175) Reported by: paul-tg

2009-11-10 15:27 +0000 [r229093]  David Vossel <dvossel@digium.com>

	* res/res_config_pgsql.c: fixes pgsql double free of threadstorage
	  A thread storage variable was being freed incorrectly, which
	  resulted in a double free if two queries were made in the same
	  thread. (closes issue #16011) Reported by: cristiandimache
	  Patches: issue16011.diff uploaded by dvossel (license 671)

2009-11-10 11:16 +0000 [r229050]  Gavin Henry <ghenry@suretecsystems.com>

	* contrib/scripts/asterisk.ldap-schema: Schema file additions *
	  Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox
	  objectClasses to allow standalone dialplan, account and mailbox
	  entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
	  AstAccountTransport, AstAccountPromiscRedir, -
	  AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
	  - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
	  redundant IPaddr (there's already IPAddress) - Gives more
	  configuration Flags for SIP-Users available (tested) - Allows to
	  create Asterisk Attributes in defined Asterisk ObjectClasses
	  without extensibleObject (which really should be the last
	  resort); gives also additional possibilities for LDAP-filter
	  (closes issue #15874) Reported by: Medozas Patches:
	  asterisk.ldap-schema.patch uploaded by Medozas (license 41)
	  Tested by: Medozas, suretec

2009-11-09 22:50 +0000 [r229015]  Terry Wilson <twilson@digium.com>

	* channels/chan_local.c: Don't crash when bridge->tech_pvt == NULL
	  This is a similar solution to what is in place for chan_agent
	  (closes issue #16003) Reported by: atis Tested by: twilson

2009-11-09 17:17 +0000 [r228979]  Tilghman Lesher <tlesher@digium.com>

	* channels/iax2-parser.c: Don't try to convert a 64-bit integer,
	  where only a 32-bit integer is stored. (closes issue #16194)
	  Reported by: habile

2009-11-09 16:28 +0000 [r228947]  Matthew Nicholson <mnicholson@digium.com>

	* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add the
	  'relative-periodic-announce' option to app_queue to allow for
	  calculating the time of announcments from the end of the previous
	  announcment rather than from the beginning. (closes issue #15260)
	  Reported by: tonils

2009-11-09 15:38 +0000 [r228897]  Leif Madsen <lmadsen@digium.com>

	* main/channel.c, /: Merged revisions 228896 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
	  | 6 lines Update WARNING message. Update a WARNING message to
	  give a suggested fix when encountered. (closes issue #16198)
	  Reported by: atis Tested by: atis ........

2009-11-09 14:37 +0000 [r228858]  Matthew Nicholson <mnicholson@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 228827 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon,
	  09 Nov 2009) | 8 lines Perform limited bounds checking when
	  destroying ast_mutex_t structures to make sure we don't try to
	  use negative indices. (closes issue #15588) Reported by: zerohalo
	  Patches: 20090820__issue15588.diff.txt uploaded by tilghman
	  (license 14) Tested by: zerohalo ........

2009-11-09 07:37 +0000 [r228798]  Tilghman Lesher <tlesher@digium.com>

	* addons/cdr_mysql.c, main/event.c, channels/chan_console.c,
	  res/res_pktccops.c, main/loader.c: Fix various problems detected
	  with Valgrind. * chan_console accessed pvts after deallocation. *
	  cdr_mysql stored a pointer that was freed by realloc() * The
	  module loader did not check usecount on shutdown, which led to
	  chan_iax2 reading a timer that was already unloaded. * The event
	  subsystem sometimes creates an event with no IEs. Due to a corner
	  condition, the code would read beyond the memory boundary. *
	  res_pktccops did not correctly check whether its monitor thread
	  was started. (closes issue #16062) Reported by: alexanderheinz
	  Patches: 20091109__issue16062.diff.txt uploaded by tilghman
	  (license 14) Tested by: tilghman

2009-11-07 17:02 +0000 [r228766]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/init.d/rc.debian.asterisk: Add LSB headers to the Debian
	  init.d script See also issue #14864 .

2009-11-06 22:35 +0000 [r228693]  David Vossel <dvossel@digium.com>

	* main/channel.c, /: Merged revisions 228692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
	  | 9 lines fixes audiohook write crash occuring in chan_spy
	  whisper mode. After writing to the audiohook list in ast_write(),
	  frames were being freed incorrectly. Under certain conditions
	  this resulted in a double free crash. (closes issue #16133)
	  Reported by: wetwired (closes issue #16045) Reported by:
	  bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
	  671) Tested by: bluecrow76, dvossel, habile ........

2009-11-06 22:32 +0000 [r228691]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, CHANGES, channels/sig_pri.c: Created
	  standard location to add options to chan_dahdi for ISDN dialing.
	  Dial(DAHDI/g1[/extension[/options]]) Current options:
	  K(<keypad_digits>) R Reverse charging indication (Collect calls)
	  The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format
	  was variable and did not allow for the easy addition of more
	  options. The earlier 'C' prefix character for reverse charge
	  indiation would conflict with the a-d DTMF digits if ISDN uses
	  them.

2009-11-06 22:07 +0000 [r228661]  David Brooks <dbrooks@digium.com>

	* tests/test_amihooks.c: ami_testhooks.c automatically registers
	  hook ami_testhooks.c was registering for AMI events upon module
	  load. Moved the registration to its own CLI command. Added CLI
	  command for unregistering the hook. Changed some of the wording,
	  removed unnecessary arguments/parameters. Reported by: rmudgett

2009-11-06 22:02 +0000 [r228658-228659]  Mark Michelson <mmichelson@digium.com>

	* addons/chan_ooh323.c: Make compilation of chan_ooh323 disabled by
	  default. All addons modules should be disabled by default,
	  requiring the user to turn them on if desired. After all, these
	  are addons we're talking about here.

	* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Get
	  chan_ooh323 to compile with gcc 4.2. For some reason, the code
	  compiles just fine with later versions of GCC, but this one
	  requires some weird double casting in order to get rid of all
	  warnings. Whatever.

2009-11-06 19:53 +0000 [r228621]  Richard Mudgett <rmudgett@digium.com>

	* main/frame.c: Fix compiler warning gcc 4.2.4 found

2009-11-06 19:47 +0000 [r228620]  Matthew Nicholson <mnicholson@digium.com>

	* funcs/func_base64.c, /, main/utils.c: Merged revisions 228378 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
	  2009) | 8 lines Properly handle '=' while decoding base64
	  messages and null terminate strings returned from BASE64_DECODE.
	  (closes issue #15271) Reported by: chappell Patches:
	  base64_fix.patch uploaded by chappell (license 8) Tested by:
	  kobaz ........

2009-11-06 19:38 +0000 [r228616]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_nbs.c, addons/chan_mobile.c: Missed these two
	  channel drivers on the codec_bits merge

2009-11-06 18:37 +0000 [r228499-228548]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 228547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
	  lines Don't overwrite caller ID name on a trunk with the
	  configured fullname when using users.conf (issue ABE-1989)
	  ........

	* doc/tex/localchannel.tex: Fix the localchannel.tex file.

2009-11-06 17:22 +0000 [r228420-228441]  David Vossel <dvossel@digium.com>

	* codecs/codec_ilbc.c: Fixes merging issue from 1.4, frame data is
	  held in data.ptr in trunk

	* /, codecs/codec_ilbc.c: Merged revisions 228418 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
	  | 13 lines fixes segfault in iLBC For reasons not yet known, it
	  appears possible for an ast_frame to have a datalen greater than
	  zero while the actual data is NULL during Packet Loss
	  Concealment. Most codecs don't support PLC so this doesn't affect
	  them. This patch catches the malformed frame and prevents the
	  crash from occuring. Additional efforts to determine why it is
	  possible for a frame to look like this are still being
	  investigated. (issue #16979) ........

2009-11-06 16:42 +0000 [r228410]  Joshua Colp <jcolp@digium.com>

	* /, main/abstract_jb.c: Merged revisions 228409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
	  lines Fix a bug caused by a partially invalid frame (from the
	  jitterbuffer) passing through the Asterisk core. (closes issue
	  #15560) Reported by: jvandal (closes issue #15709) Reported by:
	  covici ........

2009-11-06 15:42 +0000 [r228268-228339]  David Vossel <dvossel@digium.com>

	* /, main/astfd.c: Merged revisions 228338 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
	  | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
	  by: slavon ........

	* funcs/func_audiohookinherit.c: fixes memory leak in
	  func_audiohookinherit.c (closes issue #15394) Reported by: boroda
	  Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
	  (license 790) Tested by: dbrooks, boroda

2009-11-05 22:59 +0000 [r228233]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_cdr.c: Fix XML in func_cdr.c

2009-11-05 22:12 +0000 [r228191-228196]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: Yet another error message in the dialplan
	  (thanks, rmudgett/russellb)

	* apps/app_meetme.c: MEETME_INFO should not return a literal error
	  message to the dialplan. (closes issue #15450) Reported by:
	  JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks
	  (license 790) Tested by: JimVanM

2009-11-05 21:23 +0000 [r228189]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_chanspy.c: Fix the fix for chanspy option o In 224178, I
	  assumed the uploaded patch was correct as it had received
	  positive feedback. The flags were being checked in the incorrect
	  location. Upon testing the fix this time it was also found that
	  the flags from the dialplan weren't being copied to the
	  chanspy_translation_helper. (closes issue #16167) Reported by:
	  marhbere

2009-11-05 19:34 +0000 [r228145]  David Brooks <dbrooks@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 228078 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05
	  Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash
	  related to chan_misdn connection. Patch submitted by
	  gknispel_proformatique, tested by francesco_r. "I have many crash
	  since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
	  bt." This patch zeros out an ast_frame. (closes issue #16041)
	  Reported by: francesco_r ........

2009-11-05 19:16 +0000 [r228080]  Jason Parker <jparker@digium.com>

	* channels/chan_vpb.cc, /: Merged revisions 228079 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov
	  2009) | 8 lines Fix crash on VPB exception when no hardware is
	  present. (closes issue #14970) Reported by: tzafrir Patches:
	  vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
	  markwaters ........

2009-11-05 17:26 +0000 [r228015-228049]  Tilghman Lesher <tlesher@digium.com>

	* main/frame.c: Rework codecs command to comply with the 64-bit
	  scheme

	* apps/app_externalivr.c: Don't crash if no arguments are passed.
	  (closes issue #16119) Reported by: thedavidfactor

2009-11-04 23:50 +0000 [r227914-227945]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_monitor.c: Merged revisions 227944 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
	  | 14 lines Fix incorrect filename comparsion after monitor file
	  change The logic to detect if a requested file is indeed a
	  different file from the current file was incorrect. The main
	  issue being confusion of the use of filename_base which was
	  previously set without pathing information and then compared to
	  another full path. Robust file comparison logic has been added to
	  properly check if two files are the same even if symlinks are
	  used. (closes issue #15313) Reported by: caspy Patches:
	  20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
	  325) but mostly tilghman's work ........

	* addons/chan_ooh323.c: Update chan_ooh323 to support the expanded
	  codec bitfield from 227580.

2009-11-04 22:10 +0000 [r227898]  Alexandr Anikin <may@telecom-service.ru>

	* addons/ooh323c/src/oochannels.h,
	  addons/ooh323c/src/ooCmdChannel.h, addons/chan_ooh323.c,
	  addons/ooh323c/src/printHandler.h, addons/ooh323c/src/ooq931.h,
	  addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
	  addons/ooh323c/src/ooasn1.h, addons/ooh323c/src/ootypes.h,
	  addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
	  addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/ooLogChan.h,
	  addons/ooh323c/src/ooCapability.c,
	  addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/dlist.c,
	  addons/ooh323c/src/eventHandler.c,
	  addons/ooh323c/src/ooCapability.h,
	  addons/ooh323c/src/eventHandler.h, addons/Makefile,
	  addons/ooh323cDriver.c, addons/ooh323c/src/ooDateTime.c,
	  addons/ooh323c/src/rtctype.c, addons/ooh323cDriver.h,
	  addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/encode.c,
	  addons/ooh323c/src/ooUtils.c, addons/ooh323c/src/ooGkClient.c,
	  addons/ooh323c/src/ooDateTime.h, addons/ooh323c/src/ooCalls.h,
	  addons/ooh323c/src/ooh323ep.c, addons/ooh323c/src/ooGkClient.h,
	  addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooh323ep.h,
	  addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
	  addons/ooh323c/src/h323/H323-MESSAGESDec.c,
	  addons/ooh323c/src/ooh245.c, addons/ooh323c/src/memheap.h,
	  addons/ooh323c/src/ooh323.h, addons/ooh323c/src/decode.c,
	  addons/ooh323c/src/context.c, addons/ooh323c/src/perutil.c,
	  addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
	  addons/ooh323c/src/ooh245.h, addons/ooh323c/src/ooSocket.c,
	  addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
	  addons/ooh323c/src/oochannels.c,
	  addons/ooh323c/src/ooCmdChannel.c,
	  addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooSocket.h,
	  addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooq931.c,
	  addons/ooh323c/src/ootrace.c: Reworked chan_ooh323 channel
	  module. Many architectural and functional changes. Main changes
	  are threading model chanes (many thread in ooh323 stack instead
	  of one), modifications and improvements in signalling part,
	  additional codecs support (726, speex), t38 mode support. This
	  module tested and used in production environment. (closes issue
	  #15285) Reported by: may213 Tested by: sles, c0w, OrNix Review:
	  https://reviewboard.asterisk.org/r/324/

2009-11-04 21:39 +0000 [r227829-227897]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_dial.c, CHANGES: Added the 'a' option to app dial and
	  modified app_dial to set the answertime when the called channel
	  answers. This change causes answertime to be correct even if the
	  called channel hangs up during an announcement triggered by the
	  A() option. (closes issue #15936) Reported by: falves11 Patches:
	  dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
	  dial-caller-answer1.diff uploaded by mnicholson (license 96)
	  Tested by: falves11, mnicholson

	* apps/app_dial.c, /: Merged revisions 227827 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
	  2009) | 10 lines This patch modifies the Dial application to
	  monitor the calling channel for hangups while playing back
	  announcements. (closes issue #16005) Reported by: falves11
	  Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
	  (license 96) Tested by: mnicholson, falves11 Review:
	  https://reviewboard.asterisk.org/r/407/ ........

2009-11-04 20:35 +0000 [r227824]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/unaligned.h: Fixes for gcc 4.4

2009-11-04 20:13 +0000 [r227759]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Modify the SDP parsing code to parse session
	  and media level items separately. With the new code, media level
	  proprieties should no longer be confused with session level
	  proprieties. This change also reorganizes some of the SDP parsing
	  code which should make it easier to manage in the future. (closes
	  issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
	  file Review: https://reviewboard.asterisk.org/r/414/

2009-11-04 19:26 +0000 [r227712-227739]  Joshua Colp <jcolp@digium.com>

	* /, static-http/prototype.js: Merged revisions 227735 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov
	  2009) | 5 lines Fix a security issue where it may be possible for
	  someone to execute a cross-site AJAX request exploit.
	  (AST-2009-009) ........

	* /, channels/chan_sip.c: Merged revisions 227700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
	  lines Fix a security issue where sending a REGISTER with a
	  differing username in the From URI and Authorization header would
	  reveal whether it was valid or not. (AST-2009-008) ........

2009-11-04 16:41 +0000 [r227646]  Mark Michelson <mmichelson@digium.com>

	* main/frame.c: Add a couple more casts so that code compiles
	  correctly.

2009-11-04 16:35 +0000 [r227645]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/pbx.h: mmichelson reported a compilation error
	  related to codec bit expansion that should be resolved with a
	  simple include of frame_defs.h

2009-11-04 16:25 +0000 [r227643]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: fix trunk building

2009-11-04 16:17 +0000 [r227579-227615]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Two other trunk build
	  fixes (reported by seanbright on #asterisk-dev)

	* addons/format_mp3.c: Fix trunk building

	* main/udptl.c, main/autoservice.c, apps/app_dahdibarge.c,
	  main/frame.c, channels/chan_local.c, main/rtp_engine.c,
	  include/asterisk/autoconfig.h.in, apps/app_record.c,
	  apps/app_test.c, bridges/bridge_softmix.c,
	  apps/app_alarmreceiver.c, codecs/ex_alaw.h, codecs/ex_adpcm.h,
	  formats/format_wav_gsm.c, formats/format_sln16.c,
	  codecs/ex_gsm.h, channels/chan_iax2.c, main/indications.c,
	  res/res_rtp_multicast.c, channels/chan_dahdi.c,
	  include/asterisk/bridging_technology.h, pbx/pbx_spool.c,
	  channels/sig_analog.c, include/asterisk/audiohook.h,
	  channels/chan_skinny.c, configure, main/strcompat.c,
	  include/asterisk/compat.h, formats/format_pcm.c, main/features.c,
	  channels/chan_alsa.c, apps/app_amd.c, formats/format_h263.c,
	  apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c,
	  main/bridging.c, codecs/ex_ulaw.h, apps/app_milliwatt.c,
	  formats/format_gsm.c, apps/app_dial.c, main/pbx.c,
	  formats/format_wav.c, channels/chan_bridge.c, apps/app_echo.c,
	  apps/app_fax.c, include/asterisk/slin.h, channels/chan_agent.c,
	  configure.ac, formats/format_ogg_vorbis.c, apps/app_disa.c,
	  include/asterisk/unaligned.h, codecs/ex_speex.h,
	  include/asterisk/channel.h, apps/app_talkdetect.c,
	  channels/iax2-parser.c, apps/app_speech_utils.c,
	  channels/iax2-parser.h, channels/chan_misdn.c,
	  apps/app_waitforring.c, channels/iax2.h, codecs/codec_dahdi.c,
	  main/audiohook.c, apps/app_chanspy.c, formats/format_g726.c,
	  include/asterisk/frame_defs.h (added),
	  include/asterisk/translate.h, include/asterisk/slinfactory.h,
	  channels/chan_unistim.c, channels/chan_vpb.cc,
	  channels/chan_multicast_rtp.c, formats/format_sln.c,
	  apps/app_meetme.c, apps/app_dictate.c, codecs/ex_g722.h,
	  codecs/ex_g726.h, channels/chan_gtalk.c, res/res_musiconhold.c,
	  apps/app_followme.c, formats/format_siren7.c,
	  include/asterisk/abstract_jb.h, main/asterisk.exports,
	  main/channel.c, formats/format_ilbc.c, channels/chan_phone.c,
	  main/dial.c, main/manager.c, funcs/func_volume.c, res/res_agi.c,
	  apps/app_mp3.c, main/app.c, doc/codec-64bit.txt (added),
	  formats/format_h264.c, include/asterisk/rtp_engine.h,
	  include/asterisk/frame.h, formats/format_siren14.c,
	  codecs/ex_ilbc.h, channels/chan_mgcp.c, apps/app_jack.c,
	  res/res_rtp_asterisk.c, apps/app_nbscat.c, channels/chan_sip.c,
	  codecs/ex_lpc10.h, apps/app_festival.c, main/slinfactory.c,
	  main/translate.c, res/res_adsi.c, channels/chan_console.c,
	  channels/h323/chan_h323.h, channels/sig_pri.c, apps/app_queue.c,
	  channels/chan_oss.c, channels/chan_jingle.c,
	  formats/format_vox.c, include/asterisk/bridging.h,
	  main/abstract_jb.c, main/file.c, channels/chan_h323.c,
	  formats/format_g723.c, codecs/codec_ulaw.c, apps/app_sms.c,
	  include/asterisk/pbx.h, main/dsp.c, formats/format_g729.c: Expand
	  codec bitfield from 32 bits to 64 bits. Reviewboard:
	  https://reviewboard.asterisk.org/r/416/

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  chan_misdn will fail to compile if the redirect_dn member is
	  missing

2009-11-04 08:22 +0000 [r227545]  Olle Johansson <oej@edvina.net>

	* main/manager.c: Add destruction of iterators to avoid problems
	  with refcounters (per Russell's review of another patch)

2009-11-04 03:15 +0000 [r227509]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Don't crash when state_interface is NULL.

2009-11-03 22:13 +0000 [r227462-227464]  Russell Bryant <russell@digium.com>

	* res/res_pktccops.c: Resolve another warning.

	* main/manager.c, pbx/pbx_config.c: Resolve a warning from gcc
	  4.4.1.

	* channels/chan_mgcp.c: Resolve some dev-mode warnings.

2009-11-03 21:26 +0000 [r227448]  David Brooks <dbrooks@digium.com>

	* main/manager.c, include/asterisk/manager.h, tests/test_amihooks.c
	  (added): AMI hook interface This patch, originally submitted by
	  jozza, enables custom modules to send actions to AMI and receive
	  messages from AMI via a hook interface. Included is a simple test
	  module to illustrate the interface. (closes issue #14635)
	  Reported by: jozza Review:
	  https://reviewboard.asterisk.org/r/412/

2009-11-03 21:21 +0000 [r227435]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, apps/app_forkcdr.c, configs/cdr_custom.conf.sample,
	  funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
	  CHANGES: This patch adds a sequence field to CDRs that can be
	  combined with the linkedid or uniqueid field to uniquely identify
	  a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches:
	  cdr-sequence10.diff uploaded by mnicholson (license 96) Tested
	  by: mnicholson

2009-11-03 21:16 +0000 [r227424]  Joshua Colp <jcolp@digium.com>

	* configs/queues.conf.sample, apps/app_queue.c: Add support for
	  using a hint when configuring a state interface using the format
	  hint:<extension>@<context>. (closes issue #15168) Reported by:
	  p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by
	  GameGamer43 (license 894)

2009-11-03 19:59 +0000 [r227372]  Jason Parker <jparker@digium.com>

	* Makefile, main/Makefile: Fix some build issues on Solaris.
	  (closes issue #14517) (SWP-109) Reported by: asgaroth Patches:
	  bug_14517.diff uploaded by snuffy (license 35) Tested by:
	  asgaroth, snuffy, dougm, qwell

2009-11-03 19:48 +0000 [r227361-227368]  Leif Madsen <lmadsen@digium.com>

	* apps/app_controlplayback.c: Change warning message to debug
	  message. app_controlplayback outputs a warning, when in fact it
	  is normal. (closes issue #16071) Reported by: atis Patches:
	  controlplayback_warning.patch uploaded by atis (license 242)

	* configs/extensions.conf.sample: Additional fixes to the
	  extensions.conf.sample file. Update the extensions.conf.sample
	  [stdexten] context so that we use the variable instead of
	  requiring it to be passed explicitly. Also updated uses of the
	  [stdexten] context throughout. (closes issue #15858) Reported by:
	  pprindeville Patches: stdexten-context-update.txt uploaded by
	  lmadsen (license 10) Tested by: pprindeville

2009-11-03 18:22 +0000 [r227298]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Fixed a spelling error in the q850 reason
	  header option in the output of sip show settings.

2009-11-03 17:58 +0000 [r227277]  Richard Mudgett <rmudgett@digium.com>

	* /: Recorded merge of revisions 227275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
	  | 4 lines Make sure the outgoing flag is cleared if a new channel
	  fails to get created for outgoing calls. This is the relevant
	  portion of asterisk/trunk -r226648 ........

2009-11-03 17:56 +0000 [r227276]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_mgcp.c: Code guidelines fixes only

2009-11-03 17:12 +0000 [r227238]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: user.conf entries in SIP were not having
	  their peer type set. (closes issue #16120) Reported by: jsmith

2009-11-03 16:56 +0000 [r227237]  Olle Johansson <oej@edvina.net>

	* funcs/func_speex.c: Adding some clarifications to func_speex
	  doxygen docs. The functions needed doesn't exist in Speex 1.05
	  which is what a lot of distros use. 1.2 seems to have been in
	  beta status for years, and does include the sexy functions needed
	  for func_speex to work.

2009-11-03 15:37 +0000 [r227167]  Joshua Colp <jcolp@digium.com>

	* /: Merged revisions 227166 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
	  lines Fix a bug where an RPID header could be generated with a
	  blank username in the URI. (closes issue #15909) Reported by:
	  kobaz ........

2009-11-03 15:19 +0000 [r227162]  Leif Madsen <lmadsen@digium.com>

	* configs/extensions.conf.sample: Update extensions.conf.sample
	  file to fix incorrect extensions. (closes issue #15857) Reported
	  by: pprindeville Patches: stdexten.patch#2 uploaded by
	  pprindeville (license 347) Tested by: pprindeville

2009-11-03 11:11 +0000 [r227091]  Olle Johansson <oej@edvina.net>

	* Makefile, /, channels/chan_sip.c: Merged revisions 227088 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
	  lines Use proper response code when violating Contact ACL's.
	  https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
	  quick review. (EDVX-003) ........

2009-11-02 22:29 +0000 [r227049]  Tilghman Lesher <tlesher@digium.com>

	* configs/mgcp.conf.sample, include/asterisk/pktccops.h (added),
	  CHANGES, res/res_pktccops.c (added), channels/chan_mgcp.c,
	  configs/res_pktccops.conf.sample (added): Add PacketCable NCS 1.0
	  support for Docsis/Eurodocsis networks (closes issue #12950)
	  Reported by: alea-soluciones Patches:
	  ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones
	  (license 514) Tested by: alea-soluciones, adomjan, urtho,
	  nahuelgreco

2009-11-02 20:59 +0000 [r226973-226974]  David Brooks <dbrooks@digium.com>

	* channels/chan_sip.c: SIP channel name uniqueness SIP channel
	  names were supposed to be unique by way of a name suffix derived
	  from the pointer to the channel's private data. Uniqueness was
	  preserved on 32-bit systems, but not on 64-bit systems. This
	  patch, as suggested by kpfleming, replaces this suffix with a
	  simple incremented unsigned int. (closes issue #15152) Reported
	  by: palbrecht Review: https://reviewboard.asterisk.org/r/420/

	* /: SIP channel name uniqueness SIP channel names were supposed to
	  be unique by way of a name suffix derived from the pointer to the
	  channel's private data. Uniqueness was preserved on 32-bit
	  systems, but not on 64-bit systems. This patch, as suggested by
	  kpfleming, replaces this suffix with a simple incremented
	  unsigned int. (closes issue #15152) Reported by: palbrecht
	  Review: https://reviewboard.asterisk.org/r/420/

2009-11-02 20:43 +0000 [r226970]  Olle Johansson <oej@edvina.net>

	* main/http.c: Adding external reference for doxygen

2009-11-02 18:08 +0000 [r226890]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 226889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
	  11 lines Fix a bug where the recorded privacy introduction file
	  would not get removed if the caller hung up while the called
	  party had not yet answered. This was fixed by introducing an
	  argument to the 'n' option which, when enabled, removes the
	  introduction file under all scenarios. This was done to preserve
	  the behavior that has existed for quite some time. (closes issue
	  #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
	  jpeeler (license 325) ........

2009-11-02 17:34 +0000 [r226882]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, UPGRADE.txt,
	  channels/sig_pri.c: DAHDI ISDN channel names will not allow
	  device state to work. (Interim solution.) Since ISDN works like
	  SIP and not analog ports in regard to devices, the device state
	  based on the ISDN channel number could not work. This has not
	  been an issue until the advent of PTMP NT mode. Previously, ISDN
	  lines were used as trunks and did not have to keep track of
	  specific devices. As an interim solution until device states are
	  properly implemented, the channel name is being changed to the
	  following format to use the generic device state support:
	  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan
	  hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will
	  work with the following restrictions: * The number of
	  devices/phones cannot exceed the number of B channels. (i.e., BRI
	  has 2) * Each device/phone can only have one number. No shared
	  MSN's. * The phones/devices probably should not use
	  subaddressing.

2009-11-02 17:15 +0000 [r226812]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226811 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
	  | 8 lines Don't allow two separate instances of safe_asterisk
	  when restarting from the init script. (closes issue #14562)
	  Reported by: davidw Patches: Initially
	  20091022__issue14562.diff.txt uploaded by tilghman (license 14)
	  Modified to 20091030__Issue14562_diff.txt uploaded by davidw
	  (license 780) Tested by: davidw ........

2009-11-02 14:57 +0000 [r226687]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
	  adds support for a draft proposal for adding Q.850 reason headers
	  to sip messages. (closes issue #13385) Reported by: adomjan
	  Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded
	  by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch
	  uploaded by adomjan (license 487)
	  chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by
	  adomjan (license 487) sip-q850-hangupcause1.diff uploaded by
	  mnicholson (license 96) Tested by: adomjan

2009-10-30 23:26 +0000 [r226648]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_pri.c: Cleanup some flags on
	  DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI
	  channel hangup. (sig_pri split) * Make sure the outgoing flag is
	  cleared if a new channel fails to get created for outgoing calls.
	  * Remove some unused flags since sig_pri was split.

2009-10-30 04:08 +0000 [r226606]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxygen/architecture.h (added),
	  res/res_rtp_asterisk.c, res/res_rtp_multicast.c,
	  include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
	  main/asterisk.c: Add an "Asterisk Architecture Overview" section
	  to the doxygen documentation. This is a side project I've been
	  poking at this week. The intent is to discuss Asterisk
	  architecture in a top down fashion to help new developers
	  understand how Asterisk is put together. There is a ton of stuff
	  to write about, so this will just continue to evolve over time.

2009-10-29 18:13 +0000 [r226532]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
	  revisions 226531 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
	  lines Add an option to enabling passing music on hold start and
	  stop requests through instead of acting on them in chan_local.
	  (closes issue #14709) Reported by: dimas ........

2009-10-29 12:20 +0000 [r226490]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c: Doxygen documentation update

2009-10-28 20:50 +0000 [r226453]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* build_tools/get_documentation: remove empty awk pattern (//)
	  Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
	  Just remove that. No pattern at all always matches.

2009-10-28 20:11 +0000 [r226378-226384]  Leif Madsen <lmadsen@digium.com>

	* /, configs/sip.conf.sample: Merged revisions 226382 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28
	  Oct 2009) | 9 lines Update documentation in sip.conf.sample.
	  Update the documentation in sip.conf.sample in order to make it
	  more clear that directmedia/canreinvite do not cause Asterisk to
	  ignore reINVITEs. It is only used to stop Asterisk from
	  generating a reINVITE, but does not stop it from accepting them
	  if necessary. (closes issue #15644) Reported by: lmadsen ........

	* doc/tex/channelvariables.tex: Merged revisions 226377 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
	  | 7 lines Update CALLINGSUBADDR channel variable documentation.
	  (closes issue #15734) Reported by: alecdavis Patches:
	  channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
	  Tested by: alecdavis ........

2009-10-28 18:04 +0000 [r226305]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 226304 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009)
	  | 2 lines Fix documentation (pointed out by TheDavidFactor on
	  #-dev) ........

2009-10-28 08:47 +0000 [r226227-226270]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/upstart/asterisk.user.conf: Remove extra cleanup in case
	  we have more than one Asterisk. /var/run would be cleaned on
	  startup on most systems anyway.

	* contrib/upstart/asterisk.user.conf (added): another variation of
	  the upstart script

2009-10-27 21:03 +0000 [r226184]  Olle Johansson <oej@edvina.net>

	* Makefile: Adding compile time flags for Snow Leopard, Leopard and
	  some other animals

2009-10-27 20:22 +0000 [r226159]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /: Merged revisions 226138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
	  | 7 lines Manager output is not always NULL-terminated, so force
	  a NULL at the end of the filestream. (closes issue #15495)
	  Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
	  by tilghman (license 14) Tested by: pdf ........

2009-10-27 16:48 +0000 [r226099]  Terry Wilson <twilson@digium.com>

	* res/res_http_post.c: Don't prepend the URI prefix to the post
	  directory

2009-10-27 13:30 +0000 [r226060]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  support for receiving unsolicited MWI NOTIFY messages. This
	  change adds a configuration option to SIP peers,
	  unsolicited_mailbox, which configures a virtual mailbox to use
	  for received new/old MWI information. This virtual mailbox can
	  then be used by any device supporting MWI. (closes issue #13028)
	  Reported by: AsteriskRocks Patches:
	  bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj
	  (license 830)

2009-10-26 22:46 +0000 [r226018]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* /, configure, configure.ac: detect ARM Linux EABI OSARCH as
	  linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
	  if host_os is linux-gnueabi * When checking if we are Linux,
	  check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
	  the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
	  sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
	  tested for the value of 'linux-gnu' in one or two places in the
	  tree. This patch also fixes the check libcap to check for $OSARCH
	  rather than $host_os . See also:
	  http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
	  svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4

2009-10-26 22:04 +0000 [r225955-225956]  Kevin P. Fleming <kpfleming@digium.com>

	* main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt,
	  UPGRADE-1.6.txt, doc/lang/language-criteria.txt: Fix building in
	  REF_DEBUG mode.

	* main/astobj2.c: Correct broken logic from revision 225405. The
	  code committed in revision 225405 was broken; instead of removing
	  the unreference code, the logic used to decide when to do it
	  should have been reversed. This patch corrects the situation, and
	  makes reference counting work properly again.

2009-10-26 19:40 +0000 [r225912]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: ACL check not present for verifying SIP
	  INVITEs The ACL check in check_peer_ok was missing and has now
	  been restored. The missing check allowed for calls to be made on
	  prohibited networks where an ACL was defined in sip.conf and the
	  allowguest option was set to off. See the AST security advisory
	  below for more information. Merge code associated with
	  AST-2009-007. (closes issue #16091) Reported by: thom4fun

2009-10-26 16:07 +0000 [r225872]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Make conditionals create previous code
	  when libpri/ss7 are present.

2009-10-26 13:29 +0000 [r225767-225836]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c: span numbers in pri debug / error messages
	  Prefix PRI trace messages with the span number. This makes the
	  trace readable even when you have a multi-port device. (closes
	  issue #15054) Reported by: tzafrir Patches:
	  dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)

	* channels/chan_dahdi.c: Re-arange code a bit to build in dev-mode
	  without ss7 No change of functionality here. Just localized a
	  variable and indented code into blocks.

	* channels/chan_dahdi.c: Make chan_dahdi build even without PRI /
	  SS7 (Note: still some strange build warnings without SS7 in
	  dev-mode)

2009-10-24 14:40 +0000 [r225727]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Improve performance of pedantic mode dialog
	  searching in chan_sip. This patch changes chan_sip to use the new
	  astobj2 OBJ_MULTIPLE iterator support to make pedantic mode
	  dialog searching in find_call() not require a linear search of
	  all dialogs in the list of dialogs. This patch does *not* change
	  the dialog matching logic (more on that later), just improves the
	  searching performance.

2009-10-23 16:57 +0000 [r225692]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
	  channels/sig_pri.c: Add to chan_dahdi ISDN HOLD, Call deflection,
	  and keypad facility support. * Added handling of received
	  HOLD/RETRIEVE messages and the optional ability to transfer a
	  held call on disconnect similar to an analog phone. * Added
	  CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI
	  PTMP. Will reroute/deflect an outgoing call when receive the
	  message. Can use the DAHDISendCallreroutingFacility to send the
	  message for the supported switches. * Added ability to
	  send/receive keypad digits in the SETUP message. Send keypad
	  digits in SETUP message:
	  Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received
	  keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} *
	  Added support for BRI PTMP NT mode.

2009-10-23 16:40 +0000 [r225690]  Sean Bright <sean@malleable.com>

	* Makefile, agi/Makefile, agi/agi.xml (added): Optionally build and
	  install the sample AGIs in the agi/ directory.

2009-10-23 14:41 +0000 [r225650]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes an iterator memory leak and
	  uninitialized memory

2009-10-23 14:02 +0000 [r225582]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 225581 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
	  2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
	  every build. For some reason the menuselect.makeopts file was
	  listed as PHONY in the Makefile, resulting in 'make' needing to
	  rebuild it for every build. This then resulted in the embedded
	  module rules being rebuilt on every build, which can be slow and
	  is unnecessary. This patch fixes the problem by properly allowing
	  'make' to know when the menuselect.makeopts file needs to be
	  rebuilt (defining the proper dependencies). ........

2009-10-22 22:24 +0000 [r225483-225515]  Leif Madsen <lmadsen@digium.com>

	* README: Update README documentation. Update the README
	  documentation to correctly describe which CLI command you should
	  use when attempting to get help from the CLI. (closes issue
	  #16064) Reported by: thedavidfactor Patches: readme.patch
	  uploaded by thedavidfactor (license 903)

	* /, doc/valgrind.txt, contrib/valgrind.supp (added): Merged
	  revisions 225484 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
	  | 11 lines Clean valgrind output by suppressing false errors.
	  Update valgrind.txt documentation and add valgrind.supp file in
	  order to allow those who are creating valgrind output to have
	  less false errors in the logfile. (closes issue #16007) Reported
	  by: atis Patches: valgrind.txt.diff uploaded by atis (license
	  242) asterisk2.supp uploaded by atis (license 242) Tested by:
	  atis, amorsen ........

	* include/asterisk/doxyref.h,
	  include/asterisk/doxygen/asterisk-git-howto.h (added): Add
	  Asterisk Git HowTo documentation. Added documentation on how to
	  create a local git repository from SVN. This documentation was
	  added via doxygen. (closes issue #15814) Reported by: tzafrir
	  Patches: git-asterisk-howto uploaded by tzafrir (license 46)

2009-10-22 20:07 +0000 [r225446]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Search for the subaddress only within the
	  extension section of the dial string.
	  Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])

2009-10-22 19:55 +0000 [r225445]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, channels/chan_sip.c, apps/app_externalivr.c,
	  include/asterisk/tcptls.h: SIP TCP/TLS: move client connection
	  setup/write into tcp helper thread, various related
	  locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS
	  connection setup into the TCP helper thread: Connection setup
	  takes awhile and before this it was being done while holding the
	  monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread:
	  Through the use of a packet queue and an alert pipe, the TCP
	  helper thread can now be woken up to write data as well as read
	  data. 3.Locking error: sip_xmit returned an XMIT_ERROR without
	  giving up the tcptls_session lock. This lock has been completely
	  removed from sip_xmit and placed in the new sip_tcptls_write()
	  function. 4.Memory leak: When creating a tcptls_client the
	  tls_cfg was alloced but never freed unless the tcptls_session
	  failed to start. Now the session_args for a sip client are an ao2
	  object which frees the tls_cfg on destruction. 5.Pointer to stack
	  variable: During sip_prepare_socket the creation of a client's
	  ast_tcptls_session_args was done on the stack and stored as a
	  pointer in the newly created tcptls_session. Depending on the
	  events that followed, there was a slight possibility that pointer
	  could have been accessed after the stack returned. Given the new
	  changes, it is always accessed after the stack returns which is
	  why I found it. Notable code changes 1.I broke tcptls.c's
	  ast_tcptls_client_start() function into two functions. One for
	  creating and allocating the new tcptls_session, and a separate
	  one for starting and handling the new connection. This allowed me
	  to create the tcptls_session, launch the helper thread, and then
	  establish the connection within the helper thread. 2.Writes to a
	  tcptls_session are now done within the helper thread. This is
	  done by using an alert pipe to wake up the thread if new data
	  needs to be sent. The thread's sip_threadinfo object contains the
	  alert pipe as well as the packet queue. 3.Since the threadinfo
	  object contains the alert pipe, it must now be accessed outside
	  of the helper thread for every write (queuing of a packet). For
	  easy lookup, I moved the threadinfo objects from a linked list to
	  an ao2_container. (closes issue #13136) Reported by: pabelanger
	  Tested by: dvossel, whys (closes issue #15894) Reported by:
	  dvossel Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/380/

2009-10-22 19:33 +0000 [r225440]  Sean Bright <sean@malleable.com>

	* Makefile, utils/Makefile, utils/utils.xml (added),
	  doc/janitor-projects.txt: Add the programs in utils/ to
	  menuselect. Nothing in utils/ is now built by default except for
	  astcanary. Review: https://reviewboard.asterisk.org/r/353/

2009-10-22 19:10 +0000 [r225406]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Permit storage of voicemail secrets in a separate file, located
	  within the spool directory. (closes issue #14276) Reported by:
	  klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded
	  by klaus3000 (license 65) Tested by: jamesgolovich

2009-10-22 18:41 +0000 [r225405]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c: Fix a refcount error introduced by yesterday's
	  OBJ_MULTIPLE commit. When an object is being unlinked from its
	  container *and* being returned to the caller, we do not want to
	  decrement the reference count after unlinking it from the
	  container, as the reference that the container held is what we
	  are returning to the caller... and if it was the only remaining
	  reference to the object, that could result in the object being
	  destroyed.

2009-10-22 17:11 +0000 [r225360]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
	  Merged revisions 225105 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
	  | 4 lines Fix documentation for ast_softhangup() and correct the
	  misuse thereof. (closes issue #16103) Reported by: majorbloodnok
	  ........

2009-10-22 16:33 +0000 [r225357]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, funcs/func_connectedline.c,
	  include/asterisk/channel.h, CHANGES, channels/sig_pri.c,
	  funcs/func_callerid.c: Add support for calling and called
	  subaddress. Partial support for COLP subaddress. The Telecom
	  Specs in NZ suggests that SUB ADDRESS is always on, so doing
	  "desk to desk" between offices each with an asterisk box over the
	  ISDN should then be possible, without a whole load of DDI numbers
	  required. (closes issue #15604) Reported by: alecdavis Patches:
	  asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license
	  585) Some minor modificatons were made. Tested by: alecdavis,
	  rmudgett Review: https://reviewboard.asterisk.org/r/405/

2009-10-21 21:58 +0000 [r225307]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 225243 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21
	  Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames
	  with no destination call number It is possible for the PBX thread
	  to queue up signaling frames before a destination call number is
	  received. This can result in signaling frames being sent out with
	  no destination call number. Since recent versions of Asterisk
	  require accurate destination callnumbers for all Full Frames,
	  this can cause a VNAK loop to occur. To resolve this no signaling
	  frames are sent until a destination callnumber is received, and
	  destination call numbers are now only required for iax_pvt
	  matching when the frame is an ACK. Review:
	  https://reviewboard.asterisk.org/r/413/ ........

2009-10-21 21:15 +0000 [r225244-225245]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/tex/manager.tex, channels/chan_sip.c: Add 'mohsuggest'
	  configuration option to 'sip show peer' CLI command and
	  SIPShowPeer AMI action. (closes issue #15990) Reported by:
	  _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by
	  brent (license 388) Review:
	  https://reviewboard.asterisk.org/r/381/

	* main/channel.c, main/manager.c, apps/app_directed_pickup.c,
	  apps/app_softhangup.c, funcs/func_channel.c,
	  include/asterisk/astobj2.h, res/snmp/agent.c,
	  include/asterisk/channel.h, include/asterisk/lock.h,
	  apps/app_chanspy.c, main/astobj2.c, main/cli.c: Finish
	  implementaton of astobj2 OBJ_MULTIPLE, and convert
	  ast_channel_iterator to use it. This patch finishes the
	  implementation of OBJ_MULTIPLE in astobj2 (the case where
	  multiple results need to be returned; OBJ_NODATA mode already was
	  supported). In addition, it converts ast_channel_iterators (only
	  the targeted versions, not the ones that iterate over all
	  channels) to use this method. During this work, I removed the
	  'ao2_flags' arguments to the ast_channel_iterator constructor
	  functions; there were no uses of that argument yet, there is only
	  one possible flag to pass, and it made the iterators less
	  'opaque'. If at some point in the future someone really needs an
	  ast_channel_iterator that does not lock the container, we can
	  provide constructor(s) for that purpose. Review:
	  https://reviewboard.asterisk.org/r/379/

2009-10-21 16:46 +0000 [r225170-225172]  Russell Bryant <russell@digium.com>

	* /, main/translate.c: Merged revisions 225171 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009)
	  | 2 lines Revert 225169, as this doesn't account for the
	  possibility of a list of frames. ........

	* /, main/translate.c: Merged revisions 225169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009)
	  | 2 lines Isolate the frame returned from ast_translate().
	  ........

2009-10-21 15:42 +0000 [r225102]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c: Apparently, I don't need to specify the ".so"
	  suffix to get a match

2009-10-21 15:35 +0000 [r225089]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  support for specifying the IP address to use for media streams in
	  sip.conf This is the second commit for this and documents the
	  text stream using the configured IP address and fixes a bug in
	  the original patch where the UDPTL stream would also use the
	  different IP address. (closes issue #14729) Reported by: _brent_
	  Patches: media_address.patch uploaded by brent (license 388)

2009-10-21 15:21 +0000 [r225048]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, CHANGES: Turn on DENOISE filter for all
	  conference participants. (Fixes SWP-238)

2009-10-21 15:04 +0000 [r225034]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Revert
	  media_address commit, I'm going to roll a fix to the SDP
	  generation in the next version.

2009-10-21 14:39 +0000 [r225033]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, /, channels/chan_sip.c,
	  configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
	  225032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
	  | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
	  id removes '(', ' ', ')', non-trailing '.', and '-' from the
	  string. This means values such as 555.5555 and test-test result
	  in 555555 and testtest. There are instances, such as Skype
	  integration, where a specific value is passed via caller id that
	  must be preserved unmodified. This patch makes the shrinking of
	  caller id optional in chan_sip and chan_iax in order to support
	  such cases. By default this option is on to preserve previous
	  expected behavior. (closes issue #15940) Reported by: dimas
	  Patches: v2-15940.patch uploaded by dimas (license 88)
	  15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
	  Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/408/ ........

2009-10-21 13:34 +0000 [r225003]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  support for specifying the IP address to use for media streams in
	  sip.conf (closes issue #14729) Reported by: _brent_ Patches:
	  media_address.patch uploaded by brent (license 388)

2009-10-21 03:09 +0000 [r224932]  Russell Bryant <russell@digium.com>

	* main/frame.c, /, main/translate.c, include/asterisk/dsp.h,
	  codecs/codec_dahdi.c, include/asterisk/frame.h,
	  include/asterisk/translate.h, main/dsp.c: Merged revisions 224931
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
	  | 5 lines Isolate frames returned from a DSP instance or codec
	  translator. The reasoning for these changes are the same as what
	  I wrote in the commit message for rev 222878. ........

2009-10-21 02:43 +0000 [r224930]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress
	  friendly.

2009-10-20 22:09 +0000 [r224856]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224855
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
	  | 5 lines Pay attention to the return value of the manipulate
	  function. While this looks like an optimization, it prevents a
	  crash from occurring when used with certain audiohook callbacks
	  (diagnosed with SVN trunk, backported to 1.4 to keep the source
	  consistent across versions). ........

2009-10-20 17:47 +0000 [r224774]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 224773 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
	  lines Add support for relaying early media in the features
	  attended transfer option. (closes issue #14828) Reported by:
	  licedey ........

2009-10-20 12:44 +0000 [r224738]  Matthew Nicholson <mnicholson@digium.com>

	* CHANGES: Added information to CHANGES about the dynamic range
	  compression feature added to dahdi.

2009-10-19 23:47 +0000 [r224671]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_rtp_asterisk.c, /: Merged revisions 224670 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19
	  Oct 2009) | 7 lines Correct timestamp calculations when RTP
	  sample rates over 8kHz are used. While testing some endpoints
	  that support 16kHz and 32kHz sample rates, some log messages were
	  generated due to calc_rxstamp() computing timestamps in a way
	  that produced odd results, so this patch sanitizes the result of
	  the computations. ........

2009-10-19 22:02 +0000 [r224637]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
	  dynamic range compression support for analog channels. (closes
	  issue AST-29)

2009-10-19 19:49 +0000 [r224567]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 224565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
	  lines Do not attempt early media bridging (ie: direct RTP setup)
	  if options are enabled that should prevent it. (closes issue
	  #14763) Reported by: cupotka ........

2009-10-19 19:40 +0000 [r224562]  Kevin P. Fleming <kpfleming@digium.com>

	* formats/format_siren14.c: Remove useless debugging message.

2009-10-19 15:50 +0000 [r224527]  Tilghman Lesher <tlesher@digium.com>

	* doc/janitor-projects.txt: Remove a completed project and add
	  another

2009-10-19 14:32 +0000 [r224491]  Joshua Colp <jcolp@digium.com>

	* channels/sig_pri.h, channels/sig_pri.c: Add a callback to sig_pri
	  which is called when sig_pri is going to queue a control frame on
	  a channel.

2009-10-19 00:05 +0000 [r224446-224448]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Allow ODBC storage to be queried with
	  multiple mailboxes, and remove multiple goto's. This corrects an
	  issue reported on the -users list.

	* configs/res_odbc.conf.sample: Clarify that "forcecommit" is NOT
	  an alias for "autocommit", but instead controls the default
	  disposition of uncommitted transactions.

2009-10-17 16:39 +0000 [r224403]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, main/app.c: Remove unnecessary typedef

2009-10-17 02:01 +0000 [r224331-224335]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: fix typo, sorry

	* channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions
	  224330 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
	  | 13 lines Fix stale caller id data from being reported in AMI
	  NewChannel event The problem here is that chan_dahdi is designed
	  in such a way to set certain values in the dahdi_pvt only once.
	  One of those such values is the configured caller id data in
	  chan_dahdi.conf. For PRI, the configured caller id data could be
	  overwritten during a call. Instead of saving the data and
	  restoring, it was decided that for all non-analog channels it was
	  simply best to not set the configured caller id in the first
	  place and also clear it at the end of the call. (closes issue
	  #15883) Reported by: jsmith ........

2009-10-16 20:40 +0000 [r224261]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 224260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
	  | 18 lines Never released PRI channels when using Busy() or
	  Congestion() dialplan apps. When the Busy() or Congestion()
	  application is used towards ISDN (an ISDN progress is sent), the
	  responding ISDN Disconnect or Release may contain the ISDN cause
	  user busy or one of the congestion causes. In chan_dahdi.c these
	  causes will only set the needbusy or needcongestion flags and not
	  activate the softhangup procedure. Unfortunately only the latter
	  can interrupt the endless wait loop of Busy()/Congestion().
	  Result: PRI channels staying in state busy for the rest of
	  asterisk life or until the other end times out and forces the
	  call to clear. (issue #14292) Reported by: tomaso Patches:
	  disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
	  patch is unrelated to the issue.) ........

2009-10-15 22:33 +0000 [r224225]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, main/pbx.c, main/app.c: Create an API for
	  adding an optional time unit onto the ends of time periods. Two
	  examples of its use are included, and the usage could be expanded
	  in some cases into certain configuration options where time
	  periods are specified.

2009-10-15 15:57 +0000 [r224178]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_chanspy.c: Readd removed ability to allow listening to
	  one side of the call in app_chanspy (Option o) (closes issue
	  #15675) Reported by: john8675309 Patches:
	  issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested
	  by: jgutierrez on users list:
	  http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html

2009-10-15 14:37 +0000 [r224144]  Doug Bailey <dbailey@digium.com>

	* configs/chan_dahdi.conf.sample: chan_dahdi.conf.sample changes
	  for DTMF CID detect Explains new options for detecting DTMF CID
	  on fxo lines (issue #9096) Reported by: fleed Patches:
	  chan_dahid_sample_config.patch uploaded by sum (license 766)

2009-10-15 06:48 +0000 [r224074-224109]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_caldav.c: Properly handle PUT requests for
	  CALENDAR_WRITE()

	* res/res_calendar.c: Add missing 'getnum' field

2009-10-14 17:48 +0000 [r224035]  Jeff Peeler <jpeeler@digium.com>

	* configs/sip_notify.conf.sample, channels/chan_sip.c, CHANGES:
	  Allow for adding message body to the SIP NOTIFY message Ability
	  has been added to both manager command SIPnotify as well as
	  console command sip notify. Message body is stored in the
	  "Content" variable. An example is present in sip_notify.conf.
	  (closes issue #13926) Reported by: jthurman Patches:
	  sip-notify-svn189463.diff uploaded by gareth (license 208) Tested
	  by: gareth

2009-10-13 22:14 +0000 [r223992]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: use Calendar: instead of Calendar/ for
	  devstate

2009-10-13 17:11 +0000 [r223911-223912]  Richard Mudgett <rmudgett@digium.com>

	* include/asterisk/pbx.h: Fix some doxygen format problems and trim
	  trailing whitespace.

	* res/res_calendar.c: Fix compiler warning.

2009-10-13 01:58 +0000 [r223874-223875]  Terry Wilson <twilson@digium.com>

	* apps/app_originate.c: Revert inadvertant code commit to
	  app_originate

	* apps/app_originate.c, include/asterisk/calendar.h,
	  res/res_calendar.c: Fix handling of notification calls w/ the
	  dialing api

2009-10-12 23:48 +0000 [r223832]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_dial.c, /: Merged revisions 223804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
	  | 8 lines Ensure ringing continues for branched calls after
	  progress is received While waiting for an answer, don't send
	  progress for branched calls for which ringing was sent. (closes
	  issue #15028) Reported by: fnordian ........

2009-10-12 20:58 +0000 [r223756]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and
	  trunkmtu iax2 options SWP-151

2009-10-12 15:32 +0000 [r223652-223693]  Kevin P. Fleming <kpfleming@digium.com>

	* /: Recorded merge of revisions 223692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct
	  2009) | 13 lines Remove automatic switching from T.38 to voice
	  mode in chan_sip. chan_sip has some code to automatically switch
	  from T.38 mode to voice mode when a voice frame is written to the
	  channel while it is in T.38 mode; this was intended to handle the
	  situation when a FAX transmission has ended and the channel is
	  not yet hung up, but is causing problems at the beginning of FAX
	  sessions as well when there are still voice frames 'in flight' at
	  the time the T.38 negotiation completes. This patch removes the
	  automatic switchover. (issue #16025) Reported by: jamicque
	  ........

	* channels/chan_sip.c, apps/app_fax.c: Remove automatic switching
	  from T.38 to voice mode in chan_sip. chan_sip has some code to
	  automatically switch from T.38 mode to voice mode when a voice
	  frame is written to the channel while it is in T.38 mode; this
	  was intended to handle the situation when a FAX transmission has
	  ended and the channel is not yet hung up, but is causing problems
	  at the beginning of FAX sessions as well when there are still
	  voice frames 'in flight' at the time the T.38 negotiation
	  completes. This patch removes the automatic switchover, and
	  changes app_fax to explicitly switch off T.38 mode when the FAX
	  transmission process ends. (closes issue #16025) Reported by:
	  jamicque

2009-10-11 22:19 +0000 [r223617]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Check the proper page for the SENDRPID flag.
	  If a pending reinvite were sent, we might not properly send
	  connected party info since we were checking the wrong flag. This
	  was a rare occurrence, but could still happen nevertheless.

2009-10-11 18:35 +0000 [r223487-223553]  Russell Bryant <russell@digium.com>

	* /: Merged revisions 223550 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009)
	  | 2 lines Remove a duplicate ao2_iterator_destroy(). ........

	* main/autoservice.c, /: Merged revisions 223485-223486 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
	  | 6 lines Don't use data outside of its scope. The purpose of
	  this code was to have a hangup frame put on the list of deferred
	  frames. However, the code that read the hangup frame was outside
	  of the scope of where the hangup frame was declared. ........
	  r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
	  | 2 lines Remove some unnecessary code. ........

2009-10-10 20:02 +0000 [r223449]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Fix
	  handling of floating times and dates

2009-10-10 08:30 +0000 [r223413-223415]  Olle Johansson <oej@edvina.net>

	* configs/cdr_pgsql.conf.sample: Adding note about TLS usage

	* configs/res_ldap.conf.sample: Add an additional note on TLS
	  support

	* configs/res_ldap.conf.sample: Adding some information on TLS
	  support

2009-10-09 22:04 +0000 [r223370]  Terry Wilson <twilson@digium.com>

	* res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Properly
	  return "free" on confirmed events that are free CONFIRMED status
	  doesn't imply busy or free, that is handled with the TRANSP
	  field. Luckily, libical already sets the is_busy status on the
	  span for us.

2009-10-09 20:58 +0000 [r223330]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Initiate T.38 switchover when acting as called
	  party, regardless of FAX direction. SendFAX() and ReceiveFAX()
	  can be given options to indicate whether they should act as the
	  calling or called party; this mode should be used to decide
	  whether to initiate a switchover to T.38, not the direction that
	  the FAX transfer will take place. (closes issue #16039) Reported
	  by: jamicque

2009-10-09 18:34 +0000 [r223273]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, /: Merged revisions 223225 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
	  2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
	  when originating calls. (closes issue #15104) Reported by:
	  nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
	  (license 96) Tested by: nblasgen, mnicholson ........

2009-10-09 18:17 +0000 [r223211-223215]  Mark Michelson <mmichelson@digium.com>

	* /: Recorded merge of revisions 223213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct
	  2009) | 3 lines Fix potential memory leak in app_dial.c ........

	* apps/app_dial.c: Fix potential memory leaks. ABE-1998

2009-10-09 17:53 +0000 [r223206]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 223205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
	  | 10 lines fixes sip registration using authuser in user.conf
	  (closes issue #14954) Reported by: tornblad Tested by:
	  mmichelson, tornblad, dvossel ........

2009-10-09 17:14 +0000 [r223136]  Matthew Nicholson <mnicholson@digium.com>

	* cdr/cdr_sqlite3_custom.c: Don't close the sqlite database when
	  reloading. Only close the database when unloading. (closes issue
	  #15953) Reported by: frawd Patches: sqlite3_rev220097.diff
	  uploaded by frawd (license 610) Tested by: frawd

2009-10-09 16:54 +0000 [r223088-223132]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
	  (closes issue #15949) Reported by: ebroad Patches:
	  authparsefix.patch uploaded by ebroad (license 878)
	  15949_trunk.diff uploaded by dvossel (license 671) Tested by:
	  ebroad

	* channels/chan_sip.c: p->peerauth is always empty in
	  transmit_register() When using callbackextension or specifing the
	  peer name in a registration string, the peer's specific auth
	  settings set by the "auth=" strings within the peer definition
	  are not used by the registration. Thanks to ebroad for reporting
	  the issue and providing the patch. (closes issue #15955) Reported
	  by: ebroad Patches: regauthfix.patch uploaded by ebroad (license
	  878)

2009-10-09 15:00 +0000 [r223016-223053]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: Don't add Attendees during copy, replace them

	* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  res/res_calendar_caldav.c, include/asterisk/calendar.h,
	  res/res_calendar.c: Remove global variable that makes dlopen
	  unhappy This isn't the best way to do this, but it is the
	  easiest. There are some limitations that are going to need to be
	  addressed at some point with reloads and when I (or someone else)
	  work on that, then the API can be updated to handle passing the
	  private config data that the calendar tech modules need in a
	  better way as well.

2009-10-08 22:57 +0000 [r222947-223015]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixed comment line for do_magic_pickup

	* channels/chan_sip.c: Deadlock between ast_cel_report_event and
	  ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
	  channel while only the pvt lock is held. Since pbx_exec calls
	  ast_cel_report_event which attempts to lock the channel, invalid
	  locking order occurs. Channels should be locked before pvt's.
	  (closes issue #15512) Reported by: lmsteffan Patches:
	  ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)

	* channels/chan_sip.c: makes externtcpport and externtlsport static
	  variables externtcpport and externtlsport need to be declared as
	  static variables. Thanks to russell for finding and pointing this
	  out.

2009-10-08 19:52 +0000 [r222880]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, main/frame.c, /, main/file.c,
	  include/asterisk/frame.h: Merged revisions 222878 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08
	  Oct 2009) | 44 lines Make filestream frame handling safer by
	  isolating frames before returning them. This patch is related to
	  a number of issues on the bug tracker that show crashes related
	  to freeing frames that came from a filestream. A number of fixes
	  have been made over time while trying to figure out these
	  problems, but there re still people seeing the crash. (Note that
	  some of these bug reports include information about other
	  problems. I am specifically addressing the filestream frame crash
	  here.) I'm still not clear on what the exact problem is. However,
	  what is _very_ clear is that we have seen quite a few problems
	  over time related to unexpected behavior when we try to use
	  embedded frames as an optimization. In some cases, this
	  optimization doesn't really provide much due to improvements made
	  in other areas. In this case, the patch modifies filestream
	  handling such that the embedded frame will not be returned.
	  ast_frisolate() is used to ensure that we end up with a
	  completely mallocd frame. In reality, though, we will not
	  actually have to malloc every time. For filestreams, the frame
	  will almost always be allocated and freed in the same thread.
	  That means that the thread local frame cache will be used. So,
	  going this route doesn't hurt. With this patch in place, some
	  people have reported success in not seeing the crash anymore.
	  (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
	  Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
	  (license 2) Tested by: aragon, russell (closes issue #15817)
	  Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
	  Reported by: marhbere Review:
	  https://reviewboard.asterisk.org/r/386/ ........

2009-10-08 19:35 +0000 [r222873]  David Vossel <dvossel@digium.com>

	* include/asterisk/netsock.h, main/netsock.c: fixes an
	  ast_netsock_list memory leak. ABE-1998 Review:
	  https://reviewboard.asterisk.org/r/395/

2009-10-08 16:44 +0000 [r222799]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/misdn_config.c: Merged revisions 222797 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08
	  Oct 2009) | 12 lines Fix memory leak if chan_misdn config
	  parameter is repeated. Memory leak when the same config option is
	  set more than once in an misdn.conf section. Why must this be
	  considered? Templates! Defining a template with default port
	  options and later adding to or overriding some of them. Patches:
	  memleak-misdn.patch JIRA ABE-1998 ........

2009-10-07 22:58 +0000 [r222761]  David Vossel <dvossel@digium.com>

	* main/channel.c, main/pbx.c, channels/chan_misdn.c,
	  channels/chan_sip.c, main/features.c, include/asterisk/channel.h:
	  Deadlock in channel masquerade handling Channels are stored in an
	  ao2_container. When accessing an item within an ao2_container the
	  proper locking order is to first lock the container, and then the
	  items within it. In ast_do_masquerade both the clone and original
	  channel must be locked for the entire duration of the function.
	  The problem with this is that it attemptes to unlink and link
	  these channels back into the ao2_container when one of the
	  channel's name changes. This is invalid locking order as the
	  process of unlinking and linking will lock the ao2_container
	  while the channels are locked!!! Now, both the channels in
	  do_masquerade are unlinked from the ao2_container and then locked
	  for the entire function. At the end of the function both channels
	  are unlocked and linked back into the container with their new
	  names as hash values. This new method of requiring all channels
	  and tech pvts to be unlocked before ast_do_masquerade() or
	  ast_change_name() required several changes throughout the code
	  base. (closes issue #15911) Reported by: russell Patches:
	  masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested
	  by: dvossel, atis (closes issue #15618) Reported by: lmsteffan
	  Patches: deadlock_local_attended_transfers_trunk.diff uploaded by
	  dvossel (license 671) Tested by: lmsteffan, dvossel Review:
	  https://reviewboard.asterisk.org/r/387/

2009-10-07 21:56 +0000 [r222692]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 222691 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07
	  Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak
	  misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
	  does not occur. The translated frame "f2" when passing through
	  ast_dsp_process() is not freed whenever it is not used further in
	  process_ast_dsp(). Then in the end it is never ever freed.
	  Patches: translate.patch JIRA ABE-1993 ........

2009-10-07 20:08 +0000 [r222652]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Change ringt (ring timeout) styles to be
	  consistent across chan_dahdi. (closes issue #15684) Reported by:
	  alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by
	  alecdavis (license 585) Tested by: alecdavis

2009-10-07 18:57 +0000 [r222614-222615]  Olle Johansson <oej@edvina.net>

	* res/res_config_ldap.c: Formatting, moving error messages to
	  ERROR, removing references to unexisting debug output. No
	  functionality changes.

	* cel/cel_pgsql.c, res/res_config_pgsql.c, cdr/cdr_pgsql.c: Use
	  extref for doxygen references to external libraries (in this case
	  PostgreSQL)

2009-10-07 18:04 +0000 [r222548]  Jason Parker <jparker@digium.com>

	* configs/queues.conf.sample: Remove 'keepstats' queue option from
	  sample config, as it's no longer used.
	  https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
	  Reported by: kshumard

2009-10-07 17:44 +0000 [r222543]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 222542 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
	  | 8 lines crash on transfer handle_invite_replaces() attempts to
	  uplock a pvt's owner channel without first verifing that it
	  exists. (issue #16027) ........

2009-10-06 23:56 +0000 [r222463]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 222462 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06
	  Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two
	  cases in trunk) (closes issue #15683) Reported by: alecdavis
	  ........

2009-10-06 22:49 +0000 [r222398-222399]  David Vossel <dvossel@digium.com>

	* CHANGES: Updates CHANGES to reflect the new externtcpport and
	  externtlsport sip options

	* channels/chan_sip.c, configs/sip.conf.sample: contact header port
	  ignored transport when using externip This patch adds support for
	  TCP/TLS in the Contact header when using NAT, specifically
	  externip or externhost. The original issue was that Asterisk sent
	  5060 as the port in the contact header whether TLS was used or
	  not. Additionally, this patch adds 2 config options to sip.conf,
	  specifically externtcpport and externtlsport. This allows a user
	  to specify different external ports for TCP and TLS other than
	  those used internally, this is especially useful in in a PAT/port
	  redirection setup. Thanks to ebroad for reporting the issue and
	  providing the patch! (closes issue #15880) Reported by: ebroad
	  Patches: portmap.patch uploaded by ebroad (license 878)
	  externtXXport_v2.patch uploaded by ebroad (license 878) Tested
	  by: ebroad Review: https://reviewboard.asterisk.org/r/392/

2009-10-06 20:35 +0000 [r222351]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix 222298 (crash during destruction of
	  second channel when variable set with setvar). I mistakenly
	  reasoned that setvar would be used on all channels. Since it can
	  be set per channel, give each dahdi channel a copy of the
	  variable. (related to #15899)

2009-10-06 19:31 +0000 [r222309]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c, cdr/cdr_pgsql.c: Change schema query to
	  involve the use of an optional schema parameter. This change is
	  done in such a way as to allow the driver to continue to function
	  with older databases which don't have these features. (closes
	  issue #16000) Reported by: jamicque Patches:
	  20091002__issue16000.diff.txt uploaded by tilghman (license 14)
	  20091002__issue16000__1.6.1.diff.txt uploaded by tilghman
	  (license 14) Tested by: jamicque

2009-10-06 19:24 +0000 [r222298]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix crash during destruction of second
	  channel when variable set with setvar. The setvar line in
	  chan_dahdi.conf is shared among all the channels, so make sure to
	  only free the resources only when the last channel is destroyed.
	  (closes issue #15899) Reported by: tzafrir

2009-10-06 19:17 +0000 [r222273]  Tilghman Lesher <tlesher@digium.com>

	* res/ael/pval.c: When we call a gosub routine, the variables
	  should be scoped to avoid contaminating the caller. This affected
	  the ~~EXTEN~~ hack, where a subroutine might have changed the
	  value before it was used in the caller. Patch by myself, tested
	  by ebroad on #asterisk

2009-10-06 16:17 +0000 [r222237]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c: Make sure digit events are not reported as
	  "ERROR" dahdievent_to_analogevent used a simple switch statement
	  to convert DAHDI event numbers to "ANALOG_*" event numbers.
	  However "digit" events (DAHDI_EVENT_PULSEDIGIT,
	  DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP) are accompannied by the
	  digit in the low word of the event number. This fix makes
	  dahdievent_to_analogevent() return the event number as-is for
	  such an event. This is also required to fix #15924 (in addition
	  to r222108).

2009-10-06 01:24 +0000 [r222110-222176]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c, funcs/func_dialgroup.c,
	  include/asterisk/astobj2.h, res/res_phoneprov.c,
	  channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
	  channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
	  res/res_calendar.c, res/res_clialiases.c: Recorded merge of
	  revisions 222152 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct
	  2009) | 20 lines Fix ao2_iterator API to hold references to
	  containers being iterated. See Mantis issue for details of what
	  prompted this change. Additional notes: This patch changes the
	  ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
	  instead of a macro, with a name that fits our naming policy;
	  also, it is now necessary to call ao2_iterator_destroy() on any
	  iterator that has been created. Currently this only releases the
	  reference to the container being iterated, but in the future this
	  could also release other resources used by the iterator, if the
	  iterator implementation changes to use additional resources.
	  (closes issue #15987) Reported by: kpfleming Review:
	  https://reviewboard.asterisk.org/r/383/ ........

	* main/udptl.c, channels/chan_sip.c, configs/udptl.conf.sample,
	  UPGRADE.txt, configs/sip.conf.sample: Allow non-compliant T.38
	  endpoints to be supportable via configuration option. Many T.38
	  endpoints incorrectly send the maximum IFP frame size they can
	  accept as the T38FaxMaxDatagram value in their SDP, when in fact
	  this value is supposed to be the maximum UDPTL payload size
	  (datagram size) they can accept. If the value they supply is
	  small enough (a commonly supplied value is '72'), T.38 UDPTL
	  transmissions will likely fail completely because the UDPTL
	  packets will not have enough room for a primary IFP frame and the
	  redundancy used for error correction. If this occurs, the
	  Asterisk UDPTL stack will emit log messages warning that data
	  loss may occur, and that the value may need to be overridden.
	  This patch extends the 't38pt_udptl' configuration option in
	  sip.conf to allow the administrator to override the value
	  supplied by the remote endpoint and supply a value that allows
	  T.38 FAX transmissions to be successful with that endpoint. In
	  addition, in any SIP call where the override takes effect, a
	  debug message will be printed to that effect. This patch also
	  removes the T38FaxMaxDatagram configuration option from
	  udptl.conf.sample, since it has not actually had any effect for a
	  number of releases. In addition, this patch cleans up the T.38
	  documentation in sip.conf.sample (which incorrectly documented
	  that T.38 support was passthrough only). (issue #15586) Reported
	  by: globalnetinc

2009-10-05 19:20 +0000 [r222108]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Add a few missing events to
	  analog_handle_event. The reported bug was actually only for
	  pulsedigit, dtmfup, and dtmfdown handling. Also added recognition
	  for fax events (just some verbose output) and fixed handling for
	  the ec_disabled_event. In order to make comparing the analog
	  version of events to the DAHDI events easier, the ordering has
	  been changed to follow that of the DAHDI events. (closes issue
	  #15924) Reported by: tzafrir

2009-10-02 17:34 +0000 [r222030]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 222026 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
	  Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
	  memcpy. ........

2009-10-02 16:59 +0000 [r221920-221971]  Tilghman Lesher <tlesher@digium.com>

	* /, main/astobj2.c: Merged revisions 221970 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
	  | 2 lines Ensure the result of the hash function is positive.
	  Negative array offsets suck. ........

	* main/logger.c: Initialize a variable that we check immediately
	  upon startup. (closes issue #15973) Reported by: atis

2009-10-02 01:49 +0000 [r221844-221881]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c: Whitespace change.

	* channels/misdn/isdn_lib.c: Whitespace change.

	* channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
	  Merged revisions 221769 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
	  | 26 lines Occasionally losing use of B channels in chan_misdn. I
	  have not been able to reproduce the problem of losing channels.
	  However, I have seen in the code a reentrancy problem that might
	  give these symptoms. The reentrancy patch does several things: 1)
	  Guards B channel and B channel structure allocation. 2) Makes the
	  B channel structure find routines more precise in locating
	  records. 3) Never leave a B channel allocated if we received
	  cause 44. The last item may cause temporary outgoing call
	  problems, but they should clear when the line becomes idle.
	  (closes issue #15490) Reported by: slutec18 Patches:
	  issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
	  (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
	  Reported by: FabienToune Patches:
	  issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
	  (license 664) Tested by: FabienToune, rmudgett, slutec18 ........

2009-10-02 00:08 +0000 [r221777-221781]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: One more off-by-one in trunk

	* main/rtp_engine.c, /, main/say.c, main/asterisk.c: Merged
	  revisions 221776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
	  | 2 lines Fix a bunch of off-by-one errors ........

2009-10-01 20:18 +0000 [r221709]  Richard Mudgett <rmudgett@digium.com>

	* UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from
	  CHANGES to UPGRADE.txt.

2009-10-01 20:09 +0000 [r221705]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Revision 220906 (a merge from 1.4) was not
	  merged correctly, causing a problem with non-dynamic peers.

2009-10-01 19:48 +0000 [r221701]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, CHANGES: Prevent
	  deadlock if chan_dahdi attempts to change PRI channel names. The
	  PRI channels can no longer change the channel name if a different
	  B channel is selected during call negotiation. To prevent using
	  the channel name to infer what B channel a call is using and to
	  avoid name collisions, the channel name format is changed. The
	  new channel naming for PRI channels is:
	  DAHDI/ISDN-<span>-<sequence-number>

2009-10-01 19:33 +0000 [r221697]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: outbound tls connections were not defaulting
	  to port 5061 (closes issue #15854) Reported by: dvossel Patches:
	  sip_port_config_trunk.diff uploaded by dvossel (license 671)
	  Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/357/

2009-10-01 16:27 +0000 [r221592-221627]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: Sync up UPGRADE.txt with the 1.6.2 version.

	* main/udptl.c, configs/udptl.conf.sample: Remove ability to
	  control T.38 FAX error correction from udptl.conf. chan_sip has
	  had the ability to control T.38 FAX error correction mode on a
	  per-peer (or global) basis for a couple of releases now, which is
	  where it should have been all along. This patch removes the
	  ability to configure it in udptl.conf, but issues a warning if
	  the user tries to do, telling them to look at sip.conf.sample for
	  how to configure it now. For any SIP peers that are T.38 enabled
	  in sip.conf, there is already a default for FEC error correction
	  even if the user does not specify any mode, so this change will
	  not turn off error correction by default, it will have the same
	  default value that has been in the udptl.conf sample file.

2009-10-01 15:26 +0000 [r221589]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 221588 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
	  2009) | 2 lines Use unsigned ints for portinuri flags. ........

2009-10-01 07:00 +0000 [r221554]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Simplify code for porturi, use TRUE/FALSE
	  constructs when it's just TRUE or FALSE.

2009-09-30 23:04 +0000 [r221484]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Cleaned up merge from r221432

2009-09-30 21:15 +0000 [r221436]  Matthias Nick <mnick@digium.com>

	* apps/app_queue.c: Prevents from division by zero

2009-09-30 20:40 +0000 [r221432]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  221360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
	  2009) | 10 lines Fix SRV lookup and Request-URI generation in
	  chan_sip. This patch adds a new field "portinuri" to the sip
	  dialog struct and the sip peer struct. That field is used during
	  RURI generation to determine if the port should be included in
	  the RURI. It is also used in some places to determine if an SRV
	  lookup should occur. (closes issue #14418) Reported by: klaus3000
	  Tested by: klaus3000, mnicholson Review:
	  https://reviewboard.asterisk.org/r/369/ ........

2009-09-30 19:42 +0000 [r221368]  Matthias Nick <mnick@digium.com>

	* configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
	  revisions 221153,221157,221303 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
	  2 lines check bounds - prevents for buffer overflow ........
	  r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
	  8 lines added a new dialplan function 'CSV_QUOTE' and changed the
	  cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
	  Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
	  mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
	  30 Sep 2009) | 2 lines changed the prototype definition of
	  csv_quote ........

2009-09-30 18:47 +0000 [r221266-221300]  Terry Wilson <twilson@digium.com>

	* res/res_rtp_asterisk.c: Remove spurious debug

	* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
	  include/asterisk/rtp_engine.h: Use rtp properties instead of
	  adding a callback Thanks, Josh.

	* res/res_rtp_asterisk.c, main/rtp_engine.c, /,
	  channels/chan_sip.c, configs/sip.conf.sample,
	  include/asterisk/rtp_engine.h: Merged revisions 221086 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
	  | 25 lines Change the SSRC by default when our media stream
	  changes Be default, change SSRC when doing an audio stream
	  changes Asterisk doesn't honor marker bit when reinvited to
	  already-bridged RTP streams,resulting in far-end stack discarding
	  packets with "old" timestamps that areactually part of a new
	  stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
	  a reinvite, unless the 'constantssrc' is set to true in sip.conf.
	  The original issue reported to Digium support detailed the
	  following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
	  Application Server Call comes in fromITSP, Asterisk dials the app
	  server which sends a re-invite back toAsterisk--not to negotiate
	  to send media directly to the ITSP, but to indicatethat it's
	  changing the stream it's sending to Asterisk. The app
	  servergenerates a new SSRC, sequence numbers, timestamps, and
	  sets the marker bit on the new stream. Asterisk passes through
	  the teimstamp of the new stream, butdoes not reset the SSRC,
	  sequence numbers, or set the marker bit. When the timestamp on
	  the new stream is older than the timestamp on the originalstream,
	  the ITSP (which doesn't know there has been any change) discards
	  the newframes because it thinks they are too old. This patch
	  addresses this by changing the SSRC on a stream update unless
	  constantssrc=true is set in sip.conf. Review:
	  https://reviewboard.asterisk.org/r/374/ ........

2009-09-30 16:56 +0000 [r221201]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 221200 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
	  | 7 lines Avoid a potential NULL dereference. (closes issue
	  #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
	  uploaded by tilghman (license 14) Tested by: kobaz ........

2009-09-30 15:11 +0000 [r221085-221090]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: Modify VoiceMailMain()'s a() argument to
	  allow mailboxes to be specified by name. (closes issue #14740)
	  Reported by: pj Patches: issue14740_09022009.diff uploaded by
	  seanbright (license 71) Tested by: seanbright, lmadsen

	* apps/app_voicemail.c: Clarify documentation for VoiceMailMain()'s
	  a() option. We require box numbers, not names as the
	  documentation implies. (issue #14740) Reported by: pj Patches:
	  __20090729-app_voicemail-documentation.patch uploaded by lmadsen
	  (license 10) Tested by: seanbright, lmadsen

2009-09-30 04:32 +0000 [r221044]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_lock.c: Allow locks to be inherited through a
	  masquerade without causing starvation. (closes issue #14859)
	  Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
	  by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
	  uploaded by tilghman (license 14) Tested by: atis, tilghman

2009-09-29 21:28 +0000 [r220920-220995]  Mark Michelson <mmichelson@digium.com>

	* main/cel.c: Fix channel reference leak. ast_cel_report_event
	  would geet a reference to the bridged channel. However, certain
	  return paths, such as if CEL was not enabled, would result in a
	  reference leak. All return paths now properly unref the channel.
	  (closes issue #15991) Reported by: mmichelson

	* main/rtp_engine.c: Get rid of annoying and cryptic debug
	  messages.

2009-09-29 19:57 +0000 [r220906]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 220873 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
	  | 9 lines Reduce CPU usage related to building a peer merely for
	  devicestates. This fixes a 100% CPU problem in the SIP driver,
	  found by profiling the driver while the problem was occurring.
	  (closes issue #14309) Reported by: pkempgen Patches:
	  20090924__issue14309.diff.txt uploaded by tilghman (license 14)
	  Tested by: pkempgen, vrban ........

2009-09-29 19:49 +0000 [r220904]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_confbridge.c: Fix options 'm' and 's'. They were swapped
	  in the code. Also document the fact that app_confbridge does not
	  automatically answer the channel. (closes issue #15964) Reported
	  by: shrift

2009-09-29 16:58 +0000 [r220833]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Make deletion of temporary greetings work
	  properly with IMAP_STORAGE When imapgreetings was set to yes, the
	  message was being deleted but wasn't actually being expunged.
	  When imapgreetings was set to no, the file based message was not
	  being deleted at all. All good now! (closes issue #14949)
	  Reported by: noahisaac Patches: vm_tempgreeting_removal.patch
	  uploaded by noahisaac (license 748), modified by me

2009-09-28 21:02 +0000 [r220792]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor
	  changes.

2009-09-28 19:11 +0000 [r220721]  Sean Bright <sean@malleable.com>

	* /, Makefile.rules: Merged revisions 220717 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
	  2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
	  explicitly pass -O0 to the compiler so we override any default
	  optimization levels for a particular install. ........

2009-09-28 19:10 +0000 [r220718]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Fix building of registration entry in
	  build_peer when using callbackextension Check for remotesecret
	  option was unintentionally always true, which therefore caused
	  the secret option to never be used. Thanks to dvossel for
	  pointing out the exact fix. (closes issue #15943) Reported by:
	  tpsast

2009-09-28 15:27 +0000 [r220672]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/sig_pri.c: Locking issues dealing
	  with service_lock. * Removed unneeded and uninitialized
	  service_lock. * Fixed potential locking imbalance in
	  pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in
	  pri_dchannel():PRI_EVENT_RESTART.

2009-09-27 20:40 +0000 [r220629]  Michiel van Baak <michiel@vanbaak.info>

	* funcs/func_callerid.c: add name argument for the CALLERID
	  dialplan function to the xml documentation. Pointed out to me on
	  IRC by snuff-home. Thanks

2009-09-26 15:10 +0000 [r220586]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not
	  present.

2009-09-25 19:56 +0000 [r220543]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Reduce indentation in sig_pri_available().

2009-09-25 14:50 +0000 [r220494-220496]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: Eliminate unnecessary include of version.h in
	  manager.c. Including version.h here causes this file to get
	  recompiled after every commit or update, which is not needed.

	* main/channel.c: Correct sense of logic test committed in revision
	  220494.

	* main/channel.c: Don't use hash-based lookups for
	  ast_channel_get_by_name_prefix(). ast_channel_get_full() tries to
	  use OBJ_POINTER to optimize name-based channel lookups, but this
	  will not work properly when the channel's full name was not
	  supplied; for name-prefix searches, there is no value in doing a
	  hash-based lookup, and in fact doing so could result in many
	  channels being skipped.

2009-09-25 10:54 +0000 [r220457]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_jingle.c, configs/jabber.conf.sample,
	  include/asterisk/jabber.h, channels/chan_gtalk.c, CHANGES,
	  doc/jabber.txt, res/res_jabber.c: Add JABBER_RECEIVE as a
	  dialplan function, implement SendText in Jingle channels
	  JABBER_RECEIVE (along with JabberSend) makes Asterisk interact
	  with users over XMPP to process calls. SendText can be used
	  instead of JabberSend in the context of XMPP based voice channels
	  (chan_gtalk and chan_jingle). (closes issue #12569) Reported by:
	  eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
	  Review: https://reviewboard.asterisk.org/r/88/

2009-09-24 22:53 +0000 [r220417]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE.txt, main/asterisk.c: Change the default behavior of Set,
	  AGI, and pbx_realtime to 1.6 behavior by default (starting in
	  1.6.3).

2009-09-24 20:37 +0000 [r220365]  David Vossel <dvossel@digium.com>

	* main/tcptls.c: fixes tcptls_session memory leak caused by ref
	  count error (closes issue #15939) Reported by: dvossel Review:
	  https://reviewboard.asterisk.org/r/375/

2009-09-24 20:29 +0000 [r220344]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_dial.c, main/features.c, include/asterisk/features.h:
	  Add bridge related dial flags to the bridge app Most of the
	  functionality here is gained simply by setting the feature flag
	  on the bridge config. However, the dial limit functionality has
	  been moved from app_dial to the features code and has been made
	  public so both app_dial and the bridge app can use it. (closes
	  issue #13165) Reported by: tim_ringenbach Patches:
	  app_bridge_options_r138998.diff uploaded by tim ringenbach
	  (license 540), modified by me

2009-09-24 19:57 +0000 [r220295]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Documentation in the commit messages is
	  soon forgotten, please add it to the docs in the product.

2009-09-24 19:41 +0000 [r220289]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /, apps/app_disa.c, apps/app_playback.c: Merged
	  revisions 220288 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
	  | 6 lines Implicitly sending a progress signal breaks some
	  applications. Call Progress() in your dialplan if you explicitly
	  want progress to be sent. (Reverts change 216430, closes issue
	  #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
	  list
	  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
	  ........

2009-09-24 18:19 +0000 [r220217]  Sean Bright <sean@malleable.com>

	* Makefile, /: Merged revisions 220213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
	  2009) | 1 line Resolve parallel build warnings. Reported by Klaus
	  Darilion on the asterisk-dev mailing list. ........

2009-09-24 16:33 +0000 [r220174]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Ensure the numeric portion of the
	  P-Asserted-Identity header is properly escaped.

2009-09-24 14:44 +0000 [r220100]  Sean Bright <sean@malleable.com>

	* Makefile, build_tools/mkpkgconfig, /: Merged revisions 220099 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep
	  2009) | 2 lines Remove the remaining bashisms in the
	  Makefile/mkpkgconfig ........

2009-09-24 08:36 +0000 [r220028]  Michiel van Baak <michiel@vanbaak.info>

	* build_tools/mkpkgconfig, /: Merged revisions 220027 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24
	  Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use
	  /bin/sh This fixes building on all systems that don't have bash
	  at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
	  #asterisk-dev ........

2009-09-24 07:39 +0000 [r219951-219987]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_directory.c: Fix two possible crashes, one only in 1.6.1
	  and one in 1.6.1 forward. (closes issue #15739) Reported by:
	  DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by
	  tilghman (license 14) 20090922__issue15739.diff.txt uploaded by
	  tilghman (license 14) Tested by: DLNoah, jeffg

	* configs/mgcp.conf.sample, CHANGES, channels/chan_mgcp.c: Add
	  support for 'setvar=' for MGCP device lines, like other channel
	  drivers provide. (closes issue #14818) Reported by:
	  alea-soluciones Patches:
	  chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea
	  (license 514)

	* doc/lang/language-criteria.txt: Update fax number to the legal
	  fax, not the generic fax. (closes issue #15946) Reported by:
	  jtodd Patches: leif-is-a-wuss.txt uploaded by jtodd (license 870)
	  Tested by: jparker, tilghman, jtodd, russellb, mmichelson,
	  seanbright, kpfleming, and the rest of the usual suspects

2009-09-23 17:46 +0000 [r219895]  Leif Madsen <lmadsen@digium.com>

	* include/asterisk/doxyref.h,
	  include/asterisk/doxygen/mantisworkflow.h (added): Add Mantis
	  work flow documention. This commit adds the doxygen changes that
	  I've made to describe the Mantis work flow documentation for the
	  open source issue tracker. This should make it easier to
	  determine the flow of issues through the issue tracker, and what
	  those statuses mean. (closes issue #15902) Reported by: lmadsen
	  Patches: mantisworkflow.h uploaded by lmadsen (license 10)
	  Review: https://reviewboard.asterisk.org/r/367/

2009-09-22 21:43 +0000 [r219818]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 219816 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22
	  Sep 2009) | 10 lines When IMAP variables were changed during a
	  reload, Voicemail did not use the new values. This change
	  introduces a configuration version variable, which ensures that
	  connections with the old values are not reused but are allowed to
	  expire normally. (closes issue #15934) Reported by:
	  viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
	  tilghman (license 14) Tested by: viniciusfontes ........

2009-09-21 16:59 +0000 [r219721]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 219720 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
	  Sep 2009) | 3 lines Reverting merge 219520. This change was not
	  necessary. ........

2009-09-20 17:55 +0000 [r219654]  Tilghman Lesher <tlesher@digium.com>

	* /, main/file.c: Merged revisions 219653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
	  | 8 lines Really stop the stream, when ast_closestream() is
	  called. (closes issue #15129) Reported by: bmh Patches:
	  20090918__issue15129.diff.txt uploaded by tilghman (license 14)
	  Review: https://reviewboard.asterisk.org/r/372/ ........

2009-09-19 02:59 +0000 [r219587]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 219586 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18
	  Sep 2009) | 6 lines Make sure the iax_pvt exists before
	  dereferencing it. This fixes the latest crash posted on issue
	  15609. (issue #15609) ........

2009-09-18 23:20 +0000 [r219451-219520]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 219519 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18
	  Sep 2009) | 9 lines iax2 frame double free The iax frame's
	  retrans sched id was written over right before iax2_frame_free
	  was called. In iax2_frame_free that retrans id is used to delete
	  the sched item. By writing over the retrans field before the
	  sched item could be deleted, it was possible for a retransmit to
	  occur on a freed frame. ........

	* /, channels/chan_sip.c: Merged revisions 219450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
	  | 14 lines via-header branches not updated correctly on INVITE
	  INVITE requests must always contain a new unique branch id. When
	  a new branch id is created for an INVITE, the dialog's
	  invite_branch variable must be updated so CANCEL requests use the
	  correct branch id. (closes issue #15262) Reported by: maniax
	  Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
	  (license 608) invite_new_branch_trunk.diff uploaded by dvossel
	  (license 671) Tested by: maniax, dvossel ........

2009-09-18 13:54 +0000 [r219412]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Missing value setting line for
	  maxsecs/maxmessage (closes issue #15696) Reported by:
	  fhackenberger Patches: maxsecs.patch uploaded by fhackenberger
	  (license 592)

2009-09-17 22:37 +0000 [r219371]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes deadlock when performing directed
	  pickup w Invite/replaces (closes issue #15340) Reported by:
	  lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license
	  779) Tested by: lmsteffan

2009-09-17 22:22 +0000 [r219324]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 219320 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
	  2009) | 6 lines Send a 100 Trying response when we detect a
	  spiral. This was problematic during spiral tests at SIPit...
	  along with some other things as well. ........

2009-09-17 21:59 +0000 [r219304]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 219303 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
	  | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
	  the locking logic in chan_sip.c's handle_invite_replaces()
	  function as well as making use of ast_do_masquerade() rather than
	  forcing the masquerade on an ast_read(). The code had several
	  redundant unlocks that would result in 'freed more times than
	  we've locked!' errors. I cleaned these up as well as moving all
	  the unlock logic to the end of the function. This patch should
	  also resolve the issue people were having with the replacecall
	  channel never being unlocked with one legged calls. (closes issue
	  #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
	  uploaded by dvossel (license 671) Tested by: irroot, dvossel
	  Review: https://reviewboard.asterisk.org/r/371/ ........

2009-09-17 19:57 +0000 [r219264]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Ensure no spaces exist before "refresher="
	  when doing the comparison.

2009-09-17 16:25 +0000 [r219230]  Sean Bright <sean@malleable.com>

	* apps/app_chanspy.c: Get this compiling under dev-mode.

2009-09-17 15:18 +0000 [r219139]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, /, include/asterisk/cdr.h: Merged revisions
	  219136 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
	  2009) | 10 lines Prevent a potential race condition and crash
	  when hanging up a channel by removing the channel from the
	  channel list before begining channel tear down. This fix may
	  potentially cause problems with CDR backends that access the
	  channel a CDR is associated with via the channel list. This fix
	  makes the channel unavabile at the time when the CDR backend is
	  invoked. This has been documented in include/asterisk/cdr.h.
	  (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
	  Review: https://reviewboard.asterisk.org/r/362/ ........

2009-09-17 00:58 +0000 [r219007-219105]  Tilghman Lesher <tlesher@digium.com>

	* CHANGES, apps/app_chanspy.c: Add the 'E' option to exit ChanSpy,
	  once the single channel it spied upon hangs up. In addition,
	  there's a bit of cleanup to the arguments and documentation, in
	  which I discovered that the last feature added to this
	  application duplicated an option (oops!) and changed that option
	  so that it now works. (closes issue #14909) Reported by: junky
	  Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen
	  (license 10) Tested by: amilcar, junky, flujan, lmadsen

	* /, main/config.c, configs/extensions.conf.sample: Merged
	  revisions 219023 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
	  | 8 lines Properly deal with quotes in the arguments of '#exec'
	  includes. (closes issue #15583) Reported by: pkempgen Patches:
	  20090726__issue15583.diff.txt uploaded by tilghman (license 14)
	  20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
	  169) Tested by: pkempgen ........

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Detect
	  whether we actually have the long double type, before looking for
	  those functions. (closes issue #15017) Reported by: tzafrir
	  Patches: 20090916__issue15017.diff.txt uploaded by tilghman
	  (license 14) Tested by: tzafrir

2009-09-16 20:32 +0000 [r218973]  Sean Bright <sean@malleable.com>

	* res/res_jabber.c: Remove some unused defines from res_jabber.
	  (closes issue #15359) Reported by: snuffy Patches:
	  bug_res_jabber_unused_defines.diff uploaded by snuffy (license
	  35)

2009-09-16 19:25 +0000 [r218933]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Reverse order of args to fread. This way, we
	  don't always write a null byte into byte 1 of the buffer (closes
	  issue #15905) Reported by: ebroad Patches: freadfix.patch
	  uploaded by ebroad (license 878) Tested by: ebroad

2009-09-16 18:31 +0000 [r218918]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: On TCP and TLS connections do not attempt to
	  stop retransmission of the packet internally. This was preventing
	  responses from being properly processed because the packet was
	  not being found causing handle_response to return prematurely.

2009-09-16 18:06 +0000 [r218868]  David Brooks <dbrooks@digium.com>

	* main/pbx.c, /: Merged revisions 218867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
	  | 13 lines Fixes CID pattern matching behavior to mirror that of
	  extension pattern matching. Pattern matching for extensions uses
	  a type of scoring system, giving values for specificity to each
	  character in the pattern. Unfortunately, this is done character
	  by character, in order. This does lead to some less specific
	  patterns being first in line for matching, but it will usually
	  get the job done. This patch merely brings CID matching to the
	  same level as extension matching. This patch does not attempt to
	  tackle the problem shared by extension matching. (closes issue
	  #14708) Reported by: klaus3000 ........

2009-09-16 13:34 +0000 [r218799]  Russell Bryant <russell@digium.com>

	* contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
	  revisions 218798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
	  | 9 lines Remove the IAXy firmware from Asterisk. The firmware
	  can now be found on downloads.digium.com, where the rest of our
	  binary downloads live. This was the last part of our Asterisk
	  tarballs that was considered non-free by Debian. :-) (closes
	  issue #15838) Reported by: paravoid ........

2009-09-15 22:33 +0000 [r218731]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 218730 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15
	  Sep 2009) | 6 lines If the user enters the same password as
	  before, don't signal an error when the change does nothing.
	  (closes issue #15492) Reported by: cbbs70a Patches:
	  20090713__issue15492.diff.txt uploaded by tilghman (license 14)
	  ........

2009-09-15 19:22 +0000 [r218687]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: upward bound checking for port string to int
	  conversion

2009-09-15 16:15 +0000 [r218586]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 218578 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
	  2009) | 8 lines Send request contact header field with response
	  to registrer queries instead of the address of record. (closes
	  issue #14438) Reported by: ravindrad Patches: regquerypatch
	  uploaded by ravindrad (license 684) Tested by: ravindrad ........

2009-09-15 16:12 +0000 [r218583]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Add some changes related to 218430. *
	  Remove thread_spawned in handle_init_event since it was never
	  used * Always check handle_init_event in case a channel is
	  destroyed

2009-09-15 16:04 +0000 [r218579]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_followme.c: Merged revisions 218577 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
	  | 9 lines Ensure FollowMe sets language in channels it creates.
	  Also, not in the original bug report, but related fields are
	  accountcode and musicclass, and the inheritance of datastores.
	  (closes issue #15372) Reported by: Romik Patches:
	  20090828__issue15372.diff.txt uploaded by tilghman (license 14)
	  Tested by: cervajs ........

2009-09-15 15:40 +0000 [r218504-218566]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Use a better method of ensuring
	  null-termination of the buffer while reading the SDP when using
	  TCP.

	* channels/chan_sip.c: Ensure that SDP read from TCP socket is
	  null-terminated.

2009-09-15 15:02 +0000 [r218500]  Kevin P. Fleming <kpfleming@digium.com>

	* /: Merged revisions 218497 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
	  2009) | 1 line Use proper hostname for downloading sound files.
	  ........

2009-09-15 14:59 +0000 [r218499]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix off-by-one error when reading SDP sent
	  over TCP.

2009-09-15 10:24 +0000 [r218465]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_dahdi.c: Fix false error message on
	  DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0)

2009-09-14 22:38 +0000 [r218430]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /,
	  channels/sig_analog.h: Merged revisions 218401 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
	  | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
	  crash in do_monitor. After talking to rmudgett about some of his
	  recent iflist locking changes, it was determined that the only
	  place that would destroy a channel without being explicitly to do
	  so was in handle_init_event. The loop to walk the interface list
	  has been modified to wait to destroy the channel until the
	  dahdi_pvt of the channel to be destroyed is no longer needed.
	  (closes issue #15378) Reported by: samy ........

2009-09-14 20:08 +0000 [r218365]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Add support for multiple interface lists.
	  Also unlink the sig_pri_pri.pvts[] pointer in
	  destroy_dahdi_pvt().

2009-09-14 19:29 +0000 [r218361]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/voicemail.conf.sample, sounds/Makefile,
	  apps/app_voicemail.c: Recorded merge of revisions 218331 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
	  | 4 lines Don't say "Please try again" if we don't give the user
	  another chance to try again. (issue #15055, SWP-129) Reported by:
	  jthurman ........

2009-09-14 18:16 +0000 [r218295]  Joshua Colp <jcolp@digium.com>

	* main/features.c: Do not attempt to add a parking extension if an
	  error occurred while reading the configuration.

2009-09-14 14:57 +0000 [r218224]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_directed_pickup.c: Merged revisions 218223 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
	  2009) | 8 lines Ensure we don't pickup ourselves when doing
	  pickup by exten. (closes issue #15100) Reported by: lmsteffan
	  Patches: (modified) pickup.patch uploaded by lmsteffan (license
	  779) ........

2009-09-13 17:34 +0000 [r218184]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* channels/chan_phone.c: gcc 4.4: Remove a nop memset size 0 that
	  annoys gcc This memset doesn't write beyond the end of the
	  buffer. (tmpbuf has size of 4).

2009-09-13 05:51 +0000 [r218150]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition,
	  is problematic

2009-09-12 13:08 +0000 [r218107]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_rtp_asterisk.c: use the actual given ip address for 'rtp
	  set debug ip <foo>' instead of the word 'ip' (closes issue
	  #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt
	  uploaded by mvanbaak (license 7) Tested by: davidw

2009-09-11 05:58 +0000 [r217990-218050]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Check the origination priority for more matches, not
	  the current priority. Found by Pavel Troller on the -dev list.

	* /, apps/app_queue.c: Merged revisions 217989 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
	  | 3 lines Don't ring another channel, if there's not enough time
	  for a queue member to answer. (Fixes AST-228) ........

2009-09-10 23:49 +0000 [r217954-217987]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Cleanup approach in 217804 and don't reach inside the sig_pvt.

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Allow do not disturb to be set on analog
	  channels via the CLI and AMI.

2009-09-10 23:12 +0000 [r217916]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/iax-friends.sql, channels/chan_sip.c,
	  channels/chan_iax2.c: Make calltoken support work with realtime
	  users and peers. In the course of this, I also found that the
	  results of ast_gethostbyname were being used incorrectly in both
	  chan_iax2 and chan_sip, so both have been fixed.

2009-09-10 22:31 +0000 [r217873-217912]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Cleaned up chan_dahdi iflist handling and
	  locking. * Fixed walking the iflist so it is always done with the
	  iflock locked. * Simplified iflist walking routines. * Created
	  chan_dahdi iflist insertion and extraction routines. * Fixed
	  duplicate_pseudo() malloc fail handling. * Fixed infinite loop in
	  action_dahdishowchannels() when showing a single channel.

	* channels/chan_dahdi.c: Miscellaneous minor changes.

2009-09-10 21:07 +0000 [r217807]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 217806 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10
	  Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call
	  Token security patch inadvertently broke the use of encryption
	  due to the reorganization of code in the socket_process()
	  function. When encryption is used, an incoming full frame must
	  first be decrypted before the information elements can be parsed.
	  The security release mistakenly moved IE parsing before
	  decryption in order to process the new Call Token IE. To resolve
	  this, decryption of full frames is once again done before looking
	  into the frame. This involves searching for an existing callno,
	  checking the pvt to see if encryption is turned on, and
	  decrypting the packet before the internal fields of the full
	  frame are accessed. (closes issue #15834) Reported by: karesmakro
	  Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
	  (license 671) Tested by: dvossel, karesmakro Review:
	  https://reviewboard.asterisk.org/r/355/ ........

2009-09-10 20:52 +0000 [r217744-217804]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix crash during attended transfer over
	  PRI. The owner pointers in the sig_pri_chan structure were not
	  getting updated in dahdi_fixup.

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Stop caller id transmission when offhook
	  event detected. This fixes the problem that would occur if an
	  analog phone was picked up while the caller id was being sent.
	  The caller id before sent the whole spill even after pickup and
	  is now corrected.

2009-09-10 19:39 +0000 [r217730]  Matthias Nick <mnick@digium.com>

	* res/res_musiconhold.c: Sets the correct musicclass after an
	  announcement (closes issue #15279) Reported by: mbeckwell
	  Patches: patch.txt uploaded by mnick (license ) Tested by: mnick
	  (closes issue #15832) Reported by: mbeckwell Patches: patch.txt
	  uploaded by mnick (license 874) Tested by: mnick

2009-09-10 18:29 +0000 [r217663]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't assign UINT_MAX to an INT.

2009-09-10 18:17 +0000 [r217638]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Verify support
	  for wide ODBC character types before using them. (closes issue
	  #15870) Reported by: nic_bellamy

2009-09-10 12:06 +0000 [r217593]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Include ActionID in all events that are
	  responsed to AMI Action SIPShowRegistry (closes issue #15868)
	  Reported by: nic_bellamy Patches:
	  manager_SIPshowregistry_actionid.patch uploaded by nic bellamy
	  (license 299)

2009-09-10 00:35 +0000 [r217560]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and
	  pseudo channels.

2009-09-09 21:48 +0000 [r217524]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2
	  event notifications

2009-09-09 20:09 +0000 [r217482]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't report transfer success until we
	  actually know. 1xx messages are not final. Related to #12713
	  Patch by oej A big thank you to file for finally fixing the
	  transfer() dialplan application. I've been waiting for years for
	  this. Great work!

2009-09-09 18:52 +0000 [r217445]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4
	  has more strict rules for aliasing. It doesn't like a struct
	  sockaddr_in pointer pointing to a struct sockaddr. So we make it
	  a union.

2009-09-09 12:11 +0000 [r217408]  Sean Bright <sean@malleable.com>

	* main/manager.c: Properly terminate the response to the manager
	  Ping action. In passing, correct the formatting of the Timestamp
	  attribute so that there is a space after the colon and before the
	  value. (closes issue #15861) Reported by: Ivan

2009-09-09 10:39 +0000 [r217367-217368]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Not having any TLS session to write to is a
	  serious XMIT_ERROR.

	* channels/chan_sip.c: Formatting and doxygen updates

2009-09-08 23:37 +0000 [r217331-217332]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h, channels/sig_pri.c: Fix memory leak of
	  sig_xxx private structures.

	* channels/chan_dahdi.c: Miscellaneous minor code cleanup in
	  mkintf().

2009-09-08 22:17 +0000 [r217286]  Sean Bright <sean@malleable.com>

	* apps/app_meetme.c: Fix compilation of app_meetme. Reported by
	  ebroad in #asterisk-bugs

2009-09-08 21:17 +0000 [r217236]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri
	  private pointer array.

2009-09-08 20:28 +0000 [r217199]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_meetme.c: Merged revisions 217156 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
	  | 7 lines When MOH is playing on the channel, announcements sent
	  through the conference are not heard. (closes issue #14588)
	  Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
	  uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
	  tilghman ........

2009-09-08 20:06 +0000 [r217158]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/event.h: Add doxygen to ast_event_subscribe for
	  the description. Most importantly, note that a NULL description
	  will cause a crash, as I just experienced that firsthand.

2009-09-08 18:06 +0000 [r217113]  Russell Bryant <russell@digium.com>

	* addons/format_mp3.c: Fix audio problems with format_mp3. This
	  problem was introduced when the AST_FRIENDLY_OFFSET patch was
	  merged. I'm surprised that nobody noticed any trouble when
	  testing that patch, but this fixes the code that fills in the
	  buffer to start filling in after the offset portion of the
	  buffer. (closes issue #15850) Reported by: 99gixxer Patches:
	  issue15850.diff1.txt uploaded by russell (license 2) Tested by:
	  99gixxer

2009-09-08 16:37 +0000 [r217074]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Ensure
	  that the default autoconf CFLAGS are not used. A recent change to
	  the configure script that allows the user to specify CFLAGS
	  and/or LDFLAGS to the script had the unfortunate side effect of
	  letting autoconf's default CFLAGS (-g -O2) feed in to the rest of
	  the build system, thereby overriding the DONT_OPTIMIZE setting in
	  menuselect. That problem is now corrected.

2009-09-08 15:30 +0000 [r217033]  Tilghman Lesher <tlesher@digium.com>

	* res/res_limit.c: Remove what appears to be an unnecessary define.
	  (closes issue #15851) Reported by: tzafrir

2009-09-08 15:23 +0000 [r217015]  Tzafrir Cohen <tzafrir.cohen@xorcom.com>

	* contrib/scripts/live_ast: live_ast: Fix asterisk.conf instead of
	  regenerating it * Don't write asterisk.conf from scratch. Fix the
	  existing one. * Pass extra 'make' command-line arguments to
	  'install' and 'samples'. * Fix some extra typos. closes issue
	  #15019 .

2009-09-08 14:26 +0000 [r216993]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: caller id number empty parse_uri was not
	  being given the correct scheme's, as a result, uri parsing did
	  not parse the username correctly. One of the side effects of this
	  is an empty caller id. (closes issue #15839) Reported by: ebroad
	  Patches: blank_cidv2.patch uploaded by ebroad (license 878)
	  parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
	  ebroad, dvossel

2009-09-07 20:23 +0000 [r216883-216956]  Olle Johansson <oej@edvina.net>

	* doc/manager_1_1.txt: Fixing formatting

	* doc/manager_1_1.txt: Add new actions under "new actions" and not
	  in the top of the document

	* channels/chan_sip.c: Moving another function declared in the
	  middle of forward declarations. Please follow the structure of
	  the source code, thanks. Chan_sip is messy enough as it is :-)

	* channels/chan_sip.c: Move "deprecated_username" to a flag like
	  the others - unsigned int blah:1

	* channels/chan_sip.c: - Doxygen additions - Remove unused string
	  in sip_registry -- "random" - Someone added a function in the
	  middle of all forward declarations... Weird. Moved it out of that
	  section.

	* channels/chan_sip.c: Clean up the "offered_media" code - Add
	  variable for number of known media streams instead of hardcoding
	  in definition of sip_pvt - Rename "text" to "codecs" - beacuse
	  it's what it is - Add documentation for future developers so that
	  we make sure that we define new sdp media types for SRTP-variants

2009-09-07 17:15 +0000 [r216846]  Tilghman Lesher <tlesher@digium.com>

	* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Allow
	  multiple rows to be fetched within the normal mode of operation.

2009-09-07 16:35 +0000 [r216652-216842]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure we reset global_exclude_static at
	  channel reload

	* channels/chan_sip.c: Move capability into sip_cfg. While at it,
	  make sure we reset it at channel reload.

	* channels/chan_sip.c: Move global_regcontext into the sip_cfg
	  structure

	* channels/chan_sip.c: Move contact_ha to sip_cfg structure

	* channels/chan_sip.c: Doxygen updates

	* channels/chan_sip.c: Since it's possible to have more than 999
	  calls, I'm changing the call counter roof to something higher.

	* channels/chan_sip.c: add doxygen and remove duplicate declaration
	  of variable

	* channels/chan_sip.c: After many years, remove VOCAL_DATA_HACK
	  definition

	* channels/chan_sip.c: Remove unneeded header files (tested on
	  Linux and OS/X)

	* channels/chan_sip.c: Don't send MESSAGE with sendtext() if
	  recepient doesn't allow MESSAGE requests

	* channels/chan_sip.c: Add some doxygen

	* channels/chan_sip.c: Fix typo

	* channels/chan_sip.c: If there is no session timer in the INVITE,
	  set it to default value (not unset minimum = -1) Patch by oej
	  closes issue #15621 Reported by: fnordian Tested by: atis

	* configs/sip.conf.sample: Update sip.conf.sample documentation,
	  reorganize a bit

	* channels/chan_sip.c: Simplify the code in this function

2009-09-04 19:32 +0000 [r216594]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: sip peer matching by address only with
	  TCP/TLS This patch removes the contact header matching logic and
	  adds logic to match all tcp/tls connections by ip only. Thanks to
	  oej for finding the issue and suggesting solutions. Review:
	  https://reviewboard.asterisk.org/r/354/

2009-09-04 19:29 +0000 [r216593]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: Use ast_free() instead of free().

2009-09-04 17:50 +0000 [r216547-216551]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix trunk breakage.

	* main/pbx.c, UPGRADE-1.6.txt: Enable turning off the application
	  delimiter warning with the 'dontwarn' option. Suggested on the
	  -dev list, and implemented in an alternate way by me.

2009-09-04 15:05 +0000 [r216506]  Michiel van Baak <michiel@vanbaak.info>

	* /, main/utils.c: Merged revisions 216435 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
	  | 2 lines make asterisk compile under devmode with DEBUG_THREADS
	  enabled on OpenBSD ........

2009-09-04 14:02 +0000 [r216438]  Olle Johansson <oej@edvina.net>

	* main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c,
	  configs/sip.conf.sample, apps/app_playback.c: Merged revisions
	  216430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
	  lines Make apps send PROGRESS control frame for early media and
	  fix too early media issue in SIP The issue at hand is that some
	  legacy (dying) PBX systems send empty media frames on PRI links
	  *before* any call progress. The SIP channel receives these frames
	  and by default signals 183 Session progress and starts sending
	  media. This will cause phones to play silence and ignore the
	  later 180 ringing message. A bad user experience. The fix is
	  twofold: - We discovered that asterisk apps that support early
	  media ("noanswer") did not send any PROGRESS frame to indicate
	  early media. Fixed. - We introduce a setting in chan_sip so that
	  users can disable any relay of media frames before the outbound
	  channel actually indicates any sort of call progress. In 1.4,
	  1.6.0 and 1.6.1, this will be disabled for backward
	  compatibility. In later versions of Asterisk, this will be
	  enabled. We don't assume that it will change your Asterisk phone
	  experience - only for the better. We encourage third-party
	  application developers to make sure that if they have
	  applications that wants to send early media, add a PROGRESS
	  control frame transmission to make sure that all channel drivers
	  actually will start sending early media. This has not been the
	  default in Asterisk previous to this patch, so if you got
	  inspiration from our code, you need to update accordingly. Sorry
	  for the trouble and thanks for your support. This code has been
	  running for a few months in a large scale installation (over 250
	  servers with PRI and/or BRI links to old PBX systems). That's no
	  proof that this is an excellent patch, but, well, it's tested :-)
	  ........

2009-09-04 14:00 +0000 [r216431-216437]  Michiel van Baak <michiel@vanbaak.info>

	* include/asterisk/lock.h: make sure canlog is set so we can
	  compile with DEBUG_THREADS enabled on OpenBSD

	* /: Recorded merge of revisions 216432 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009)
	  | 2 lines make chan_sip compile under devmode again ........

	* /: Recorded merge of revisions 216369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009)
	  | 4 lines Make sure 'start' is always initialized. This is the
	  same as rev 216222 in trunk but 1.4 is affected as well ........

2009-09-04 13:14 +0000 [r216368]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Do not treat every SIP peer as if they were
	  configured with insecure=port. There was a problem in the
	  function responsible for doing peer matching by IP address and
	  port number such that during the second pass for checking for a
	  peer configured with insecure=port, it would end up treating
	  every peer as if it had been configured that way. These changes
	  fix the logic in the peer IP and port comparison callback to
	  handle insecure=port checking properly. This problem was
	  introduced when SIP peers were converted to astobj2. Many thanks
	  to dvossel for noticing this while working on another peer
	  matching issue.

2009-09-04 12:05 +0000 [r216335]  Olle Johansson <oej@edvina.net>

	* doc/janitor-projects.txt: Adding to the janitor list. For new
	  readers: The janitor list is a list of tasks we need help with in
	  the Asterisk project. Taking up one of these is often a good way
	  to get into Asterisk development and getting a lot of developers
	  in the project to be grateful. It's stuff we could spend time on
	  when the bug tracker is empty, when our employers hasn't filled
	  our task lists and our servers is running bugfree and happily
	  without any issues. If you want to start working on one of these
	  small projects, feel free to ask for help in the #asterisk-dev
	  channel on IRC or asterisk-dev mailing list. We'll be more than
	  happy to help you to start and reach goal. Thank you for your
	  help. </end of long commit message>

2009-09-04 10:48 +0000 [r216264]  Russell Bryant <russell@digium.com>

	* /, doc/IAX2-security.txt (added): Merged revisions 216263 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r216263 | russell | 2009-09-04 05:48:00 -0500
	  (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
	  Sep 2009) | 2 lines Add a plain text version of the IAX2 security
	  document. ........ ................

2009-09-04 06:08 +0000 [r216222]  Michiel van Baak <michiel@vanbaak.info>

	* main/astobj2.c: make sure 'start' is always initialized. Makes
	  asterisk compile with --enable-dev-mode

2009-09-03 21:09 +0000 [r216186]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_pri.c: Lets try not to use
	  C++ keywords for variable names.

2009-09-03 19:40 +0000 [r216094]  Doug Bailey <dbailey@digium.com>

	* include/asterisk/callerid.h, channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/sig_analog.h: Added detection
	  DTMF CID without polarity change alert. Added detection of DTMF
	  tone energy levels on FXO channels in chan_dahdi monitoring loop
	  so DTMF CID can be detected without the need of a polarity change
	  precursor. (closes issue #9096) Reported by: fleed Patches:
	  9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
	  Tested by: cyberplant, sum, maturs

2009-09-03 19:38 +0000 [r216009-216092]  Russell Bryant <russell@digium.com>

	* /, UPGRADE.txt: Merged revisions 216085 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r216085 | russell | 2009-09-03 14:36:46 -0500
	  (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
	  Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
	  ........ ................

	* /, doc/IAX2-security.pdf (added): Merged revisions 216008 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r216008 | russell | 2009-09-03 13:44:58 -0500
	  (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
	  Sep 2009) | 2 lines Add IAX2 security document related to
	  AST-2009-006. ........ ................

2009-09-03 18:42 +0000 [r216006]  Kevin P. Fleming <kpfleming@digium.com>

	* main/file.c, doc/lang/language-criteria.txt (added): Document
	  language prompt submission process. This patch adds a document
	  describing the language prompt submission process, licensing
	  terms and other issues related to that process. In addition, it
	  modifies the sound file searching process to support language
	  codes with any number of suffices (not limited to just "xx" or
	  "xx_YY"), so that prompts can be named with gender,
	  customer/company, etc. suffices as well. (closes issue #15771)
	  Reported by: jtodd Patches: language-criteria.txt uploaded by
	  jtodd

2009-09-03 16:31 +0000 [r215955]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, include/asterisk/acl.h,
	  channels/iax2-parser.h, include/asterisk/astobj2.h,
	  channels/iax2.h, main/acl.c, channels/chan_iax2.c,
	  channels/iax2-parser.c, main/astobj2.c: Merge code associated
	  with AST-2009-006 (closes issue #12912) Reported by: rathaus
	  Tested by: tilghman, russell, dvossel, dbrooks

2009-09-03 13:02 +0000 [r215891]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add known internal IP address when
	  autodomain=yes (closes issue #14573) Reported by: pj Patches:
	  sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
	  modified by oej Tested by: pj

2009-09-03 05:57 +0000 [r215838]  Michiel van Baak <michiel@vanbaak.info>

	* doc/manager_1_1.txt: Document that SIPshowpeer and SKINNYshowline
	  now include the configured parkinglot in their response. Prodded
	  by snuff-work on #asterisk-dev IRC channel

2009-09-03 03:43 +0000 [r215800-215801]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Default the callback extension to "s". This
	  is a regression. (closes issue #15764) Reported by: elguero
	  Change-type: bugfix

	* include/asterisk.h: Revert attempt to standardize with
	  _POSIX_C_SOURCE. This did not function in the way that was
	  intended, causing more compatibility issues than it solved. It is
	  best, therefore, that it be simply removed. (Discussed with
	  kpfleming; agreement to remove was reached.)

2009-09-02 23:31 +0000 [r215758]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 215682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
	  | 18 lines Re-send non-100 provisional responses to prevent
	  cancellation From section 13.3.1.1 of RFC 3261: If the UAS
	  desires an extended period of time to answer the INVITE, it will
	  need to ask for an "extension" in order to prevent proxies from
	  canceling the transaction. A proxy has the option of canceling a
	  transaction when there is a gap of 3 minutes between responses in
	  a transaction. To prevent cancellation, the UAS MUST send a
	  non-100 provisional response at every minute, to handle the
	  possibility of lost provisional responses. (closes issue #11157)
	  Reported by: rjain Tested by: twilson Review:
	  https://reviewboard.asterisk.org/r/315/ ........

2009-09-02 23:25 +0000 [r215757]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, CHANGES, channels/sig_pri.c: Made
	  chan_dahdi able to ignore incoming calls that are not in a MSN
	  list for ISDN PTMP CPE spans.

2009-09-02 21:39 +0000 [r215681]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: port string to int conversion using sscanf
	  There are several instances where a port is parsed from a uri or
	  some other source and converted to an int value using atoi(), if
	  for some reason the port string is empty, then a standard port is
	  used. This logic is used over and over, so I created a function
	  to handle it in a safer way using sscanf().

2009-09-02 21:23 +0000 [r215622-215665]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info
	  to sip show peer <foo> and skinny show line <foo> If we had this
	  from the start, debugging the 'parking not using configured
	  parkinglot' bug would have been easier.

	* main/features.c: - lock channel before looking for a channel
	  variable - Init the parkings list member of struct parkinglot.
	  Thanks Sean for the explanation why this should be here.

2009-09-02 19:49 +0000 [r215608]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Fix issue where
	  DTMF CID detect was placing channels into signed linear mode made
	  analog_set_linear_mode return back the mode that was being
	  overwritten so it could be restored later.

2009-09-02 18:37 +0000 [r215567]  Tilghman Lesher <tlesher@digium.com>

	* main/Makefile, main/app.c: Close up to the soft open file limit
	  (same on Linux, but varies drastically on OS X). Also, a Makefile
	  fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd
	  Patches: 20090901__issue14542.diff.txt uploaded by tilghman
	  (license 14) Tested by: jtodd, tilghman Change-type: bugfix

2009-09-02 17:26 +0000 [r215522]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP uri parsing cleanup Now, the scheme
	  passed to parse_uri can either be a single scheme, or a list of
	  schemes ',' delimited. This gets rid of the whole problem of
	  having to create two buffers and calling parse_uri twice to check
	  for separate schemes. Review:
	  https://reviewboard.asterisk.org/r/343/

2009-09-02 16:20 +0000 [r215479]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: like in chan_sip's sip_new skinny should
	  copy the configured parkinglot from a line to the newly created
	  channel. This makes callparking honor the configured parkinglot
	  for skinny lines as well.

2009-09-02 16:08 +0000 [r215466]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP support for keep-alive event keep-alive
	  events are used by Sipura/Linksys for NAT keepalive. There
	  currently don't appear to be any problems with NAT, but everytime
	  a keep-alive event is received, Asterisk responds with a "489 Bad
	  event". This error may indicate to a user that NAT problems exist
	  just because this even is not supported. Now, rather than respond
	  with an error, the packet is consumed and a "200 ok" is sent just
	  to indicate we received the packet. (issue #15084) Patches:
	  chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)

2009-09-02 15:56 +0000 [r215419-215462]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_sip.c: Honor configured parkinglot when parking and
	  retrieving parked calls Thank oej for pointing out the fact that
	  sip_new did not copy parkinglot from the peer into the newly
	  created channel. (closes issue #15538) Reported by: gracedman
	  Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by
	  mvanbaak (license 7) With mod by me to also fix callparking as
	  well (this uploaded patch only fixed retrieving a parked call)
	  Tested by: gracedman, mvanbaak

	* include/asterisk.h: Let's compile again on OpenBSD

2009-09-02 06:23 +0000 [r215382]  Olle Johansson <oej@edvina.net>

	* CHANGES, res/res_mutestream.c (added): Adding MUTEAUDIO()
	  dialplan function and MuteAudio AMI action (pinepeach) Review:
	  https://reviewboard.asterisk.org/r/345/

2009-09-02 01:16 +0000 [r215338]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* /, apps/app_softhangup.c: Merged revisions 215270 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01
	  Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly
	  truncate multi-hyphen channel names In general channel names are
	  in the form Foo/Bar-Z, but the channel name could have multiple
	  hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
	  channel name at the last hyphen. (closes issue #15810) Reported
	  by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
	  dhubbard (license 733) ........

2009-09-01 23:41 +0000 [r215222-215301]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add
	  MASTER_CHANNEL() dialplan function, as well as a useful usage.
	  (closes issue #13140) Reported by: cpina Patches:
	  20090807__issue13140.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen Change-type: feature

	* channels/chan_sip.c: Fix register such that lines with a
	  transport string, but without an authuser, parse correctly.
	  (AST-228)

2009-09-01 20:44 +0000 [r215212]  Russell Bryant <russell@digium.com>

	* addons/format_mp3.c: Fix memory corruption caused by format_mp3.
	  format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames
	  returned by read(). However, it lied. This means that other parts
	  of the code that attempted to make use of the offset buffer would
	  end up corrupting the fields in the ast_filestream structure.
	  This resulted in quite a few crashes due to unexpected values for
	  fields in ast_filestream. This patch closes out quite a few bugs.
	  However, some of these bugs have been open for a while and have
	  been an area where more than one bug has been discussed. So with
	  that said, anyone that is following one of the issues closed
	  here, if you still have a problem, please open a new bug report
	  for the specific problem you are still having. If you do, please
	  ensure that the bug report is based on the newest version of
	  Asterisk, and that this patch is applied if format_mp3 is in use.
	  Thanks! (closes issue #15109) Reported by: jvandal Tested by:
	  aragon, russell, zerohalo, marhbere, rgj (closes issue #14958)
	  Reported by: aragon (closes issue #15123) Reported by:
	  axisinternet (closes issue #15041) Reported by: maxnuv (closes
	  issue #15396) Reported by: aragon (closes issue #15195) Reported
	  by: amorsen Tested by: amorsen (closes issue #15781) Reported by:
	  jensvb (closes issue #15735) Reported by: thom4fun (closes issue
	  #15460) Reported by: marhbere

2009-09-01 19:50 +0000 [r215161]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: Ensure that frame dumps of
	  AST_CONTROL_T38_PARAMETERS frames are properly decoded.

2009-09-01 14:40 +0000 [r215110]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Removing whitespace that causes red dots in
	  reviewboard

2009-08-31 22:02 +0000 [r215069-215070]  Tilghman Lesher <tlesher@digium.com>

	* main/http.c: Fix a trunk compilation warning.

	* main/manager.c: Properly initialize the session to prevent a
	  crash. (closes issue #15774) Reported by: lasko Patches:
	  20090831__issue15774.diff.txt uploaded by tilghman (license 14)
	  Tested by: lasko

2009-08-31 18:17 +0000 [r215023]  Olle Johansson <oej@edvina.net>

	* funcs/func_volume.c: By copying this code I got bad comments in
	  reviewboard... Better fix the original.

2009-08-31 16:18 +0000 [r214819-214945]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c, /: Merged revisions 214940 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31
	  Aug 2009) | 7 lines Also unlock the "other" channel, when
	  returning, due to glare. (closes issue #15787) Reported by:
	  tim_ringenbach Patches: chan_local.diff uploaded by tim
	  ringenbach (license 540) Tested by: tim_ringenbach ........

	* Makefile: Force Darwin on ppc platforms to compile with a target
	  level that supports aliasing.

	* include/asterisk.h, main/poll.c: Various patches, to enable
	  Asterisk to once again compile on Mac OS X. One note on defining
	  _POSIX_C_SOURCE: while this feature test macro works to require
	  certain behaviors on Linux, it works differently on *BSD
	  platforms to REMOVE certain API calls that are not in the POSIX
	  specification, such as vasprintf(3). Thus, defining it while
	  depending upon vasprintf (and other extensions to the POSIX
	  standard) to be defined is a recipe to ensure that Asterisk is
	  only buildable on Linux. Hence, this define which was meant to
	  INCREASE portability, effectively ensures the opposite.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  pbx/pbx_lua.c: If lua is detected with the lua5.1 prefix (or
	  not), adjust the include path accordingly. Based upon feedback to
	  a release announcement on the -users list. See
	  http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html

2009-08-28 22:44 +0000 [r214777]  Russell Bryant <russell@digium.com>

	* configure: Update configure script so that CONFIG_CFLAGS and
	  CONFIG_LDFLAGS doesn't break the build.

2009-08-28 20:14 +0000 [r214702]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 214701 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
	  | 8 lines Modify comment to be a bit more accurate. We have kept
	  this comment around long enough, that it's pretty clear that
	  we're keeping the code, because changing the code would require a
	  pretty fundamental architectural shift. We've also taken
	  criticism in some quarters, because it was believed that it was
	  referring to the code being nasty. No, the code isn't nasty, just
	  the operation itself is rather odd. Fixed for eternity (probably
	  not). ........

2009-08-28 20:01 +0000 [r214696]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, include/asterisk/autoconfig.h.in, configure.ac,
	  makeopts.in: Ensure that CFLAGS and/or LDFLAGS provided to
	  configure script are preserved. Cross-compilation environments
	  want to provide 'defaults' for compiler and linker options, and
	  frequently do this by specifying CFLAGS and LDFLAGS in the
	  environment or as command-line arguments to the configure script.
	  This patch modifies the configure script and Makefile to preserve
	  these settings and ensure they are used in the build process.

2009-08-28 19:13 +0000 [r214654]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Move discardremoteholdretrieval test so it
	  applies only to the specific notification indicator values.

2009-08-28 18:41 +0000 [r214650]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/sched.h: Fix some incorrect documentation of
	  sched_thread functions.

2009-08-28 16:50 +0000 [r214360-214611]  Tilghman Lesher <tlesher@digium.com>

	* res/res_musiconhold.c: Remove unnecessary define for Solaris
	  (closes issue #15358) Reported by: snuffy Patches:
	  bug_res_moh_remove_unneeded_include.diff uploaded by snuffy
	  (license 35)

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  autoconf/libcurl.m4 (added): Merged revisions 214517 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27
	  Aug 2009) | 7 lines Use autoconf to detect libcurl, as this
	  enables cross-compilation checks, something we didn't allow
	  before. (closes issue #15714) Reported by: pprindeville Patches:
	  20090813__issue15714.diff.txt uploaded by tilghman (license 14)
	  Tested by: pprindeville ........

	* main/manager.c: Ensure that we check for the special value
	  CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
	  a_villacis Patches:
	  asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
	  uploaded by a villacis (license 660) (Plus a few of my own, to
	  catch the remaining places within manager.c where it could have
	  been a problem)

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  autoconf/ast_ext_lib.m4: Merged revisions 214436 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27
	  Aug 2009) | 2 lines One more build system change, to make the
	  descriptions look better, if we have better information. ........

	* /, configure, include/asterisk/autoconfig.h.in,
	  autoconf/ast_ext_lib.m4: Merged revisions 214357 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27
	  Aug 2009) | 3 lines Make autoheader descriptions render correctly
	  in our autoconfig.h file. (Figured out while working with issue
	  #14906) ........

2009-08-27 15:57 +0000 [r214309-214355]  Jeff Peeler <jpeeler@digium.com>

	* doc/tex/channelvariables.tex: Add forgotten documentation for new
	  channel variables added in 214309.

	* main/features.c, CHANGES: Add two new dialplan variables when
	  using features Added DYNAMIC_FEATURENAME which holds the last
	  triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the
	  unique channel name on the other side and is set when a dynamic
	  feature is triggered. (closes issue #14663) Reported by: tamiel
	  Patches: 20090313_features.diff uploaded by tamiel (license 712)
	  Tested by: tamiel

2009-08-26 21:56 +0000 [r214272]  Richard Mudgett <rmudgett@digium.com>

	* configs/chan_dahdi.conf.sample: Minor punctuation change.

2009-08-26 16:53 +0000 [r214199]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Typo fix ("SIP/2.0 XXX" is 11 chars, not 10)
	  (closes issue #15362) Reported by: klaus3000 Patches:
	  chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
	  65)

2009-08-26 16:38 +0000 [r214195]  David Vossel <dvossel@digium.com>

	* main/channel.c, /: Merged revisions 214194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
	  | 19 lines ast_write() ignores ast_audiohook_write() results In
	  ast_write(), if a channel has a list of audiohooks, those lists
	  are written to and the resulting frame is what ast_write() should
	  continue with. The problem was the returned audiohook frame was
	  not being handled at all, and the original frame passed into it
	  did not contain the mixed audio, so essentially audio was being
	  lost. One result of this was chan_spy's whisper mode no longer
	  worked. To complicate the issue, frames passed into ast_write may
	  either be a single frame, or a list of frames. So, as the list of
	  frames is processed in the audiohook_write, the returned frames
	  had to be added to a new list. (closes issue #15660) Reported by:
	  corruptor Tested by: dvossel ........

2009-08-25 22:39 +0000 [r213900-214152]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Not
	  all versions of gnu-linux use glibc, which contains iconv. Some
	  (especially embedded systems) don't have iconv at all. (closes
	  issue #15169) Reported by: pprindeville

	* /, main/say.c: Merged revisions 214068-214069 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
	  | 6 lines Fix pronunciation of German dates. (closes issue
	  #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
	  by Benjamin Kluck (license 803) ........ r214069 | tilghman |
	  2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
	  always compile before committing... ........

	* pbx/pbx_dundi.c: DUNDILOOKUP function in 1.6 should use comma
	  delimiters. (closes issue #15322) Reported by: chappell Patches:
	  dundilookup-0015322.patch uploaded by chappell (license 8)

	* main/pbx.c, /: Merged revisions 213970 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
	  | 7 lines Improve error message by informing user exactly which
	  function is missing a parethesis. (closes issue #15242) Reported
	  by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
	  dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
	  loloski (license 68) ........

	* Makefile: The DTD should be installed in the same path as the
	  rest of the XML documentation. (closes issue #15344) Reported by:
	  tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir
	  (license 46)

	* Makefile, /: Merged revisions 213899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
	  | 4 lines Use the default runlevels for Debian derivatives,
	  instead of making up our own. (closes issue #14730) Reported by:
	  pkempgen ........

2009-08-24 16:43 +0000 [r213833]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Fix storage of greetings when using
	  IMAP_STORAGE The store macro was not getting called preventing
	  storage of IMAP greetings at all. This has been corrected along
	  with fixing checking if the imapgreetings option is turned on to
	  store the greeting in IMAP. Lastly, the attachment filename was
	  incorrectly using the full path instead of just the basename,
	  which was causing problems with retrieval of the greeting.
	  (closes issue #14950) Reported by: noahisaac (closes issue
	  #15729) Reported by: lmadsen

2009-08-24 04:46 +0000 [r213790]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: improve handling of
	  openr2_chan_disconnect_call API failure, unlikely, but happened
	  on openr2 library bug

2009-08-21 23:18 +0000 [r213748]  Richard Mudgett <rmudgett@digium.com>

	* configure, configure.ac, channels/sig_pri.c: Update configure
	  script for libpri COLP feature dependency requirements.

2009-08-21 22:36 +0000 [r213738]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Clarifying comments in sip_register, and
	  removing a dead section

2009-08-21 22:22 +0000 [r213716]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Register request line contains wrong address
	  when user domain and register host differ (closes issue #15539)
	  Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch
	  uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded
	  by dvossel (license 671) Tested by: Nick_Lewis, dvossel

2009-08-21 21:39 +0000 [r213697]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: Ensure that realtime mailboxes properly
	  report status on subscription. This patch modifies
	  app_voicemail's response to mailbox status subscriptions (via the
	  internal event system) to ensure that a subscription triggers an
	  explicit poll of the mailbox, so the subscriber can get an
	  immediate cached event with that status. Previously, the cache
	  was only populated with the status of non-realtime mailboxes.
	  (closes issue #15717) Reported by: natmlt

2009-08-21 21:02 +0000 [r213635]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fixes sip register parsing when user@domain
	  is used (issue #15008) (issue #15672)

2009-08-21 16:53 +0000 [r213560]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk.h, /: Merged revisions 213559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
	  | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
	  (closes issue #15698) Reported by: slavon Patches:
	  20090817__issue15698.diff.txt uploaded by tilghman (license 14)
	  Tested by: slavon, tilghman ........

2009-08-21 16:04 +0000 [r213494]  Jason Parker <jparker@digium.com>

	* /, configs/queues.conf.sample: Merged revisions 213493 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
	  5 lines Clarify queues.conf comments to specify that variables
	  should be set in the dialplan. (closes issue #15755) Reported by:
	  trendboy ........

2009-08-21 04:09 +0000 [r213454]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: increment the mfcr2 monitor count when
	  clearing the call request

2009-08-21 03:48 +0000 [r213450]  Terry Wilson <twilson@digium.com>

	* main/loader.c: Make LOAD_ORDER actually work

2009-08-20 22:13 +0000 [r213414]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Add original position, when logging a caller
	  entering a queue. (closes issue #15146) Reported by: arabe
	  Patches: asterisk-trunk.patch uploaded by arabe (license 786)

2009-08-20 21:33 +0000 [r213404]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.c: Fix greeting retrieval from IMAP Properly
	  check for the current voicemail state and if it doesn't exist,
	  create it. (closes issue #14597) Reported by: wtca Patches:
	  14597_v2.patch uploaded by mmichelson (license 60) Tested by:
	  jpeeler

2009-08-20 20:29 +0000 [r213327]  Matthew Nicholson <mnicholson@digium.com>

	* main/features.c: Fix a crash by checking the proper pointer for
	  validity before deferencing it. (closes issue #15751) Reported
	  by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis
	  (license 242)

2009-08-20 19:56 +0000 [r213284]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_voicemail.exports (added), /: Merged revisions 213283
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009)
	  | 2 lines Make all the symbols for the C-client callbacks global
	  ........

2009-08-20 15:29 +0000 [r213248]  Tilghman Lesher <tlesher@digium.com>

	* addons/res_config_mysql.c: Select uncommented lines, not
	  commented ones. (closes issue #15746) Reported by: makoto

2009-08-20 03:26 +0000 [r213216]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: fixed bug caused by calling ast_request
	  without calling ast_call on an R2 channel, ie, CHANISAVAIL

2009-08-19 22:38 +0000 [r213179]  Jason Parker <jparker@digium.com>

	* main/ulaw.c, main/alaw.c: Fix compile when certain G711
	  menuselect options are enabled. (closes issue #15697) Reported
	  by: slavon

2009-08-19 21:21 +0000 [r213113]  David Vossel <dvossel@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 213103 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19
	  Aug 2009) | 8 lines Fixes memory leak caused by incorrectly
	  freeing mixmonitor (closes issue #15699) Reported by: edantie
	  Patches: mixmonitor.patch uploaded by edantie (license 862)
	  ........

2009-08-19 21:05 +0000 [r213093-213098]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Better parsing for
	  the "register" line Allows characters that are otherwise used as
	  delimiters to be used within certain fields (like the secret).
	  (closes issue #15008, closes issue #15672) Reported by: tilghman
	  Patches: 20090818__issue15008.diff.txt uploaded by tilghman
	  (license 14) Tested by: lmadsen, tilghman

	* channels/chan_sip.c: If we have realtime caching enabled, 'sip
	  reload' must purge users/peers, even if the config files haven't
	  changed. (closes issue #12869) Reported by: bcnit Patches:
	  20090819__issue12869__2.diff.txt uploaded by tilghman (license
	  14) Tested by: lasko

2009-08-19 15:32 +0000 [r213046]  Russell Bryant <russell@digium.com>

	* main/features.c: Don't blow up on a NULL cdr. Reported in
	  #asterisk-dev.

2009-08-18 23:53 +0000 [r213007]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, CHANGES, channels/sig_pri.c: Add COLP support
	  to chan_dahdi/sig_pri. Add Connected Line Presentation (COLP)
	  support to chan_dahdi/libpri as an addition to issue 8824. This
	  is the chan_dahdi/sig_pri portion. COLP support is now available
	  for any switch for which libpri supports COLP (currently ETSI
	  PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068)
	  Tested by: rmudgett Review:
	  https://reviewboard.asterisk.org/r/340/

2009-08-18 20:33 +0000 [r212922-212939]  Kevin P. Fleming <kpfleming@digium.com>

	* /: Remove some accidentally-committed properties.

	* CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml,
	  build_tools/prep_tarball, sounds/Makefile, doc/tex/asterisk.tex:
	  Convert this branch to Opsound music-on-hold. For more details:
	  http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/

2009-08-18 19:49 +0000 [r212857-212883]  Tilghman Lesher <tlesher@digium.com>

	* addons/res_config_mysql.c: Clarify some of the error messages, to
	  help upgraders.

	* configs/extconfig.conf.sample: Make the default extconfig.conf
	  match entries with the sample res_mysql.conf. This eliminates a
	  future source of possible confusion with the configuration of
	  1.6.1 and higher.

2009-08-18 18:57 +0000 [r212844]  Olle Johansson <oej@edvina.net>

	* apps/app_meetme.c: Small doxygen changes

2009-08-18 16:38 +0000 [r212764]  Sean Bright <sean@malleable.com>

	* main/manager.c, /: Merged revisions 212763 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
	  2009) | 11 lines Delay the creation of temporary files until we
	  have a valid manager command to handle. Without this patch,
	  asterisk creates a temporary file before determining if the
	  specified command is valid. If invalid, we weren't properly
	  cleaning up the file. (closes issue #15730) Reported by: zmehmood
	  Patches: M15730.diff uploaded by junky (license 177) Tested by:
	  zmehmood ........

2009-08-18 16:29 +0000 [r212758]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/misdn/isdn_lib.c: Merged revisions 212727 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009)
	  | 1 line Removed some deadwood and added some doxygen comments.
	  ........

2009-08-17 20:40 +0000 [r212672]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk.h: Relax check for XOPEN_VERSION. It's not clear
	  that we actually require XOPEN_VERSION to be 600 or greater at
	  this time, so skip the check for now.

2009-08-17 19:57 +0000 [r212627]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Check the return value of opendir(3), or we
	  may crash. (closes issue #15720) Reported by: tobias_e

2009-08-17 18:50 +0000 [r212574-212581]  Sean Bright <sean@malleable.com>

	* channels/chan_agent.c: Correct spelling of AGENTACCEPTDTMF in
	  chan_agent. (closes issue #15668) Reported by: davidw

	* main/logger.c: Correct the return value check for
	  ast_safe_system. The logic here was reversed as ast_safe_system
	  returns -1 on error and not on success. Fix suggested by
	  reporter. (closes issue #15667) Reported by: loic

2009-08-17 16:50 +0000 [r212506]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/misdn_config.c: Merged revisions 212498 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17
	  Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If
	  more ports were specified than configured in misdn.conf a reload
	  would crash asterisk. The problem was the unconfigured port was
	  using data from the previously configured port. When the data for
	  an unconfigured port was freed a crash would result from the
	  double free. (closes issue #12113) Reported by: agupta Patches:
	  bug12113.patch uploaded by jpeeler (license 325) ........

2009-08-17 16:25 +0000 [r212463]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk.h, main/xml.c: Define our desires for POSIX and
	  X/OPEN API features properly. Based on a post on the gcc-help
	  mailing list and some subsequent reading, we can increase our
	  portability to various platforms by directly defining the POSIX
	  and X/OPEN API feature sets we wish to have available. This patch
	  does that, and also includes a double-check to ensure that the
	  system we are compiling on can actually provide the requested
	  feature sets.

2009-08-17 15:42 +0000 [r212431]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  212430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
	  uninitialized variable causing random MWI indications. (closes
	  issue #15727) Reported by: doda Patches: dahdi_changes.patch
	  uploaded by doda (license 853) ........ r212430 | rmudgett |
	  2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
	  uninitialized variable. ........

2009-08-16 19:27 +0000 [r212390]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, include/asterisk/rtp_engine.h: Add two more
	  API calls for getting the current glue and channel in bridging
	  code.

2009-08-15 11:36 +0000 [r212339-212343]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_calendar.c: cast time_t type variables to long where
	  needed. This makes res_calendar.c compile on OpenBSD and the same
	  cast is used in a lot of other places where time_t type vars are
	  used. (closes issue #15656) Reported by: mvanbaak Patches:
	  2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak
	  (license 7)

	* main/xmldoc.c: Add an empty line after each option when printing
	  the documentation of a function/application. This will make
	  reading the docs on the CLI way more easy. (closes issue #15694)
	  Reported by: mvanbaak Patches:
	  2009081100-extralinesoptionlist.diff.txt uploaded by mvanbaak
	  (license 7)

2009-08-14 23:07 +0000 [r212287-212291]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_analog.c: Add braces where missing and a few
	  whitespace fixes in sig_analog (closes issue #15678) Reported by:
	  alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by
	  alecdavis (license 585) Tested by: alecdavis

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: More code that somehow got left out of
	  sig_analog * confirmanswer option now respected * check and set
	  waiting for dialtone timer * unneeded needcallerid flag removed
	  from analog_subchannel * ss_astchan does not need to be a void
	  pointer * swap_channels callback updated to trunk * analog_hangup
	  now resets channel to default law

2009-08-14 17:36 +0000 [r212249]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_curl.c: Add SSL_VERIFYPEER, as requested on the -users
	  list

2009-08-13 17:33 +0000 [r212199]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Send a generic return result when we
	  receive a CallDeflection facility message in chan_misdn. ETSI
	  300-196 implies that a facility return result without arguments
	  does not have the operation-value. This fact implies for ETSI
	  that you can only use the invoke-id to match requests with
	  responses.

2009-08-13 16:44 +0000 [r212161]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, include/asterisk/rtp_engine.h: Add an API call
	  for retrieving the engine in use by an RTP instance.

2009-08-13 15:46 +0000 [r212113]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Ensure that T38FaxVersion is put into
	  outgoing SDP in the proper case.

2009-08-13 13:51 +0000 [r212067]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Check an actual populated variable when
	  seeing if we need to do video or not.

2009-08-13 11:37 +0000 [r212027]  Gavin Henry <ghenry@suretecsystems.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: Fixed typo (closes issue #15710)
	  Reported by: suretec

2009-08-12 23:14 +0000 [r211947-211957]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 211953 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
	  2009) | 10 lines This patch adds additional checking when
	  generating queue log TRANSFER events. The additional checks
	  prevent generation of false TRANSFER events in certain
	  situations. (closes issue #14536) Reported by: aragon Patches:
	  queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
	  Tested by: aragon, mnicholson ........

	* channels/chan_sip.c, configs/sip.conf.sample: This patch adds
	  support for choosing a realm based on the domain in the From or
	  To header in the incoming request. Eligible domains are taken
	  from the domains list in the config file. This functionality is
	  enabled when domainsasrealm is enabled in the config file.
	  (closes issue #11361) Reported by: arkadia Patches:
	  sip_realm_mnich_to_added_2.patch uploaded by arkadia (license
	  233) Tested by: arkadia

2009-08-12 20:47 +0000 [r211908]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Fix chan_dahdi option ringtimeout
	  dahdi_read relies on the dahdi_pvt copy of ringt which was not
	  getting set in sig_analog. This patch adds a callback to do so.
	  (closes issue #15288) Reported by: alecdavis Patches:
	  chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license
	  585) Tested by: alecdavis

2009-08-12 19:53 +0000 [r211876]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Make asterisk handle 423 Interval Too Short
	  messages better. This change uses separate values for the
	  acceptable minimum expiry provided by the 423 error and the
	  expiry value stored in the configuration file. Previously, the
	  value pulled from the configuration file would be overwritten.
	  (closes issue #14366) Reported by: Nick_Lewis Patches:
	  sip-expiry-fix1.diff uploaded by mnicholson (license 96)
	  chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested
	  by: mnicholson

2009-08-12 16:00 +0000 [r211767]  Gavin Henry <ghenry@suretecsystems.com>

	* res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: Added three new attributes and
	  applied a patch to res_config_ldap.c attributetype (
	  AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
	  'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
	  caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
	  attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
	  'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
	  caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
	  attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
	  DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
	  SUBSTR caseIgnoreSubstringsMatch SYNTAX
	  1.3.6.1.4.1.1466.115.121.1.15) and patch
	  fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
	  Reported by: macogeek Patches:
	  fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
	  863) Tested by: suretec

2009-08-12 10:11 +0000 [r211732]  Russell Bryant <russell@digium.com>

	* channels/chan_jingle.c, channels/chan_unistim.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_gtalk.c, channels/chan_mgcp.c: Always specify which
	  RTP engine is desired for a new RTP instance. This fixes a crash
	  reported in #asterisk-dev where chan_mgcp unexpectedly allocated
	  an RTP instance from res_rtp_multicast, since by not specifying
	  an engine, you get the first one in the list of engines.

2009-08-10 23:21 +0000 [r211675]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Encapsulate testing for which signaling
	  styles are used by sig_pri. Created the
	  dahdi_sig_pri_lib_handles() function and SIG_PRI_LIB_HANDLE_CASES
	  macro to simplify testing for which signaling styles are handled
	  by sig_pri.

2009-08-10 19:49 +0000 [r211539-211584]  Tilghman Lesher <tlesher@digium.com>

	* doc/CODING-GUIDELINES, /: Merged revisions 211583 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
	  Aug 2009) | 1 line Conversion specifiers, not format specifiers
	  ........

	* cel/cel_pgsql.c, funcs/func_speex.c, funcs/func_rand.c,
	  apps/app_dahdibarge.c, main/frame.c, addons/chan_ooh323.c,
	  apps/app_readfile.c, /, apps/app_record.c,
	  apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
	  res/res_http_post.c, channels/chan_iax2.c, main/indications.c,
	  main/config.c, main/cli.c, pbx/pbx_loopback.c,
	  channels/chan_dahdi.c, pbx/pbx_spool.c, res/res_smdi.c,
	  channels/chan_skinny.c, main/features.c, main/http.c, main/pbx.c,
	  funcs/func_sprintf.c, funcs/func_timeout.c, apps/app_privacy.c,
	  codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c,
	  apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c,
	  funcs/func_cut.c, apps/app_talkdetect.c, main/netsock.c,
	  res/res_config_curl.c, channels/chan_misdn.c,
	  apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
	  addons/cdr_mysql.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
	  apps/app_chanspy.c, main/asterisk.c, res/res_odbc.c,
	  cel/cel_adaptive_odbc.c, main/timing.c, apps/app_voicemail.c,
	  doc/CODING-GUIDELINES, addons/app_mysql.c, utils/muted.c,
	  apps/app_meetme.c, main/utils.c, res/res_musiconhold.c,
	  cdr/cdr_pgsql.c, apps/app_followme.c, res/res_config_sqlite.c,
	  main/enum.c, utils/frame.c, channels/misdn_config.c,
	  main/channel.c, res/ael/pval.c, main/cdr.c, funcs/func_enum.c,
	  channels/chan_phone.c, main/manager.c, apps/app_setcallerid.c,
	  apps/app_osplookup.c, funcs/func_odbc.c, res/res_agi.c,
	  apps/app_minivm.c, channels/xpmr/xpmr.c, res/res_config_ldap.c,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
	  res/res_config_pgsql.c, funcs/func_dialplan.c, main/dnsmgr.c,
	  channels/chan_sip.c, res/res_limit.c, apps/app_waitforsilence.c,
	  agi/eagi-test.c, main/acl.c, apps/app_waituntil.c,
	  apps/app_originate.c, channels/sig_pri.c, apps/app_queue.c,
	  channels/chan_oss.c, agi/eagi-sphinx-test.c,
	  channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
	  apps/app_sms.c, utils/extconf.c, apps/app_stack.c,
	  apps/app_verbose.c, addons/app_saycountpl.c, main/dsp.c,
	  addons/res_config_mysql.c: AST-2009-005

2009-08-10 18:01 +0000 [r211475]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: add manager events when a skinny device
	  registers/unregisters like we have in chan_sip (closes issue
	  #15499) Reported by: arifzaman Patches:
	  2009072600-skinnymanagerevents.diff.txt uploaded by mvanbaak
	  (license 7)

2009-08-10 17:17 +0000 [r211435]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_pri.c: Fix PRI/BRI channels
	  when in alarm condition to only be marked for hangup if T309 is
	  not enabled.

2009-08-10 15:53 +0000 [r211392]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Restoring some code to sig_pri. Not sure if it is really needed.
	  Putting some DSP code back into sig_pri that was removed by the
	  chan_dahdi/sig_pri reorganization.

2009-08-10 15:46 +0000 [r211390]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix up some issues with getting a channel by
	  "name". Even though the get_channel_by_name() API advertised that
	  you could search by name or uniqueid (just as the old API did),
	  searching by uniqueid was not actually implemented. This patch
	  fixes that problem. The ast_channel_get_full() function now makes
	  a second search attempt by uniqueid if the parameter was a name.
	  The channel comparison function also now knows how to compare by
	  unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER
	  was being passed in some scenarios where it should not have been.

2009-08-10 14:07 +0000 [r211347]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix retrieval of the port used for the video
	  stream when adding SDP to a SIP message. (closes issue #15121)
	  Reported by: jsmith

2009-08-09 15:42 +0000 [r211232-211275]  Tilghman Lesher <tlesher@digium.com>

	* /, main/astfd.c: Merged revisions 211274 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
	  | 2 lines Small oops. Clear the flags which have been checked.
	  ........

	* apps/app_stack.c: Check for NULL frame, before dereferencing
	  pointer. (closes issue #15617) Reported by: rain

2009-08-07 23:30 +0000 [r211191-211197]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Fixed some unsafe down cast pointer
	  operations for sig_pri. You cannot cast the struct
	  dahdi_pvt.sig_pvt pointer to a specific signaling private pointer
	  without first checking that it is in fact pointing to the correct
	  signaling private structure.

	* channels/sig_pri.c: Fix static on line when PRI does overlap
	  dialing. The wrong encoding law was used because = was used when
	  it should have been ==.

2009-08-07 20:12 +0000 [r211113]  Russell Bryant <russell@digium.com>

	* /: Recorded merge of revisions 211112 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
	  | 4 lines Resolve a deadlock involving app_chanspy and
	  masquerades. (ABE-1936) ........

2009-08-07 18:17 +0000 [r211040]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_queue.c: Merged revisions 211038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
	  | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
	  not the membername. This is a partial revert of revision 82590,
	  which was an attempted cleanup, but in reality, it broke
	  QUEUE_MEMBER_LIST, which has always been intended as a method by
	  which component interfaces could be queried from the queue.
	  Membername isn't useful here, because that field cannot be used
	  to obtain further information about the member. See the
	  documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
	  QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
	  member argument for further justification. (closes issue #15664)
	  Reported by: rain Patches: app_queue-queue_member_list.diff
	  uploaded by rain (license 327) ........

2009-08-07 13:08 +0000 [r210992]  Kevin P. Fleming <kpfleming@digium.com>

	* main/udptl.c: Workaround broken T.38 endpoints that offer tiny
	  MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
	  the maximum IFP size that should be sent to them, rather than the
	  maximum packet payload size. If such an endpoint also requests
	  UDPRedundancy as the error correction mode, we'll end up
	  calculating a tiny maximum IFP size, so small as to be unusable.
	  This patch sets a lower bound on what we'll consider the remote's
	  maximum IFP size to be, assuming that endpoints that do this
	  really can accept larger packets than they've offered to accept.
	  (closes issue #15649) Reported by: dazza76

2009-08-06 21:46 +0000 [r210908-210914]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 210913 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
	  | 7 lines Because channel information can be accessed outside of
	  the channel thread, we must lock the channel prior to modifying
	  it. (closes issue #15397) Reported by: caspy Patches:
	  20090714__issue15397.diff.txt uploaded by tilghman (license 14)
	  Tested by: caspy ........

	* include/asterisk/app.h, main/app.c, apps/app_stack.c: Allow Gosub
	  to recognize quote delimiters without consuming them. (closes
	  issue #15557) Reported by: rain Patches:
	  20090723__issue15557.diff.txt uploaded by tilghman (license 14)
	  Tested by: rain Review: https://reviewboard.asterisk.org/r/316/

2009-08-06 20:15 +0000 [r210866-210869]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_analog.c: Miscellaneous minor fixes to sig_analog. *
	  Sanity adjustments to __analog_ss_thread for sig_analog
	  environment. * Deleted some duplicated code. * Fixed
	  analog_ss_thread_start passing the wrong pointer.

	* channels/sig_pri.c: Sanity adjustments to pri_ss_thread for
	  sig_pri environment.

2009-08-06 17:47 +0000 [r210817]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Accept additional T.38 reinvites after an
	  initial one has been handled. Discussion of this subject has
	  yielded that it is not actually acceptable to change T.38
	  parameters after the initial reinvite but declining is harsh and
	  can cause the fax to fail when it may be possible to allow it to
	  continue. This patch changes things so that additional T.38
	  reinvites are accepted but parameter changes ignored. This gives
	  the fax a fighting chance. (closes issue #15610) Reported by:
	  huangtx2009

2009-08-06 16:07 +0000 [r210777]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
	  configure.ac: Minor improvements to app_fax. This patch makes
	  some small changes to handle watchdog timeouts in a better way,
	  and also uses a 'cleaner' method of including the spandsp header
	  files. (closes issue #14769) Reported by: andrew Patches:
	  app_fax-20090406.diff uploaded by andrew (license 240)
	  v1-14769.patch uploaded by dimas (license 88) Tested by: freh,
	  deti, caspy, dimas, sgimeno, Dovid

2009-08-05 23:44 +0000 [r210640-210732]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.c: Fix potential deadlock issue with
	  USERUSERINFO channel variable.

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  More changes from chan_dahdi that did not make it into sig_pri. *
	  Q.SIG channel mapping option. * discardremoteholdretrieval
	  option. * libPRI debug defines. * pri_set_overlapdial() now set
	  correctly. * pthread creation of pri_ss_thread now matches.

	* /, channels/sig_pri.c: Merged revisions 210575 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
	  | 14 lines Dialplan starts execution before the channel setup is
	  complete. * Issue 15655: For the case where dialing is complete
	  for an incoming call, dahdi_new() was asked to start the PBX and
	  then the code set more channel variables. If the dialplan hungup
	  before these channel variables got set, asterisk would likely
	  crash. * Fixed potential for overlap incoming call to erroneously
	  set channel variables as global dialplan variables if the
	  ast_channel structure failed to get allocated. * Added missing
	  set of CALLINGSUBADDR in the dialing is complete case. (closes
	  issue #15655) Reported by: alecdavis ........

2009-08-05 18:49 +0000 [r210564]  Leif Madsen <lmadsen@digium.com>

	* doc/tex/imapstorage.tex, /: Merged revisions 210563 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05
	  Aug 2009) | 11 lines Update imapstorage.txt documentation.
	  Updated the imapstorage.txt documentation to reflect that issues
	  with c-client versions older than 2007 seem to cause crashing
	  issues that are not seen with more recent versions. Documentation
	  has been updated to reflect this. (closes issue #14496) Reported
	  by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
	  uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
	  dbrooks ........

2009-08-05 14:09 +0000 [r210522]  Russell Bryant <russell@digium.com>

	* main/file.c: Revert some silly code that snuck into trunk from my
	  working copy. Sorry!

2009-08-05 08:03 +0000 [r210478]  Michiel van Baak <michiel@vanbaak.info>

	* addons/mp3: ignore the .i files when compiling in 'DONT_OPTIMIZE'
	  in the addons/mp3 directory

2009-08-04 17:46 +0000 [r210353-210387]  Richard Mudgett <rmudgett@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Fix CALLERID() values for sig_pri on incoming calls.

	* main/channel.c, include/asterisk/channel.h: Initial minimum
	  ast_party_caller support.

	* channels/chan_dahdi.c: Removed some dead code.

2009-08-04 15:35 +0000 [r210302]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Fix broken call pickup The find_channel_by_group
	  callback was only looking at the channel that was attempting to
	  make the pickup instead of the other channels in the container.

2009-08-04 14:53 +0000 [r210190-210238]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 210237 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
	  2009) | 10 lines Eliminate spurious compiler warnings from system
	  headers on *BSD platforms. Ensure that system headers located in
	  /usr/local/include are actually treated as system headers by the
	  compiler, and not as local headers which are subject to warnings
	  from the -Wundef compiler option and others. (closes issue
	  #15606) Reported by: mvanbaak ........

	* contrib/scripts/realtime_pgsql.sql, channels/chan_sip.c,
	  channels/chan_skinny.c, configs/mgcp.conf.sample,
	  doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
	  configs/res_ldap.conf.sample, configs/sip.conf.sample,
	  configs/skinny.conf.sample, channels/chan_mgcp.c,
	  doc/chan_sip-perf-testing.txt: Rename 'canreinvite' option to
	  'directmedia', with backwards compatibility. It is clear from
	  multiple mailing list, forum, wiki and other sorts of posts that
	  users don't really understand the effects that the 'canreinvite'
	  config option actually has, and that in some cases they think
	  that setting it to 'no' will actually cause various other
	  features (T.38, MOH, etc.) to not work properly, when in fact
	  this is not the case. This patch changes the proper name of the
	  option to what it should have been from the beginning
	  ('directmedia'), but preserves backwards compatibility for
	  existing configurations.

2009-08-03 18:05 +0000 [r210094-210154]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, channels/sig_pri.c: Changes from
	  chan_dahdi that did not make it into sig_pri. * Moved
	  SUPPORT_USERUSER to sig_pri.c * Fix PRI_DEADLOCK_AVOIDANCE
	  parameter. * Whitespace changes. * Added missing unlock in
	  pri_dchannel():PRI_EVENT_RING case. * Balanced curly braces. *
	  ast_debug/ast_log changes from chan_dahdi. * sig_pri_indicate()
	  should default to return -1 if the indication is not handled.

	* channels/sig_pri.h, channels/sig_analog.c, channels/sig_pri.c:
	  Trim trailing whitespace.

2009-08-03 14:29 +0000 [r210027]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Fix order and redundancy of channel rename
	  manager events in ast_do_masquerade. Patch contributed by Mark
	  Spencer.

2009-08-03 14:01 +0000 [r209993]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c, configs/chan_mobile.conf.sample: Add an
	  'sms' option to mobile.conf to manually enable or disable SMS
	  support. (closes issue #15071) Reported by: ughnz Patches:
	  optional-sms1.diff uploaded by mnicholson (license 96) Tested by:
	  ughnz, mnicholson

2009-08-01 23:33 +0000 [r209958-209959]  Bradley Latus <brad.latus@gmail.com>

	* doc/tex/realtime.tex: Update documentation in relation to
	  UnixODBC (closes issue #15516) Reported by: snuffy Patches:
	  bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35)

	* doc/CODING-GUIDELINES: (closes issue #15515)

2009-08-01 11:29 +0000 [r209835-209887]  Russell Bryant <russell@digium.com>

	* /, main/db1-ast/mpool/mpool.c: Merged revisions 209879 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
	  | 5 lines Resolve a valgrind warning about a read from
	  uninitialized memory. (issue #15396) Reported by: aragon ........

	* /, apps/app_milliwatt.c: Merged revisions 209838 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01
	  Aug 2009) | 13 lines Modify how Playtones() is used in
	  Milliwatt() to resolve gain issue. When Milliwatt() was changed
	  internally to use Playtones() so that the proper tone was used,
	  it introduced a drop in gain in the output signal. So, use the
	  playtones API directly and specify a volume argument such that
	  the output matches the gain of the original Milliwatt() code.
	  (closes issue #15386) Reported by: rue_mohr Patches:
	  issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
	  rue_mohr ........

	* main/event.c: Fix ast_event_queue_and_cache() to actually do the
	  cache() part. (closes issue #15624) Reported by: ffossard Tested
	  by: russell

2009-08-01 01:04 +0000 [r209760-209761]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Revert accidental Makefile change.

	* Makefile, channels/chan_dahdi.c, channels/chan_misdn.c, /,
	  main/Makefile, channels/misdn/ie.c, pbx/pbx_config.c,
	  utils/frame.c: Merged revisions 209759 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
	  2009) | 7 lines Minor changes inspired by testing with latest
	  GCC. The latest GCC (what will become 4.5.x) has a few new
	  warnings, that in these cases found some either downright buggy
	  code, or at least seriously poorly designed code that could be
	  improved. ........

2009-07-31 21:53 +0000 [r209711]  Russell Bryant <russell@digium.com>

	* main/event.c: Fix some places where ast_event_type was used
	  instead of ast_event_ie_type.

2009-07-31 17:57 +0000 [r209673-209674]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample: Add configuration sample code for
	  previous commit.

	* channels/chan_sip.c: Improve chan_sip's ability to determine what
	  methods should and should not be used in a dialog. The previous
	  effort here was to store what a peer is capable of receiving by
	  parsing REGISTER requests from the peer and keeping that
	  information for as long as the registration was active. The
	  problem with this is that there are a great number of SIP devices
	  which give no indication of the methods allowed in their REGISTER
	  requests, and it is unreasonable to try to guess what the device
	  may or may not support. In addition, some SIP devices have been
	  found to claim support for a specific method, but their handling
	  the method is less than ideal, or they are actually lying. With
	  this patch, we now determine what methods a device supports by
	  parsing the Allow header we receive from them, and we do this
	  with each new dialog. In addition, a configuration option has
	  been added so that an administrator can essentially blacklist
	  certain methods from being used with certain peers if the admin
	  knows that support for a specific method is dodgy or nonexistent.
	  ABE-1822

2009-07-30 23:37 +0000 [r209623]  Sean Bright <sean@malleable.com>

	* configure, configure.ac, makeopts.in: Allow passing 'noisy' to
	  configure's --enable-dev-mode argument to turn on verbose builds.
	  (closes issue #15607) Reported by: mvanbaak Patches:
	  20090730_issue15607.patch uploaded by seanbright (license 71)
	  Tested by: seanbright

2009-07-30 23:31 +0000 [r209619]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.h, channels/sig_pri.c: Add missing ifdef-s for
	  service maintenance message functionality (closes issue #15614)
	  Reported by: fabled

2009-07-30 16:07 +0000 [r209554]  David Brooks <dbrooks@digium.com>

	* channels/sig_pri.h, apps/app_forkcdr.c, channels/chan_dahdi.c,
	  contrib/init.d/rc.debian.asterisk, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooGkClient.h, funcs/func_math.c,
	  apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c,
	  include/asterisk/abstract_jb.h: Fixes numerous spelling errors.
	  Patch submitted by alecdavis. (closes issue #15595) Reported by:
	  alecdavis

2009-07-30 14:38 +0000 [r209516]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a crash that can result if text codecs
	  are allowed but textsupport is disabled. (closes issue #15596)
	  Reported by: fabled Patches: sip-red.patch uploaded by fabled
	  (license 448)

2009-07-29 21:46 +0000 [r209453-209484]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c: This patch adds the ability to send a CUSD
	  command to a bluetooth device. (closes issue #15278) Reported by:
	  Artem Patches: cusd5.patch uploaded by Artem (license 800) Tested
	  by: mnicholson, Artem Review:
	  https://reviewboard.asterisk.org/r/274/

	* addons/chan_mobile.c: Fixed a comment for hfp_parse_clip

2009-07-28 13:49 +0000 [r209400]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_usbradio.c, include/asterisk/utils.h,
	  channels/chan_sip.c, channels/chan_alsa.c,
	  channels/chan_console.c, channels/chan_oss.c, main/poll.c: Define
	  side-effect-safe MIN and MAX macros and remove duplicate
	  definitions from various files.

2009-07-28 00:20 +0000 [r209317-209331]  Tilghman Lesher <tlesher@digium.com>

	* sounds/sounds.xml: Regex FTL

	* /, sounds/sounds.xml: Merged revisions 209315 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
	  | 2 lines Publish French extra sounds ........

2009-07-27 21:43 +0000 [r209256-209279]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Cleanup T.38 negotiation changes. Convert
	  LOG_NOTICE messages about T.38 negotiation in debug level 1
	  messages, clean up some looping logic, and correct an improper
	  use of ast_free() for freeing an ast_frame.

	* apps/app_fax.c: Make T.38 switchover in ReceiveFAX synchronous.
	  In receive mode, if the channel that ReceiveFAX is running on
	  supports T.38, we should *always* attempt to switch T.38, rather
	  than listening for an incoming CNG tone and only triggering on
	  that. The channel may be using a low-bitrate codec that distorts
	  the CNG tone, the sending FAX endpoint may not send CNG at all,
	  or there could be a variety of other reasons that we don't detect
	  it, but in all those cases if T.38 is available we certainly want
	  to use it.

2009-07-27 20:54 +0000 [r209132-209235]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c: Gracefully handle malformed RTP text
	  packets. AST-2009-004

	* res/res_musiconhold.c: Honor channel's music class when using
	  realtime music on hold. (closes issue #15051) Reported by: alexh
	  Patches: 15051.patch uploaded by mmichelson (license 60) Tested
	  by: alexh

	* main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
	  209131 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
	  2009) | 18 lines Allow for UDPTL to use only even-numbered ports
	  if desired. There are some VoIP providers out there that will not
	  accept SDP offers with odd numbered UDPTL ports. While it is my
	  personal opinion that these VoIP providers are misinterpreting
	  RFC 2327, it really is not a big deal to play along with their
	  silly little games. Of course, since restricting UDPTL ports to
	  only even numbers reduces the range of available ports by half,
	  so the option to use only even port numbers is off by default. A
	  user can enable the behavior by setting use_even_ports=yes in
	  udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
	  15182.patch uploaded by mmichelson (license 60) Tested by:
	  CGMChris ........

2009-07-27 16:33 +0000 [r209098]  David Brooks <dbrooks@digium.com>

	* channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c,
	  include/asterisk/module.h, main/features.c, pbx/pbx_dundi.c,
	  res/res_jabber.c, addons/chan_mobile.c, apps/app_rpt.c,
	  main/loader.c: Fixing typos. Replaces "recieved" with "received"
	  and "initilize" with "initialize" (closes issue #15571) Reported
	  by: alecdavis

2009-07-27 15:38 +0000 [r209056]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
	  underscore-variants to sub-makes. During the recent Makefile
	  improvements I made, it seemed the 'make' was automatically
	  carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
	  I removed the explict export of them. However, there are some
	  circumstances where make does this, and some where it does not,
	  so I've brought them back to ensure they are always exported. I
	  also removed an extraneous double setting of _ASTLDFLAGS on *BSD
	  platforms.

2009-07-27 01:20 +0000 [r208924]  Jeff Peeler <jpeeler@digium.com>

	* /, main/translate.c, channels/chan_iax2.c: Merged revisions
	  208923 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
	  | 2 lines Fix logic errors from 208746 ........

2009-07-26 14:00 +0000 [r208886]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/install_prereq: add OpenBSD to the install_prereq
	  script

2009-07-25 12:28 +0000 [r208813-208848]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/install_prereq: libxml2-dev is needed as well by
	  default.

	* configs/cli_aliases.conf.sample, main/cli.c: add default alias
	  reload to run module reload. Requiring 'module reload' to reload
	  everything, including core etc makes russell very unhappy. The
	  default configuration already loads the 'friendly' aliases
	  template. Added 'reload=module reload' to that template. Also
	  removed the comment in main/cli.c that reload should come back.

2009-07-25 06:23 +0000 [r208749]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/chan_skinny.c, main/translate.c,
	  channels/chan_iax2.c: Merged revisions 208746 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
	  | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
	  trivial changes, but I did not know of any other way to fix the
	  "dereferencing type-punned pointer will break strict-aliasing
	  rules" error without creating a tmp variable in chan_skinny.
	  ........

2009-07-24 21:12 +0000 [r208593-208709]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Remove trailing whitespace.

	* main/cli.c: Note that "reload" needs to be added back. I keep
	  getting annoyed at having to type "module reload" to reload
	  everything, so I'm adding a note that we need to add "reload"
	  back. "module reload" doesn't really make sense as the command to
	  reload everything, including the core.

	* main/cli.c: Don't log a warning for something that does not
	  affect operation.

	* apps/app_dial.c, /: Merged revisions 208592 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
	  | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
	  This does not indicate an error. A return of -1 just means that
	  the channel has been hung up. (reported in #asterisk-dev)
	  ........

2009-07-24 18:31 +0000 [r208588]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 208587 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
	  2009) | 10 lines Only send a BYE when hanging up a channel that
	  is up. For cases where Asterisk sends an INVITE and receives a
	  non 2XX final response, Asterisk would follow the INVITE
	  transaction by immediately sending a BYE, which was unnecessary.
	  (closes issue #14575) Reported by: chris-mac ........

2009-07-24 15:02 +0000 [r208548]  Kevin P. Fleming <kpfleming@digium.com>

	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
	  Resolve a T.38 negotiation issue left over from the udptl-updates
	  merge. The udptl-updates branch that was merged yesterday failed
	  to properly send back T.38 SDP responses with the correct error
	  correction mode, if the incoming SDP from the other end caused us
	  to change error correction modes. This patch corrects that
	  situation.

2009-07-24 14:35 +0000 [r208542]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/install_prereq: use aptitude for debian based
	  systems The function to check wether we need to install packages
	  was using dpkg-query which was gives wrong output on Debian 5
	  Also, the apt-get has been replaced with aptitude because
	  aptitude is now the preferred way to handle packages on Debian
	  (closes issue #15570) Reported by: mvanbaak Patches:
	  2009072400_installprereq-aptitude.diff uploaded by mvanbaak
	  (license 7)

2009-07-23 22:32 +0000 [r208464-208504]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: T.38 change note is not necessary in this branch

	* main/channel.c, main/udptl.c, main/frame.c, main/rtp_engine.c,
	  channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt,
	  include/asterisk/udptl.h, include/asterisk/frame.h: Rework of
	  T.38 negotiation and UDPTL API to address interoperability
	  problems Over the past couple of months, a number of issues with
	  Asterisk negotiating (and successfully completing) T.38 sessions
	  with various endpoints have been found. This patch attempts to
	  address many of them, primarily focused around ensuring that the
	  endpoints' MaxDatagram size is honored, and in addition by
	  ensuring that T.38 session parameter negotiation is performed
	  correctly according to the ITU T.38 Recommendation. The major
	  changes here are: 1) T.38 applications in Asterisk (app_fax) only
	  generate/receive IFP packets, they do not ever work with UDPTL
	  packets. As a result of this, they cannot be allowed to generate
	  packets that would overflow the other endpoints' MaxDatagram size
	  after the UDPTL stack adds any error correction information. With
	  this patch, the application is told the maximum *IFP* size it can
	  generate, based on a calculation using the far end MaxDatagram
	  size and the active error correction mode on the T.38 session.
	  The same is true for sending *our* MaxDatagram size to the remote
	  endpoint; it is computed from the value that the application says
	  it can accept (for a single IFP packet) combined with the active
	  error correction mode. 2) All treatment of T.38 session
	  parameters as 'capabilities' in chan_sip has been removed; these
	  parameters are not at all like audio/video stream capabilities.
	  There are strict rules to follow for computing an answer to a
	  T.38 offer, and chan_sip now follows those rules, using the
	  desired parameters from the application (or channel) that wants
	  to accept the T.38 negotiation. 3) chan_sip now stores and
	  forwards ast_control_t38_parameters structures for tracking 'our'
	  and 'their' T.38 session parameters; this greatly simplifies
	  negotiation, especially for pass-through calls. 4) Since T.38
	  negotiation without specifying parameters or receiving the final
	  negotiated parameters is not very worthwhile, the AST_CONTROL_T38
	  control frame has been removed. A note has been added to
	  UPGRADE.txt about this removal, since any out-of-tree
	  applications that use it will no longer function properly until
	  they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
	  https://reviewboard.asterisk.org/r/310/

2009-07-23 19:34 +0000 [r208388]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 208386 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
	  2009) | 17 lines Fix a problem where a 491 response could be sent
	  out of dialog. This generalizes the fix for issue 13849. The
	  initial fix corrected the problem that Asterisk would reply with
	  a 491 if a reinvite were received from an endpoint and we had not
	  yet received an ACK from that endpoint for the initial INVITE it
	  had sent us. This expansion also allows Asterisk to appropriately
	  handle an INVITE with authorization credentials if Asterisk had
	  not received an ACK from the previous transaction in which
	  Asterisk had responded to an unauthorized INVITE with a 407.
	  (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
	  uploaded by mmichelson (license 60) Tested by: klaus3000 ........

2009-07-23 19:21 +0000 [r208383]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 208380 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23
	  Jul 2009) | 6 lines Only set the priindication setting when not
	  performing a reload (closes issue #14696) Reported by: fdecher
	  ........

2009-07-23 16:29 +0000 [r208314]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 208312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
	  2009) | 3 lines Remove inaccurate XXX comment. ........

2009-07-23 15:59 +0000 [r208267]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
	  Fix sending of interface identifier unconditionally in sig_pri
	  The wrong logic was being used in chan_dahdi to convert a
	  sig_pri_chan to the proper libpri channel number. The most
	  significant bit must only be set only when trunk groups are being
	  used. (closes issue #15452) Reported by: alecdavis Patches:
	  bug15452.patch uploaded by jpeeler (license 325) Tested by:
	  alecdavis

2009-07-23 15:46 +0000 [r208229-208263]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 208262 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
	  2009) | 8 lines Properly handle 183 responses which do not
	  contain an SDP. (closes issue #15442) Reported by: ffloimair
	  Patches: 15442.patch uploaded by mmichelson (license 60) Tested
	  by: tkarl, ffloimair ........

	* channels/chan_sip.c: Fix potential crash if p->owner is NULL.
	  Problem was observed when a call-forwarding loop was accidentally
	  configured. ABE-1906

2009-07-23 01:31 +0000 [r208193]  Russell Bryant <russell@digium.com>

	* main/cel.c: Resolve compiler warning on mac.

2009-07-22 22:42 +0000 [r208155]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Reset the fax buffers back to default
	  settings regardless of signaling in use - Pointed out by Matt F.
	  Also in the case of not using a signaling module, set the law
	  back to the default as well.

2009-07-22 22:35 +0000 [r208151]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/compat.h, main/strcompat.c,
	  main/asterisk.exports: Merged revisions 208083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009)
	  | 4 lines Export symbols for functions included in our
	  compatibility headers. (closes issue #15556) Reported by: smw1218
	  ........

2009-07-22 21:43 +0000 [r208113]  Jason Parker <jparker@digium.com>

	* apps/app_festival.c: Restore an int declaration on PPC platforms.
	  This x is one crafty little bugger... It was used for 2 different
	  things (one of which was only done on PPC) in 1.4. One of the
	  uses were removed in trunk, and with it went the declaration.
	  (closes issue #14038) Reported by: ffloimair

2009-07-22 16:49 +0000 [r208052]  Tilghman Lesher <tlesher@digium.com>

	* res/res_realtime.c: Clarify documentation on 'realtime update2'
	  to show more than one condition. (closes issue #15357) Reported
	  by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy
	  (license 35) (slightly modified by me)

2009-07-22 14:35 +0000 [r208018]  Russell Bryant <russell@digium.com>

	* include/asterisk/channel.h: Remove trailing whitespace.

2009-07-22 14:35 +0000 [r208017]  Mark Michelson <mmichelson@digium.com>

	* apps/app_directed_pickup.c: Fix the crash in directed pickups.
	  For real this time. A shallow pointer copy was causing an
	  ast_party_connected_line structure to be freed multiple times,
	  thus causing a crash. (closes issue #15441) Reported by:
	  lmsteffan Patches: 15441.patch uploaded by mmichelson (license
	  60) Tested by: lmsteffan

2009-07-21 22:51 +0000 [r207950]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.c: Do not dial digits when none were specified
	  for sig_pri based calls (closes issue #15524) Reported by:
	  elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero
	  (license 37)

2009-07-21 22:45 +0000 [r207946]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 207945 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21
	  Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE
	  (because the documentation states the argument is not optional).
	  This change makes URIENCODE and QUOTE behave similarly, since the
	  documentation states that the argument is not optional, for both.
	  (closes issue #15439) Reported by: pkempgen Patches:
	  20090706__issue15439.diff.txt uploaded by tilghman (license 14)
	  ........

2009-07-21 22:24 +0000 [r207934]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: whitespace fix only

2009-07-21 22:22 +0000 [r207925]  Russell Bryant <russell@digium.com>

	* doc/CODING-GUIDELINES: Note that we use tabs instead of spaces
	  for indentation. I'm surprised this was never actually in here...

2009-07-21 22:02 +0000 [r207854-207902]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Fix my_is_off_hook to check rxbits only
	  for FXS signaling

	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
	  207827 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
	  | 9 lines Wait for wink before dialing when using E&M wink
	  signaling There was already code for other signaling types in
	  dahdi_handle_event to handle dialing if a dial operation dial
	  string was present. Simply add SIG_EMWINK to the list. (closes
	  issue #14434) Reported by: araasch ........

2009-07-21 14:29 +0000 [r207723]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 207714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
	  2009) | 5 lines Document default timeout for AMI originations.
	  AST-224 ........

2009-07-21 13:28 +0000 [r207680]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
	  res/Makefile, pbx/Makefile, Makefile.rules, channels/Makefile,
	  doc/video_console.txt, Makefile, utils/Makefile, codecs/Makefile,
	  agi/Makefile, addons/Makefile, funcs/Makefile,
	  codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions
	  207647 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
	  2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
	  honored. This commit changes the build system so that
	  user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
	  the compiler/linker *after* all flags provided by the build
	  system itself, so that the user can effectively override the
	  build system's flags if desired. In addition, ASTCFLAGS and
	  ASTLDFLAGS can now be provided *either* in the environment before
	  running 'make', or as variable assignments on the 'make' command
	  line. As a result, the use of COPTS and LDOPTS is no longer
	  necessary, so they are no longer documented, but are still
	  supported so as not to break existing build systems that supply
	  them when building Asterisk. ........

2009-07-20 23:08 +0000 [r207522-207551]  Mark Michelson <mmichelson@digium.com>

	* apps/app_directed_pickup.c: Okay, that didn't fix the crash. It
	  didn't really do anything useful.

	* apps/app_directed_pickup.c: Initialize connected line instance
	  when doing a directed pickup. This helps to prevent a crash which
	  may occur due to our freeing garbage due to a struct being
	  uninitialized.

2009-07-20 20:45 +0000 [r207484]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: reg->username is parsed only once on sip
	  reload The registration string can contain an expanded user
	  portion of the form user@domain. This expanded user portion was
	  stored in reg->username and parsed each time there is a
	  registration refresh. Now, the domain portion of the user is
	  parsed and stored separately in the regdomain field. (closes
	  issue #14331) Reported by: Nick_Lewis Patches:
	  chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
	  Tested by: Nick_Lewis, dvossel

2009-07-20 19:48 +0000 [r207424]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 207423 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
	  2009) | 33 lines Answer video SDP offers properly when
	  videosupport is not enabled. Copied from Review board: In issue
	  12434, the reporter describes a situation in which audio and
	  video is offered on the call, but because videosupport is
	  disabled in sip.conf, Asterisk gives no response at all to the
	  video offer. According to RFC 3264, all media offers should have
	  a corresponding answer. For offers we do not intend to actually
	  reply to with meaningful values, we should still reply with the
	  port for the media stream set to 0. In this patch, we take note
	  of what types of media have been offered and save the information
	  on the sip_pvt. The SDP in the response will take into account
	  whether media was offered. If we are not otherwise going to
	  answer a media offer, we will insert an appropriate m= line with
	  the port set to 0. It is important to note that this patch is
	  pretty much a bandage being applied to a broken bone. The patch
	  *only* helps for situations where video is offered but
	  videosupport is disabled and when udptl_pt is disabled but T.38
	  is offered. Asterisk is not guaranteed to respond to every media
	  offer. Notable cases are when multiple streams of the same type
	  are offered. The 2 media stream limit is still present with this
	  patch, too. In trunk and the 1.6.X branches, things will be a bit
	  different since Asterisk also supports text in SDPs as well.
	  (closes issue #12434) Reported by: mnnojd Review:
	  https://reviewboard.asterisk.org/r/311 Review:
	  https://reviewboard.asterisk.org/r/313 ........

2009-07-20 16:36 +0000 [r207361]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 207360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
	  | 9 lines Only do the chan->fdno check in ast_read() in a
	  developer build. I changed this check to only happen in a
	  dev-mode build. I also added a comment explaining what is going
	  on. I also made it so that detection of this situation does not
	  affect ast_read() operation. (closes issue #14723) Reported by:
	  seadweller ........

2009-07-18 04:17 +0000 [r207318]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, CHANGES: Merged 207316 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
	  .......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri,
	  17 Jul 2009) | 20 lines Fixed incoming calls being matched to
	  MSNs without type-of-number prefix added. For an incoming ISDN
	  call the dialed.number is incorrectly matched against the
	  configured MSNs in misdn.conf. The numbers passed to the dialplan
	  include the configured prefix for the dialed.number_type, whereas
	  the check against the configured MSNs (to decide if the call is
	  accepted at all), is executed without the configured prefix.
	  e.g., dialed.number = 241168020, TON = national, configured
	  national prefix is "0". (This is the TON which is used by ISDN
	  providers in the Netherlands.) In chan_misdn.c:cb_events() in
	  case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the
	  unnormalized number 241168020, but 57 lines later the call to
	  read_config() adds the prefix, and the dialed.number is now
	  0241168020, which is then used in the dialplan.
	  misdn_cfg_is_msn_valid() must use the normalized number, too.
	  JIRA ABE-1912

2009-07-18 04:16 +0000 [r207317]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Flag field in wrong position. Reported by
	  "Hoggins!" on asterisk-dev list.

2009-07-18 01:31 +0000 [r207285]  Richard Mudgett <rmudgett@digium.com>

	* /: Recorded merge of revisions 145293,158010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
	  (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
	  channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
	  to make merging easier later. ........ r145200 | rmudgett |
	  2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
	  Miscellaneous formatting changes to make v1.4 and trunk more
	  merge compatible in the mISDN area. channels/chan_misdn.c *
	  Eliminated redundant code in cb_events() EVENT_SETUP ........
	  r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
	  | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
	  rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
	  Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
	  16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
	  channels/chan_misdn.c * Made bearer2str() use
	  allowed_bearers_array[] * Made use the causes.h defines instead
	  of hardcoded numbers. * Made use Asterisk presentation indicator
	  values if either of the mISDN presentation or screen options are
	  negative. * Updated the misdn_set_opt application option
	  descriptions. * Renamed the awkward Caller ID presentation
	  misdn_set_opt application option value not_screened to
	  restricted. Deprecated the not_screened option value.
	  channels/misdn/isdn_lib.c * Made use the causes.h defines instead
	  of hardcoded numbers. * Fixed some spelling errors and typos. *
	  Added all defined facility code strings to fac2str().
	  channels/misdn/isdn_lib.h * Added doxygen comments to struct
	  misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
	  comments to struct misdn_stack. channels/misdn_config.c
	  configs/misdn.conf.sample * Updated the mISDN presentation and
	  screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
	  * Updated the misdn_set_opt application option descriptions. *
	  Fixed some spelling errors and typos. ................ r158010 |
	  rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
	  Merged revision 157977 from
	  https://origsvn.digium.com/svn/asterisk/team/group/issue8824
	  ........ Fixes JIRA ABE-1726 The dial extension could be empty if
	  you are using MISDN_KEYPAD to control ISDN provider features.
	  ................

2009-07-17 22:29 +0000 [r207255]  Tilghman Lesher <tlesher@digium.com>

	* doc/voicemail_odbc_postgresql.txt: Add flag here, too (as
	  requested by jsmith)

2009-07-17 22:07 +0000 [r207225]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: fixes an error in r203638 CEL commit
	  (closes issue #15525) Reported by: elguero Patches:
	  iax2-double-unlock.patch uploaded by elguero (license 37)
	  15525.diff uploaded by dvossel (license 671) Tested by: dvossel

2009-07-17 22:04 +0000 [r207224]  Tilghman Lesher <tlesher@digium.com>

	* doc/tex/odbcstorage.tex, UPGRADE.txt: Document the "flag" field
	  in the voicemessages table.

2009-07-17 19:37 +0000 [r207095-207156]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 207155 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17
	  Jul 2009) | 7 lines Fix format specifier to print out an unsigned
	  long long. Yep, it's even ifdefed out code. But it made it to the
	  RR list... (closes issue #14726) Reported by: lmadsen ........

	* configs/chan_dahdi.conf.sample: Update some missing allowed
	  options for overlapdial

2009-07-17 17:51 +0000 [r207029]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: sip option flags handled incorrectly (closes
	  issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel,
	  Takehiko_Ooshima

2009-07-17 17:02 +0000 [r206998]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c: Fix segfault in
	  sig_analog when using callwaiting, respect callwaiting options
	  Sig_analog handles allocating the sub channel for callwaiting, so
	  no longer try to do it in chan_dahdi. Modified analog_alloc_sub
	  to only mark the sub as allocated upon success of the alloc_sub
	  callback, which was responsible for the segfault. Also, the
	  callwaiting and callwaitingcallerid options were being
	  unconditionally set to true. Now, the options are properly set
	  from chan_dahdi.conf. (closes issue #15508) Reported by: elguero
	  Tested by: elguero

2009-07-17 16:13 +0000 [r206868-206939]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 206938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
	  | 14 lines SIP incorrect From: header information when callpres
	  is prohib Some ITSP make use of the "Anonymous" display name to
	  detect a requirement to withhold caller id across the PSTN. This
	  does not work if the display name is "Unknown". (closes issue
	  #14465) Reported by: Nick_Lewis Patches:
	  chan_sip.c-callerpres.patch uploaded by Nick (license 657)
	  chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
	  671) Tested by: Nick_Lewis, dvossel ........

	* funcs/func_timeout.c: TIMEOUT(absolute) returned negative value.
	  (closes issue #15513) Reported by: ys

	* configs/iax.conf.sample, /: Merged revisions 206872 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16
	  Jul 2009) | 6 lines error in iax.conf related IP-based access
	  control (closes issue #15518) Reported by: pkempgen ........

	* /, main/callerid.c: Merged revisions 206867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
	  | 8 lines avoid segfault caused by user error If the CALLERPRES()
	  dialplan function is set to nothing, a segfault occurs. This is
	  user error to begin with, but I'd rather see a cli warning
	  message than have Asterisk crash on me. ........

2009-07-16 16:51 +0000 [r206808]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_realtime.c: Merged revisions 206807 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16
	  Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517)
	  Reported by: adomjan Patches:
	  func_realtime.c-ast_variable_destroy.diff uploaded by adomjan
	  (license 487) ........

2009-07-15 22:04 +0000 [r206768]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Session timer were not activated if
	  Supported header field in INVITE had both "timer" and other
	  options. (closes issue #15403) Reported by: makoto Patches:
	  sip-session-timer.patch uploaded by makoto (license 38)

2009-07-15 22:02 +0000 [r206767]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h, channels/sig_pri.c: The dialing flag was
	  mistakingly removed from sig_pri. This readds the proper setting
	  of the flag and is really a continuation of r205731. The flag was
	  being set properly in sig_analog, but use of the newly added
	  set_dialing callback allowed for some simplification in
	  chan_dahdi. (closes issue #15486) Reported by: rmudgett

2009-07-15 21:14 +0000 [r206707]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
	  Merged revisions 206706 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
	  (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
	  https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
	  .......... Fixed chan_misdn crash because mISDNuser library is
	  not thread safe. With Asterisk the mISDNuser library is driven by
	  two threads concurrently: 1.
	  channels/misdn/isdn_lib.c::manager_event_handler() 2.
	  channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
	  into the library are done concurrently and recursively from
	  isdn_lib.c. Both threads can fiddle with the master/child
	  layer3_proc_t lists. One thread may traverse the list when the
	  other interrupts it and then removes the list element which the
	  first thread was currently handling. This is exactly what caused
	  the crash. About 60 calls were needed to a Gigaset CX475 before
	  it occurred once. This patch adds locking when calling into the
	  mISDNuser library. This also fixes some cb_log calls with wrong
	  port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
	  (Modified with mostly cosmetic changes) ..........
	  ................

2009-07-15 20:20 +0000 [r206702]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: callerid(num) is wrong when username is
	  missing A domain only sip uri <sip:123.123.123.123> would return
	  123.123.123.123 as callid num. Now, if the username is missing
	  from a uri, the callerid num field is left empty. (closes issue
	  #15476) Reported by: viraptor

2009-07-15 16:00 +0000 [r206636]  Sean Bright <sean@malleable.com>

	* /, codecs/codec_dahdi.c: Merged revisions 206635 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
	  15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
	  are asking for it. ........

2009-07-14 20:38 +0000 [r206603]  Jeff Peeler <jpeeler@digium.com>

	* configs/chan_dahdi.conf.sample: fix a typo in sample config file
	  for option change

2009-07-14 20:14 +0000 [r206567]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_meetme.c, contrib/scripts/meetme.sql: Document all
	  meetme realtime fields, and in the process, make some field
	  lengths more consistent. (closes issue #15493) Reported by: lasko
	  Patches: meetme.diff uploaded by lasko (license 833)

2009-07-14 20:01 +0000 [r206566]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/sig_analog.h: Restore some missing functionality to
	  sig_analog. The main purpose of this commit is to restore missing
	  functionality present in the ss_thread before all the sig related
	  work was done. Two of the biggest missing things were distinctive
	  ring detection and cid handling for V23. fxsoffhookstate and
	  associated mwi variables have been moved inside sig_analog as
	  they were not being set properly as well.

2009-07-14 17:03 +0000 [r206490]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: I AM A TERRIBLE PERSON

2009-07-14 17:01 +0000 [r206489]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/isdn_lib.c: Merged revisions 206487 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14
	  Jul 2009) | 28 lines Fixes several call transfer issues with
	  chan_misdn. * issue #14355 - Crash if attempt to transfer a call
	  to an application. Masquerade the other pair of the four asterisk
	  channels involved in the two calls. The held call already must be
	  a bridged call (not an applicaton) or it would have been
	  rejected. * issue #14692 - Held calls are not automatically
	  cleared after transfer. Allow the core to initate disconnect of
	  held calls to the ISDN port. This also fixes a similar case where
	  the party on hold hangs up before being transferred or taken off
	  hold. * JIRA ABE-1903 - Orphaned held calls left in
	  music-on-hold. Do not simply block passing the hangup event on
	  held calls to asterisk core. * Fixed to allow held calls to be
	  transferred to ringing calls. Previously, held calls could only
	  be transferred to connected calls. * Eliminated unused call
	  states to simplify hangup code. * Eliminated most uses of
	  "holded" because it is not a word. (closes issue #14355) (closes
	  issue #14692) Reported by: sodom Patches:
	  misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
	  Tested by: rmudgett ........

2009-07-14 16:09 +0000 [r206455]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Reset the sentringing indication when redirects
	  occur. If a redirecting control frame is processed or a call
	  forward occurs, we need to reset the sentringing flag so that we
	  can send another ringing indication to the phone that may contain
	  a connected line update. AST-164

2009-07-14 14:51 +0000 [r206386]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 206385 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r206385 | russell | 2009-07-14 09:48:00 -0500
	  (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
	  | 6 lines Ensure apathetic replies are sent out on the proper
	  socket. chan_iax2 supports multiple address bindings. The
	  send_apathetic_reply() function did not attempt to send its
	  response on the same socket that the incoming message came in on.
	  ........ ................

2009-07-14 00:48 +0000 [r206341]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 206284 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
	  | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
	  ........

2009-07-13 23:26 +0000 [r206280]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: dns lookup of peername rather than peer's
	  host in transmit_register() (closes issue #15052) Reported by:
	  fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch
	  uploaded by fsantulli (license 818) Tested by: fsantulli

2009-07-13 18:46 +0000 [r206225]  Sean Bright <sean@malleable.com>

	* contrib/upstart/asterisk.upstart-0.3.9: Make sure that since we
	  are passing -c to asterisk that we have a console. Without this
	  line, Asterisk will busy-loop trying to read and write to
	  /dev/null (woops... my bad).

2009-07-13 16:23 +0000 [r206185]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Remove reference to non-existent help file
	  (closes issue #15427) Reported by: brushtyler Patches:
	  app_voicemail.c.diff uploaded by brushtyler (license 821)

2009-07-13 14:06 +0000 [r206092-206094]  Kevin P. Fleming <kpfleming@digium.com>

	* .cleancount: Bump up cleancount so that existing checkouts will
	  update themselves properly for the 'Addons' -> 'ADDONS' change.

	* addons/Makefile: Make the menuselect category for Add-Ons
	  consistent with the other directories (uppercase).

2009-07-11 19:30 +0000 [r206021-206049]  Russell Bryant <russell@digium.com>

	* CHANGES: note the security events API in CHANGES

	* doc/tex/security-events.tex (added), tests/test_security_events.c
	  (added), main/manager.c, main/security_events.c (added),
	  include/asterisk/event_defs.h, main/event.c,
	  include/asterisk/security_events.h (added), doc/tex/asterisk.tex,
	  include/asterisk/security_events_defs.h (added),
	  res/res_security_log.c (added), tests/test_ami_security_events.sh
	  (added): Add an API for reporting security events, and a security
	  event logging module. This commit introduces the security events
	  API. This API is to be used by Asterisk components to report
	  events that have security implications. A simple example is when
	  a connection is made but fails authentication. These events can
	  be used by external tools manipulate firewall rules or something
	  similar after detecting unusual activity based on security
	  events. Inside of Asterisk, the events go through the ast_event
	  API. This means that they have a binary encoding, and it is easy
	  to write code to subscribe to these events and do something with
	  them. One module is provided that is a subscriber to these events
	  - res_security_log. This module turns security events into a
	  parseable text format and sends them to the "security" logger
	  level. Using logger.conf, these log entries may be sent to a
	  file, or to syslog. One service, AMI, has been fully updated for
	  reporting security events. AMI was chosen as it was a fairly
	  straight forward service to convert. The next target will be
	  chan_sip. That will be more complicated and will be done as its
	  own project as the next phase of security events work. For more
	  information on the security events framework, see the
	  documentation generated from doc/tex/. "make asterisk.pdf"
	  Review: https://reviewboard.asterisk.org/r/273/

2009-07-10 21:42 +0000 [r205985]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP register not using peer's outbound proxy
	  If callbackextension is defined for a peer it successfully causes
	  a registration to occur, but the registration ignores the
	  outboundproxy settings for the peer. This patch allows the peer
	  to be passed to obproxy_get() in transmit_register(). (closes
	  issue #14344) Reported by: Nick_Lewis Patches:
	  callbackextension_peer_trunk.diff uploaded by dvossel (license
	  671) Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/294/

2009-07-10 18:44 +0000 [r205939]  Kevin P. Fleming <kpfleming@digium.com>

	* main/udptl.c: Update comments about the level of T.38 support in
	  Asterisk.

2009-07-10 17:39 +0000 [r205878]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 205877 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
	  (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
	  (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
	  2009) | 10 lines Ensure that outbound NOTIFY requests are
	  properly routed through stateful proxies. With this change, we
	  make note of Record-Route headers present in any SUBSCRIBE
	  request that we receive so that our outbound NOTIFY requests will
	  have the proper Route headers in them. (closes issue #14725)
	  Reported by: ibc ........ ................ ................

2009-07-10 16:42 +0000 [r205840]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 205804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
	  | 31 lines SIP registration auth loop caused by stale nonce If an
	  endpoint sends two registration requests in a very short period
	  of time with the same nonce, both receive 401 responses from
	  Asterisk, each with a different nonce (the second 401 containing
	  the current nonce and the first one being stale). If the endpoint
	  responds to the first 401, it does not match the current nonce so
	  Asterisk sends a third 401 with a newly generated nonce (which
	  updates the current nonce)... Now if the endpoint responds to the
	  second 401, it does not match the current nonce either and
	  Asterisk sends a fourth 401 with a newly generated nonce... This
	  loop goes on and on. There appears to be a simple fix for this.
	  If the nonce from the request does not match our nonce, but is a
	  good response to a previous nonce, instead of sending a 401 with
	  a newly generated nonce, use the current one instead. This breaks
	  the loop as the nonce is not updated until a response is
	  received. Additional logic has been added to make sure no nonce
	  can be responded to twice though. (closes issue #15102) Reported
	  by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
	  809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
	  Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........

2009-07-10 16:00 +0000 [r205780]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Eliminate extraneous LOG_DEBUG messages generated
	  by app_fax. The transmit_audio() and transmit_t38() functions in
	  app_fax have processing loops that are supposed to wait for
	  frames to arrive on the channel and then handle them, but they
	  also have short timeouts so that the loops can have watchdog
	  timers and do other required processing. This commit changes the
	  loops to not actually call ast_read() and attempt to process the
	  returned frame unless a frame actually arrived, eliminating
	  hundreds of LOG_DEBUG messages and slightly improving
	  performance.

2009-07-10 15:56 +0000 [r205776]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 205775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
	  2009) | 10 lines Ensure that outbound NOTIFY requests are
	  properly routed through stateful proxies. With this change, we
	  make note of Record-Route headers present in any SUBSCRIBE
	  request that we receive so that our outbound NOTIFY requests will
	  have the proper Route headers in them. (closes issue #14725)
	  Reported by: ibc ........

2009-07-10 15:28 +0000 [r205770]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_fax.c: Fix some remaining T.38 negotiation problems in
	  app_fax. Revision 205696 did not quite fix all the issues with
	  the T.38 negotiation changes and app_fax; this patch corrects
	  them, along with a couple of other minor issues. (closes issue
	  #15480) Reported by: dimas Patches: test2-15480.patch uploaded by
	  dimas (license 88)

2009-07-09 21:32 +0000 [r205700]  Matthew Nicholson <mnicholson@digium.com>

	* addons/chan_mobile.c: Fix mbl_fixup() in chan_mobile to update
	  newchan->tech_pvt instead of oldchan. (closes issue #15299)
	  Reported by: nikkk

2009-07-09 21:20 +0000 [r205696]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h:
	  Repair ability of SendFAX/ReceiveFAX to respond to T.38
	  switchover. Recent changes in T.38 negotiation in Asterisk caused
	  these applications to not respond when the other endpoint
	  initiated a switchover to T.38; this resulted in the T.38
	  switchover failing, and the FAX attempt to be made using an audio
	  connection, instead of T.38 (which would usually cause the FAX to
	  fail completely). This patch corrects this problem, and the
	  applications will now correctly respond to the T.38 switchover
	  request. In addition, the response will include the appopriate
	  T.38 session parameters based on what the other end offered and
	  what our end is capable of. (closes issue #14849) Reported by:
	  afosorio

2009-07-09 20:04 +0000 [r205666]  Matthew Nicholson <mnicholson@digium.com>

	* funcs/func_odbc.c: Convert func_odbc to use
	  ast_dummy_alloc_channel() Review:
	  https://reviewboard.asterisk.org/r/290/

2009-07-09 16:19 +0000 [r205600]  David Vossel <dvossel@digium.com>

	* /, include/asterisk/time.h: Merged revisions 205599 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
	  Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
	  point. ........

2009-07-09 14:10 +0000 [r205532-205562]  Michiel van Baak <michiel@vanbaak.info>

	* main/cel.c: make this compile again under devmode

	* main/ssl.c: pthread_self returns a pthread_t which is not an
	  unsigned int on all pthread implementations. Casting it to an
	  unsigned int fixes compiler warnings. Tested on OpenBSD and Linux
	  both 32 and 64 bit

2009-07-08 23:19 +0000 [r205479]  David Vossel <dvossel@digium.com>

	* res/res_rtp_asterisk.c, /, channels/chan_iax2.c,
	  include/asterisk/frame.h: Merged revisions 205471 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08
	  Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations
	  assume 8khz is the codec rate. This is not always the case. This
	  patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am
	  sure there are other areas that make this assumption as well.
	  Review: https://reviewboard.asterisk.org/r/306/ ........

2009-07-08 23:07 +0000 [r205469]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c: Fix a CEL related regression with hints updating by
	  subscribing to AST_DEVICE_STATE instead of
	  AST_DEVICE_STATE_CHANGED. (closes issue #15440) Reported by:
	  lmsteffan

2009-07-08 22:15 +0000 [r205410-205412]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, /,
	  main/devicestate.c, include/asterisk/pbx.h: Merged revisions
	  205409 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
	  | 6 lines moving ast_devstate_to_extenstate to pbx.c from
	  devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
	  change fixes a compile time error with chan_vpb as well. ........

	* main/devicestate.c: missing comma in devstatestring array

2009-07-08 19:26 +0000 [r205350]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 205349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
	  2009) | 14 lines Prevent phantom calls to queue members. If a
	  caller were to hang up while a periodic announcement or position
	  were being said, the return value for those functions would
	  incorrectly indicate that the caller was still in the queue. With
	  these changes, the problem does not occur. (closes issue #14631)
	  Reported by: latinsud Patches: queue_announce_ghost_call2.diff
	  uploaded by latinsud (license 745) (with small modification from
	  me) ........

2009-07-08 18:19 +0000 [r205291]  Jason Parker <jparker@digium.com>

	* config.sub, /, config.guess: Merged revisions 205288 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
	  2009) | 1 line Update config.guess and config.sub from the
	  savannah.gnu.org git repo. ........

2009-07-08 17:26 +0000 [r205254]  David Brooks <dbrooks@digium.com>

	* main/features.c: Fixes Park() argument handling Park() was not
	  respecting the arguments passed to it. Any
	  extension/context/priority given to it was being ignored. This
	  patch remedies this. (closes issue #15380) Reported by: DLNoah

2009-07-08 16:59 +0000 [r205221]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Oops, fixing build

2009-07-08 16:54 +0000 [r205216]  David Vossel <dvossel@digium.com>

	* /, include/asterisk/time.h: Merged revisions 205215 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08
	  Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz
	  audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is
	  16000. The .5 is currently stripped off because we don't
	  calculate using floating points. This causes madness with 16khz
	  audio. (issue ABE-1899) Review:
	  https://reviewboard.asterisk.org/r/305/ ........

2009-07-08 16:43 +0000 [r205214]  Sean Bright <sean@malleable.com>

	* utils/muted.c, configure, include/asterisk/autoconfig.h.in,
	  configure.ac, main/dns.c: Fix a few compilation problems found
	  when building Asterisk against uClibc.

2009-07-08 16:27 +0000 [r205196]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c: Merged revisions 205188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
	  | 2 lines Add redirection warnings for the invalid language codes
	  previously removed. ........

2009-07-08 15:56 +0000 [r205120-205151]  Russell Bryant <russell@digium.com>

	* main/ssl.c: Use tabs instead of spaces for indentation.

	* res/res_crypto.c, main/ssl.c (added),
	  include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c:
	  Move OpenSSL initialization to a single place, make library usage
	  thread-safe. While doing some reading about OpenSSL, I noticed a
	  couple of things that needed to be improved with our usage of
	  OpenSSL. 1) We had initialization of the library done in multiple
	  modules. This has now been moved to a core function that gets
	  executed during Asterisk startup. We already link OpenSSL into
	  the core for TCP/TLS functionality, so this was the most logical
	  place to do it. 2) OpenSSL is not thread-safe by default.
	  However, making it thread safe is very easy. We just have to
	  provide a couple of callbacks. One callback returns a thread ID.
	  The other handles locking. For more information, start with the
	  "Is OpenSSL thread-safe?" question on the FAQ page of
	  openssl.org.

2009-07-08 14:45 +0000 [r205118]  Luigi Rizzo <rizzo@icir.org>

	* bootstrap.sh: FreeBSD now has autoconf 2.62 in the ports, 2.61
	  has disappeared.

2009-07-07 21:10 +0000 [r205086]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Permit setting custom headers from the peer
	  definition. (closes issue #14059) Reported by: fnordian

2009-07-07 18:24 +0000 [r205014-205047]  Matthew Nicholson <mnicholson@digium.com>

	* channels/sig_analog.c: Fix a deadlock in sig_analog

	* channels/sig_analog.c: Add CEL transfer events to analog
	  (chan_dahdi) transfers.

2009-07-06 21:37 +0000 [r204986]  Tilghman Lesher <tlesher@digium.com>

	* addons/res_config_mysql.c: Merged revisions 981 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk-addons/branches/1.4
	  ........ r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul
	  2009) | 7 lines Don't reset reconnect time, unless a reconnect
	  really occurred. (closes issue #15375) Reported by: kowalma
	  Patches: 20090628__issue15375.diff.txt uploaded by tilghman
	  (license 14) Tested by: kowalma, jacco ........

2009-07-06 13:38 +0000 [r204948]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c: Improve handling of AST_CONTROL_T38 and
	  AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
	  change allows applications that request T.38 negotiation on a
	  channel that does not support it to get the proper indication
	  that it is not supported, rather than thinking that negotiation
	  was started when it was not.

2009-07-03 15:44 +0000 [r204893-204919]  Sean Bright <sean@malleable.com>

	* channels/sig_pri.h, channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  channels/sig_pri.c: Add a configure check for Reverse Charging
	  Indication support in LibPRI. Also go back and wrap all of the
	  places that use the specific reverse charge APIs with
	  preprocessor conditionals.

	* include/asterisk/rtp_engine.h: Wrap rtp_engine.h header comments
	  to 80 characters.

2009-07-02 22:01 +0000 [r204835]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 204834 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02
	  Jul 2009) | 10 lines Removed confusing warning message "Got Busy
	  in Connected State" If an incoming mISDN call is answered with
	  the Answer application and a subsequent Dial gets a busy endpoint
	  then it is valid for that already connected channel to get the
	  busy indication. Asterisk will play the busy tones until the
	  dialplan plays something else or hangs up the call. (closes issue
	  #11974) Reported by: fvdb ........

2009-07-02 20:37 +0000 [r204807]  Matthew Nicholson <mnicholson@digium.com>

	* main/channel.c, main/features.c: Moved trigger for BRIDGE_END CEL
	  event so that it is more accurate.

2009-07-02 17:46 +0000 [r204749]  Sean Bright <sean@malleable.com>

	* channels/sig_pri.h, channels/chan_dahdi.c,
	  configs/chan_dahdi.conf.sample, funcs/func_channel.c, CHANGES,
	  channels/sig_pri.c: Support setting and receiving Reverse
	  Charging Indication over ISDN PRI. This is a continuation of
	  revision 885 to LibPRI (Capture and expose the Reverse Charging
	  Indication IE on ISDN PRI) which added the ability to get/set
	  Reverse Charging Indication in LibPRI. This patch adds the
	  ability to specify RCI on the outbound leg of a PRI call from
	  within Asterisk, by prefixing the dialed number with a capital
	  'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an
	  inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
	  Thanks again to rmudgett for the thorough review. (closes issue
	  #13760) Reported by: mrgabu Review:
	  https://reviewboard.asterisk.org/r/303/

2009-07-02 16:03 +0000 [r204710]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, /,
	  main/devicestate.c: Merged revisions 204681 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
	  | 14 lines Improved mapping of extension states from combined
	  device states. This fixes a few issues with incorrect extension
	  states and adds a cli command, core show device2extenstate, to
	  display all possible state mappings. (closes issue #15413)
	  Reported by: legart Patches: exten_helper.diff uploaded by
	  dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
	  https://reviewboard.asterisk.org/r/301/ ........

2009-07-01 19:47 +0000 [r204654]  Ryan Brindley <rbrindley@digium.com>

	* configs/http.conf.sample: - cfgbasic.html has been replaced by
	  index.html in the GUI for some time now

2009-07-01 16:06 +0000 [r204622]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: A bunch of CODING_GUIDELINES related fixes.
	  Not even close to done.

2009-06-30 20:41 +0000 [r204563]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c, UPGRADE.txt: Merged revisions 204556 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30
	  Jun 2009) | 6 lines More incorrect language codes, plus ensuring
	  that regionalizations use the specified language, and not English
	  for grammar. (closes issue #15022) Reported by: greenfieldtech
	  Patches: 20090519__issue15022.diff.txt uploaded by tilghman
	  (license 14) ........

2009-06-30 20:39 +0000 [r204561]  Sean Bright <sean@malleable.com>

	* apps/app_voicemail.c: Remove an unnecessary #ifdef

2009-06-30 19:59 +0000 [r204530-204532]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Move the masquerade in
	  local_attended_transfer to a point where we hold the channel
	  lock. Masquerading without the channel's lock held is a
	  *horrible* idea.

	* channels/chan_sip.c: Remove some bogus deadlock avoidance code
	  from local_attended_transfer. First of all, the code was
	  unnecessary. The goal was to lock a channel which was already
	  locked. Second, the assumption of the deadlock avoidance loop was
	  that the sip_pvt was already locked and we were trying to get the
	  channel lock. The problem is that the sip_pvt was unlocked a few
	  lines above. Basically, I'm removing 5 lines of no-op.

2009-06-30 18:48 +0000 [r204475]  Jason Parker <jparker@digium.com>

	* /, main/say.c: Merged revisions 204474 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
	  1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
	  comment typo in passing. ........

2009-06-30 18:36 +0000 [r204470]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c, UPGRADE.txt, apps/app_voicemail.c: Recorded merge
	  of revisions 204469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
	  | 11 lines "tw" is the language specification for Twi (from
	  Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
	  Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
	  (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
	  tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
	  uploaded by tilghman (license 14)
	  20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
	  (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
	  tilghman (license 14) Tested by: volivier ........

2009-06-30 17:22 +0000 [r204417-204440]  Russell Bryant <russell@digium.com>

	* configs/res_config_sqlite.conf (removed),
	  configs/res_config_sqlite.conf.sample (added): Rename
	  res_config_sqlite.conf to res_config_sqlite.conf.sample (missing
	  .sample).

	* addons/chan_ooh323.c, configs/chan_ooh323.conf.sample (added),
	  configs/ooh323.conf.sample (removed): Rename ooh323.conf to
	  chan_ooh323.conf, make module support both names

	* configs/mobile.conf.sample (removed), addons/chan_mobile.c,
	  configs/chan_mobile.conf.sample (added): Rename mobile.conf to
	  chan_mobile.conf, make module support old name, too

	* configs/res_config_mysql.conf.sample (added),
	  configs/res_mysql.conf.sample (removed),
	  addons/res_config_mysql.c: Rename res_mysql.conf to
	  res_config_mysql.conf, make module support both

	* Makefile: Make addons build last - this is for Qwell.

	* addons/app_mysql.c, configs/app_mysql.conf.sample (added),
	  configs/mysql.conf.sample (removed): Rename mysql.conf to
	  app_mysql.conf, make module support both names

	* addons/Makefile, addons/cdr_mysql.c (added),
	  addons/cdr_addon_mysql.c (removed): Rename cdr_addon_mysql to
	  cdr_mysql

	* addons/app_mysql.c (added), addons/app_addon_sql_mysql.c
	  (removed), addons/Makefile: Rename app_addon_sql_mysql to
	  app_mysql

2009-06-30 17:04 +0000 [r204415]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/embed_modules.xml, Makefile.moddir_rules,
	  addons/Makefile: Add-ons related build system improvements.
	  Ensure that add-on modules can be embedded, fix up
	  Makefile.moddir_rules to allow module directory Makefiles to more
	  easily specify the modules to be built, and explicitly list the
	  addons modules in its Makefile, since the module names don't
	  follow any pattern.

2009-06-30 16:40 +0000 [r204413]  Russell Bryant <russell@digium.com>

	* autoconf/ast_ext_tool_check.m4, addons/ooh323c/src/oochannels.h,
	  addons/ooh323c/src/printHandler.h, addons/chan_ooh323.c,
	  addons/ooh323c/src/ooq931.h, include/asterisk/autoconfig.h.in,
	  addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
	  addons/ooh323c/src/ooasn1.h, configs/res_mysql.conf.sample
	  (added), addons/ooh323c/src/ooStackCmds.c,
	  addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooStackCmds.h,
	  addons/ooh323c/src/eventHandler.c,
	  addons/ooh323c/src/h323/H235-SECURITY-MESSAGES.h,
	  addons/mp3/huffman.h, configure,
	  addons/ooh323c/src/eventHandler.h, addons/ooh323cDriver.c,
	  include/asterisk/mod_format.h, addons/mp3/interface.c,
	  doc/tex/asterisk.tex, addons/ooh323cDriver.h,
	  addons/cdr_addon_mysql.c, addons/ooh323c/src/encode.c,
	  addons/mp3/MPGLIB_README,
	  addons/ooh323c/src/h323/H235-SECURITY-MESSAGESEnc.c,
	  configure.ac, doc/tex/chan_mobile.tex (added),
	  addons/ooh323c/src/ooports.c, addons/mp3/mpg123.h,
	  addons/mp3/mpglib.h, addons (added),
	  addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.c,
	  addons/ooh323c/src/ooports.h, addons/ooh323c/src/memheap.c,
	  Makefile, addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.h,
	  addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
	  addons/ooh323c/src/memheap.h, addons/ooh323c/src/perutil.c,
	  addons/mp3/decode_i386.c, addons/ooh323c/src/ooh245.h,
	  addons/mp3/dct64_i386.c, addons/ooh323c/src/ooSocket.c,
	  addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
	  addons/mp3/layer3.c, addons/ooh323c/src/ooper.h,
	  addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooSocket.h,
	  addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooCmdChannel.h,
	  addons/ooh323c/COPYING, addons/format_mp3.c,
	  addons/ooh323c/src/Makefile.in, configs/mobile.conf.sample
	  (added), addons/ooh323c/src/ootypes.h, addons/mp3,
	  addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooTimer.c,
	  addons/ooh323c/src/ooLogChan.h, addons/ooh323c/src/dlist.c,
	  addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/oohdr.h,
	  README-addons.txt (added), addons/app_addon_sql_mysql.c,
	  addons/ooh323c/src/ooTimer.h, addons/ooh323c/src/ooCapability.h,
	  addons/ooh323c/src/dlist.h, addons/mp3/Makefile, addons/Makefile,
	  addons/ooh323c/README, addons/ooh323c, doc/tex/cdrdriver.tex,
	  addons/ooh323c/src/h323/H323-MESSAGESEnc.c, addons/chan_mobile.c,
	  configs/cdr_mysql.conf.sample (added),
	  addons/ooh323c/src/ooDateTime.c, addons/ooh323c/src/rtctype.c,
	  addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooGkClient.c,
	  addons/ooh323c/src/h323, addons/ooh323c/src/ooUtils.c,
	  addons/ooh323c/src/ooDateTime.h,
	  addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLEnc.c,
	  addons/ooh323c/src/rtctype.h, addons/ooh323c/src/ooCalls.h,
	  configs/mysql.conf.sample (added), addons/ooh323c/src/ooh323ep.c,
	  addons/ooh323c/src/ooGkClient.h,
	  addons/ooh323c/src/h323/H323-MESSAGES.c,
	  addons/ooh323c/src/ooUtils.h, addons/mp3/README, UPGRADE.txt,
	  addons/mp3/MPGLIB_TODO, addons/ooh323c/src/ooh323ep.h,
	  addons/ooh323c/src/h323/H323-MESSAGES.h,
	  addons/mp3/decode_ntom.c, configs/ooh323.conf.sample (added),
	  addons/ooh323c/src/ooh323.c,
	  addons/ooh323c/src/h323/H323-MESSAGESDec.c, addons/ooh323c/src,
	  build_tools/menuselect-deps.in, addons/mp3/tabinit.c,
	  addons/ooh323c/src/ooh323.h, doc/tex/Makefile,
	  addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
	  main/file.c,
	  addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
	  makeopts.in, addons/ooh323c/src/oochannels.c,
	  addons/app_saycountpl.c, addons/ooh323c/src/printHandler.c,
	  addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c,
	  addons/res_config_mysql.c: Move Asterisk-addons modules into the
	  main Asterisk source tree. Someone asked yesterday, "is there a
	  good reason why we can't just put these modules in Asterisk?".
	  After a brief discussion, as long as the modules are clearly set
	  aside in their own directory and not enabled by default, it is
	  perfectly fine. For more information about why a module goes in
	  addons, see README-addons.txt. chan_ooh323 does not currently
	  compile as it is behind some trunk API updates. However, it will
	  not build by default, so it should be okay for now.

2009-06-29 23:50 +0000 [r204355]  Sean Bright <sean@malleable.com>

	* apps/app_meetme.c: A few const changes in app_meetme.c that I
	  noticed while browsing the source.

2009-06-29 22:50 +0000 [r204247-204301]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 204300 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
	  2009) | 9 lines Add error message so that it is clear why a SIP
	  peer was not processed when a DNS lookup fails on a host or
	  outboundproxy. (closes issue #13432) Reported by: p_lindheimer
	  Patches: outboundproxy.patch uploaded by p (license 558) ........

	* /, channels/chan_sip.c: Merged revisions 204243,204246 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
	  2009) | 22 lines Fix a problem where chan_sip would ignore "old"
	  but valid responses. chan_sip has had a problem for quite a long
	  time that would manifest when Asterisk would send multiple SIP
	  responses on the same dialog before receiving a response. The
	  problem occurred because chan_sip only kept track of the highest
	  outgoing sequence number used on the dialog. If Asterisk sent two
	  requests out, and a response arrived for the first request sent,
	  then Asterisk would ignore the response. The result was that
	  Asterisk would continue retransmitting the requests and ignoring
	  the responses until the maximum number of retransmissions had
	  been reached. The fix here is to rearrange the code a bit so that
	  instead of simply comparing the sequence number of the response
	  to our latest outgoing sequence number, we walk our list of
	  outstanding packets and determine if there is a match. If there
	  is, we continue. If not, then we ignore the response. In doing
	  this, I found a few completely useless variables that I have now
	  removed. (closes issue #11231) Reported by: flefoll Review:
	  https://reviewboard.asterisk.org/r/298 ........ r204246 |
	  mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
	  lines Fix build oops. ........

2009-06-29 20:29 +0000 [r204119-204217]  Sean Bright <sean@malleable.com>

	* configs/cel_adaptive_odbc.conf.sample: Reorganize this adaptive
	  CEL config a bit.

	* apps/app_rpt.c: Get app_rpt compiling again. I doubt seriously
	  that it actually works. Also, the code in this module is
	  horrendous and we should remove it from the tree. I'm not sure
	  who is supposed to be maintaning this thing, but they clearly are
	  not. I don't see the sense of leaving it in the main tree. If it
	  lives *anywhere* it should be in addons.

	* configs/cel_sqlite3_custom.conf.sample, configs/cel.conf.sample,
	  configs/cel_adaptive_odbc.conf.sample,
	  configs/cel_pgsql.conf.sample, configs/cel_custom.conf.sample:
	  Add common headers to CEL related configs.

2009-06-29 17:56 +0000 [r204069-204118]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, include/asterisk/channel.h: Allow trunk to once
	  again compile under MALLOC_DEBUG

	* configs/cel_adaptive_odbc.conf.sample: Remove invalid entries in
	  the config. This might seem like a legitimate comment that merely
	  needed semicolon prefixes, but in reality, the adaptive layer is
	  designed to allow arbitrary CDR variables, without needing the
	  use of a userfield to store multiple items. It's therefore not
	  only invalid syntax but also goes against the intent of the
	  adaptive method.

2009-06-27 20:26 +0000 [r203985]  Sean Bright <sean@malleable.com>

	* CHANGES: Another CHANGES spelling fix.

2009-06-27 10:04 +0000 [r203960-203962]  Russell Bryant <russell@digium.com>

	* main/app.c: Only update total silence counter after a counter
	  reset. (closes issue #2264) Reported by: pfn Patches:
	  silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by:
	  pfn

	* UPGRADE.txt, CHANGES: Minor tweaks and spelling fixes for CHANGES
	  and UPGRADE.txt.

2009-06-27 01:07 +0000 [r203909]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/sig_pri.c: Merged revisions 203908 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
	  | 16 lines The ISDN CPE side should not exclusively pick B
	  channels normally. Before this patch, Asterisk unconditionally
	  picked B channels exclusively on the CPE side and normally
	  allowed alternative B channels on the network side. Now Asterisk
	  does the opposite. Reasons for the CPE side to normally not pick
	  B channels exclusively: * For CPE point-to-multipoint mode (i.e.
	  phone side), the CPE side does not have enough information to
	  exclusively pick B channels. (There may be other devices on the
	  line.) * Q.931 gives preference to the network side picking B
	  channels. * Some telcos require the CPE side to not pick B
	  channels exclusively. (closes issue #14383) Reported by:
	  mbrancaleoni ........

2009-06-26 22:11 +0000 [r203853]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 203848 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26
	  Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo
	  channel after dahdi restart (closes issue #14477) Reported by:
	  timking ........

2009-06-26 22:08 +0000 [r203846]  Sean Bright <sean@malleable.com>

	* cdr/cdr_syslog.c (added), build_tools/menuselect-deps.in,
	  configure, configure.ac, configs/cdr_syslog.conf.sample (added),
	  CHANGES: Add a new module, cdr_syslog, which allows writing CDRs
	  to syslog. The original patch for this was written by Brett
	  Bryant, and I split it out into it's own module. (closes issue
	  #12876) Reported by: bbryant Patches:
	  06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
	  05212009_cdr_syslog.patch uploaded by seanbright (license 71)
	  Tested by: seanbright Review:
	  https://reviewboard.asterisk.org/r/297/

2009-06-26 21:48 +0000 [r203802-203842]  Russell Bryant <russell@digium.com>

	* CHANGES, apps/app_chanspy.c: Add 's' option to ChanSpy, which
	  makes the app exit when no channels are left to spy on. (closes
	  issue #14594) Reported by: JimDickenson Patches: chanspy.diff
	  uploaded by JimDickenson (license 710)

	* /, main/file.c: Merged revisions 203785 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
	  | 15 lines Don't fast forward past the end of a message. This is
	  nice change for users of the voicemail application. If someone
	  gets a little carried away with fast forwarding through a
	  message, they can easily get to the end and accidentally exit the
	  voicemail application by hitting the fast forward key during the
	  following prompt. This adds some safety by not allowing a fast
	  forward past the end of a message. (closes issue #14554) Reported
	  by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
	  707) Tested by: lacoursj ........

2009-06-26 20:52 +0000 [r203783]  Mark Michelson <mmichelson@digium.com>

	* doc/manager_1_1.txt, main/manager.c: Add timestamp to response to
	  "Ping" manager action. (closes issue #14596) Reported by:
	  JimDickenson Patches: pong2.diff uploaded by JimDickenson
	  (license 710)

2009-06-26 20:45 +0000 [r203779]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Ensure the TCP read buffer is fully
	  initialized before handling each packet. (closes issue #14452)
	  Reported by: umberto71

2009-06-26 20:19 +0000 [r203735]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Fix the
	  'nat' option to actually do RFC3581 as expected and extend the
	  configurable values for finer control. (closes issue #8855)
	  Reported by: mikma Tested by: klaus3000, file

2009-06-26 20:13 +0000 [r203721]  David Brooks <dbrooks@digium.com>

	* apps/app_voicemail.c: Fixing voicemail's error in checking max
	  silence vs min message length Max silence was represented in
	  milliseconds, yet vmminsecs (minmessage) was represented as
	  seconds. Also, the inequality was reversed. The warning, if
	  triggered, was "Max silence should be less than minmessage or you
	  may get empty messages", which should have been logged if max
	  silence was greater than minmessage, but the check was for less
	  than. Also, conforming if statement to coding guidelines. closes
	  issue #15331) Reported by: markd Review:
	  https://reviewboard.asterisk.org/r/293/

2009-06-26 19:47 +0000 [r203710]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: moving debug message from level 0 to 1.
	  (closes issue #15404) Reported by: leobrown Patches:
	  iax_codec_debug.patch uploaded by leobrown (license 541)

2009-06-26 19:31 +0000 [r203702]  Russell Bryant <russell@digium.com>

	* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c:
	  Make invalid hints report Unavailable instead of Idle. (closes
	  issue #14413) Reported by: pj

2009-06-26 19:27 +0000 [r203699]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, main/frame.c, main/rtp_engine.c,
	  channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample,
	  include/asterisk/frame.h: Improve T.38 negotiation by exchanging
	  session parameters between application and channel.

2009-06-26 19:03 +0000 [r203672]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_analog.c: Check if polarityonanswerdelay has elapsed
	  before setting a channel as answered after a polarity reversal.
	  Previously on a polarity switch event chan_dahdi would set the
	  channel immediately as answered. This would cause problems if a
	  polarity reversal occurred when the line was picked up as the
	  dial would not have yet occurred. Now if the polarity reversal
	  occurs before delay has elapsed after coming off hook or an
	  answer, it is ignored. Also, some refactoring was done in
	  _handle_event. (closes issue #13917) Reported by: alecdavis
	  Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
	  alecdavis (license 585) Tested by: alecdavis

2009-06-26 15:42 +0000 [r203638-203640]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxyref.h, include/asterisk/channel.h: Note a
	  new API call, and one that changed in doxygen.

	* cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample (added),
	  cdr/cdr_sqlite3_custom.c, configs/cel.conf.sample (added),
	  channels/chan_local.c, include/asterisk/cel.h (added),
	  main/devicestate.c, apps/app_chanisavail.c, channels/chan_iax2.c,
	  doc/tex/cel-doc.tex (added), main/loader.c, main/cli.c,
	  channels/chan_dahdi.c, channels/sig_analog.c,
	  channels/chan_skinny.c, include/asterisk/event_defs.h,
	  main/features.c, res/ais/evt.c, channels/sig_analog.h,
	  channels/chan_alsa.c, doc/tex/asterisk.tex, cdr/cdr_manager.c,
	  apps/app_dial.c, main/pbx.c, include/asterisk/utils.h,
	  channels/chan_bridge.c, cel/cel_tds.c, channels/chan_agent.c,
	  configs/cel_adaptive_odbc.conf.sample (added),
	  include/asterisk/cdr.h, include/asterisk/channel.h, CHANGES,
	  main/cel.c (added), Makefile, channels/chan_misdn.c,
	  funcs/func_channel.c, funcs/func_cdr.c, doc/tex/celdriver.tex
	  (added), main/asterisk.c, cel/cel_adaptive_odbc.c,
	  apps/app_voicemail.c, res/res_calendar.c,
	  channels/chan_unistim.c, tests/test_substitution.c,
	  cel/cel_radius.c, channels/chan_multicast_rtp.c,
	  channels/chan_vpb.cc, apps/app_meetme.c, channels/chan_gtalk.c,
	  apps/app_followme.c, configs/cel_tds.conf.sample (added),
	  main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c,
	  main/manager.c, include/asterisk/event.h,
	  bridges/bridge_builtin_features.c, funcs/func_odbc.c,
	  cel/cel_custom.c, cel/cel_manager.c, cdr/cdr_sqlite.c,
	  res/res_agi.c, apps/app_minivm.c, main/logger.c,
	  apps/app_confbridge.c, configs/cel_custom.conf.sample (added),
	  channels/chan_mgcp.c, apps/app_parkandannounce.c,
	  cdr/cdr_custom.c, channels/chan_sip.c, cel (added),
	  configs/cel_pgsql.conf.sample (added), channels/chan_console.c,
	  include/asterisk/_private.h, channels/sig_pri.c,
	  apps/app_queue.c, channels/chan_oss.c, channels/sig_pri.h,
	  channels/chan_usbradio.c, channels/chan_jingle.c, cel/Makefile,
	  apps/app_celgenuserevent.c (added), apps/app_directed_pickup.c,
	  channels/chan_h323.c, cel/cel_sqlite3_custom.c, main/event.c,
	  channels/chan_nbs.c: Merge the new Channel Event Logging (CEL)
	  subsystem. CEL is the new system for logging channel events. This
	  was inspired after facing many problems trying to represent what
	  is possible to happen to a call in Asterisk using CDR records.
	  For more information on CEL, see the built in HTML or PDF
	  documentation generated from the files in doc/tex/. Many thanks
	  to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
	  work developing this code. Also, thanks to Matt Nicholson
	  (mnicholson) and Sean Bright (seanbright) for their assistance in
	  the final push to get this code ready for Asterisk trunk. Review:
	  https://reviewboard.asterisk.org/r/239/

2009-06-26 13:00 +0000 [r203569-203605]  Sean Bright <sean@malleable.com>

	* include/asterisk/syslog.h, main/syslog.c: Add functions to map
	  syslog facilities and priorities constants to strings. Also
	  change the default casing of the string contants to lowercase.
	  This really just saves us from have to lowercase them later when
	  displaying them.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/syslog.c: Add checks in configure for non-POSIX syslog
	  facilities.

2009-06-26 00:23 +0000 [r203525-203534]  Russell Bryant <russell@digium.com>

	* main/syslog.c: One more formatting nit ... use spaces for inline
	  indentation.

	* main/syslog.c: Convert spaces to tabs for indentation.

2009-06-25 23:54 +0000 [r203508]  Sean Bright <sean@malleable.com>

	* include/asterisk/syslog.h (added), main/logger.c, main/syslog.c
	  (added): Move syslog utility functions into a separate file so
	  they can be re-used. This has the pleasant side effect of
	  cleaning up the header inclusion process in logger.c.

2009-06-25 22:48 +0000 [r203479]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: make sure chan_dahdi compiles with only
	  libss7 and not libpri installed

2009-06-25 21:45 +0000 [r203444]  David Vossel <dvossel@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c: fixes a few redundant
	  conditions (issue #15269)

2009-06-25 21:34 +0000 [r203443]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Picking nits

2009-06-25 21:22 +0000 [r203402]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Remove
	  some unnecessary code and update sample config file with respect
	  to GR-303.

2009-06-25 21:15 +0000 [r203381]  Terry Wilson <twilson@digium.com>

	* /, main/cli.c: Merged revisions 203380 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
	  | 4 lines I didn't see that Mark already fixed the underlying
	  issue! Yay for removing useless code. ........

2009-06-25 21:04 +0000 [r203376]  Russell Bryant <russell@digium.com>

	* /, main/features.c: Merged revisions 203375 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
	  | 9 lines Fix a case where CDR answer time could be before the
	  start time involving parking. (closes issue #13794) Reported by:
	  davidw Patches: 13794.patch uploaded by murf (license 17)
	  13794.patch.160 uploaded by murf (license 17) Tested by: murf,
	  dbrooks ........

2009-06-25 20:25 +0000 [r203338]  Terry Wilson <twilson@digium.com>

	* /, main/cli.c: Merged revisions 203311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009)
	  | 2 lines Don't try to free NULL ........

2009-06-25 19:54 +0000 [r203304]  Jeff Peeler <jpeeler@digium.com>

	* channels/sig_pri.h (added), channels/chan_dahdi.c,
	  channels/sig_analog.c, channels/sig_analog.h, channels/sig_pri.c
	  (added), channels/Makefile: New signaling module to handle
	  PRI/BRI operations in chan_dahdi This merge splits the PRI/BRI
	  signaling logic out of chan_dahdi.c into sig_pri.c. Functionality
	  in theory should not change (mostly). A few trivial changes were
	  made in sig_analog with verbose messages and commenting.

2009-06-25 19:22 +0000 [r203258]  Jason Parker <jparker@digium.com>

	* channels/chan_dahdi.c: Unmute when we get a dtmfup (we muted on
	  dtmfdown) event. This would occasionally cause one-way audio when
	  using hardware DTMF detection. (closes issue #14761) Reported by:
	  tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
	  Tested by: tzafrir, dimas

2009-06-25 18:25 +0000 [r203227]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_multicast.c (added), channels/chan_multicast_rtp.c
	  (added), CHANGES: Add support for multicast RTP paging. (closes
	  issue #11797) Reported by: macbrody Review:
	  https://reviewboard.asterisk.org/r/270/

2009-06-25 17:01 +0000 [r203188]  Sean Bright <sean@malleable.com>

	* main/logger.c: Pass a logmsg to ast_log_vsyslog instead of
	  separate arguments.

2009-06-25 16:18 +0000 [r203126]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: Insure ring cadence is set for fxs ports
	  Moved SETCADENCE ioctl call to before call into new analog signal
	  module to insure that it gets set. (closes issue #15381) Reported
	  by: alecdavis Patches: fix15381.diff uploaded by dbailey (license
	  819) Tested by: dbailey

2009-06-25 16:04 +0000 [r203116]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 203115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
	  | 11 lines Resolve a crash related to a T.38 reinvite race
	  condition. This change resolves a crash observed locally during
	  some T.38 testing. A call was set up using a call file, and when
	  the T.38 reinvite came in, the channel state was still
	  AST_STATE_DOWN. The reason is explained by a comment in the code
	  that previously lived in the handling of AST_STATE_RINGING. This
	  change modifies the logic to handle the same race condition for
	  any channel state that is not UP. (closes ABE-1895) ........

2009-06-24 21:08 +0000 [r203037]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 203036 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24
	  Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error
	  checking. Valid format is: pritimer=timer_name,timer_value *
	  Fixed segfault if the ',' is missing. * Completely check the
	  range returned by pri_timer2idx() to prevent possible access
	  outside array bounds. ........

2009-06-24 18:29 +0000 [r202967]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 202966 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
	  2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
	  the same thing in-line. ........

2009-06-24 18:08 +0000 [r202925]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Ensure the default settings are applied for
	  T.38 when we set it up for a peer.

2009-06-24 13:53 +0000 [r202840-202889]  Sean Bright <sean@malleable.com>

	* doc/tex: Ignore some files generated when asterisk.pdf is
	  created.

	* configs/cdr_tds.conf.sample, cdr/cdr_tds.c: Update sample cdr_tds
	  configuration to try and eliminate some confusion. Also change
	  the preferred configuration option from 'hostname' (which was
	  misleading because it didn't actually treat the value as a
	  hostname) to 'connection' and added some verbage explaining that
	  the user would need to refer to their freetds.conf file for those
	  settings. 'hostname' was kept as a backwards compatible
	  configuration parameter.

	* doc/tex/billing.tex, doc/tex/cdrdriver.tex: Change some section
	  names in the CDR tex documentation.

	* doc/tex/cdrdriver.tex: Remove some trailing whitespace before
	  making content changes.

2009-06-23 22:47 +0000 [r202804]  Russell Bryant <russell@digium.com>

	* doc/tex/cdrdriver.tex: Clean up section hierarchy for the CDR
	  chapter.

2009-06-23 22:08 +0000 [r202761]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: I could have sworn I committed this patch
	  ages ago, but... bug fix with setting NAI properly on linksets in
	  certain situations.

2009-06-23 21:38 +0000 [r202755]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Make outgoing_colp=2 misdn.conf port
	  parameter not send redirecting or transfer messages. If the
	  outgoing_colp parameter is set to not send COLP information, then
	  it does not make sense to send redirecting or transfer messages
	  announcing new COLP information that is blocked. The service
	  provider may supply the listed number for that line when it
	  passes the messages to the next hop. Why tell the switch that
	  these events happened when the information is otherwise
	  suppressed? Also blocked the number of previous redirects that
	  may have occurred to calls going out the port when outgoing_colp
	  is 2. Follow on to JIRA ABE-1853.

2009-06-23 21:25 +0000 [r202753]  Ryan Brindley <rbrindley@digium.com>

	* main/config.c: If we delete the info, lets also delete the lines
	  (closes issue #14509) Reported by: timeshell Patches:
	  20090504__bug14509.diff.txt uploaded by tilghman (license 14)
	  Tested by: awk, timeshell

2009-06-23 16:31 +0000 [r202672]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 202671 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
	  | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
	  non-standard port and transport (closes issue #14659) Reported
	  by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
	  by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
	  by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
	  https://reviewboard.asterisk.org/r/288/ ........

2009-06-23 14:54 +0000 [r202497-202570]  Russell Bryant <russell@digium.com>

	* main/app.c, CHANGES: Ignore voicemail messages that are just
	  silence. (closes issue #2264) Reported by: pfn Patches:
	  silent-vm-1.6.2.txt uploaded by pfn (license 810)

	* main/channel.c, /: Merged revisions 202496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
	  | 4 lines Report CallerID change during a masquerade. Reported
	  by: markster ........

2009-06-22 16:09 +0000 [r202417]  Sean Bright <sean@malleable.com>

	* cdr/cdr_sqlite3_custom.c: Fix lock usage in cdr_sqlite3_custom to
	  avoid potential crashes during reload. Pointed out by Russell
	  while working on the CEL branch.

2009-06-22 16:05 +0000 [r202415]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 202414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
	  | 2 lines Make Polycom subscription type override check more
	  explicit. ........

2009-06-22 15:33 +0000 [r202410]  David Vossel <dvossel@digium.com>

	* include/asterisk/module.h, main/loader.c: attempting to load
	  running modules Modules placed in the priority heap for loading
	  were not properly removed from the linked list. This resulted in
	  some modules attempting to load twice.

2009-06-22 14:58 +0000 [r202337-202343]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 202341-202342 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
	  2009) | 26 lines Fix a situation in which Asterisk would not stop
	  retransmitting 487s. If a CANCEL were received by Asterisk, we
	  would send a 487 in response to the original INVITE and a 200 OK
	  for the CANCEL. If there were a network hiccup which caused the
	  200 OK and the 487 to be lost, then the UA communicating with
	  Asterisk may try to retransmit its CANCEL. Asterisk's response to
	  this used to be to try sending another 487 to the canceled INVITE
	  and another 200 OK to the CANCEL. The problem here is that the
	  originally-sent 487 was sent "reliably" meaning that it will be
	  retransmitted until it is received properly. So when we receive
	  the second CANCEL it is likely that the first batch of 487s we
	  sent is still going strong and reaches the UA. The result was
	  that the second set of 487s would be retransmitted constantly
	  until the maximum number of retries had been reached. The fix for
	  this is that if we receive a second CANCEL for an INVITE, then we
	  cancel the retransmission of the first set of 487s and start a
	  second set. This causes the dialog to be terminated reasonably.
	  (closes issue #14584) Reported by: klaus3000 Patches:
	  14584_v2.patch uploaded by mmichelson (license 60) Tested by:
	  klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
	  -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
	  left from previous commit. ........

	* /, channels/chan_sip.c: Merged revisions 202336 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
	  2009) | 25 lines Fix a possible infinite loop in SDP parsing
	  during glare situation. There was a while loop in
	  get_ip_and_port_from_sdp which was controlled by a call to
	  get_sdp_iterate. The loop would exit either if what we were
	  searching for was found or if the return was NULL. The problem is
	  that get_sdp_iterate never returns NULL. This means that if what
	  we were searching for was not present, the loop would run
	  infinitely. This modification of the loop fixes the problem.
	  (closes issue #15213) Reported by: schmidts (closes issue #15349)
	  Reported by: samy (closes issue #14464) Reported by: pj (closes
	  issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
	  uploaded by mmichelson (license 60) Tested by: aragon ........

2009-06-21 16:36 +0000 [r202223-202301]  Russell Bryant <russell@digium.com>

	* cdr/cdr_sqlite3_custom.c: Note a bug in cdr_sqlite3_custom so I
	  don't forget about it.

	* cdr/cdr_manager.c: Fix possibility of crashiness during reload in
	  custom fields handling.

	* cdr/cdr_manager.c: Standardize return values of load_config() so
	  reload() doesn't report an error on success.

	* cdr/cdr_manager.c: Leave a note about some unsafe code in
	  cdr_manager

2009-06-20 19:09 +0000 [r202183]  Sean Bright <sean@malleable.com>

	* apps/app_fax.c: Fix version detection for API changes in spandsp.
	  (closes issue #15355) Reported by: deuffy

2009-06-20 14:09 +0000 [r202109]  Russell Bryant <russell@digium.com>

	* main/cdr.c, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Remove
	  unnecessary usleep() from a couple of module unload callbacks. In
	  passing, also tweak cdr_unregister() to hold the list lock a bit
	  less time.

2009-06-19 21:25 +0000 [r202039]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Use sched_yield() instead of usleep(1)

2009-06-19 20:24 +0000 [r201994]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 201993 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19
	  Jun 2009) | 8 lines timestamp was being converted to host order
	  as a short rather than a long (closes issue #15361) Reported by:
	  ffloimair Patches: ts_issue.diff uploaded by dvossel (license
	  671) ........

2009-06-19 17:40 +0000 [r201944]  Terry Wilson <twilson@digium.com>

	* CHANGES: Add note about the addition of calendar support

2009-06-19 15:47 +0000 [r201904]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c: Fix 2 typos and add support for wide
	  character types. Reported by Benny Amorsen via the asterisk-users
	  mailing list.
	  http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html

2009-06-19 15:41 +0000 [r201902]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, channels/chan_sip.c,
	  include/asterisk/rtp_engine.h: Add support for allowing an RTP
	  engine to decide on whether it is possible for specific formats
	  to be transcoded for an RTP instance.

2009-06-19 00:43 +0000 [r201745-201829]  Tilghman Lesher <tlesher@digium.com>

	* /, main/features.c: Merged revisions 201828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
	  | 6 lines If the "h" extension fails, give it another chance in
	  main/pbx.c. If the "h" extension fails, give it another chance in
	  main/pbx.c, when it returns from the bridge code. Fixes an issue
	  where the "h" extension may occasionally not fire, when a Dial is
	  executed from a Macro. Debugged in #asterisk with user tompaw.
	  ........

	* apps/Makefile: One of the changes in 1.6.1 was to allow
	  app_directory to use functionality within app_voicemail for
	  directory functions. It is therefore no longer necessary for
	  app_directory to be linked against the ODBC libraries (and it
	  never was necessary for app_directory to be linked against IMAP,
	  though it was).

	* funcs/func_cut.c: Clarify CUT code, and in the process, fix a bug
	  in trunk only (closes issue #15320) Reported by: chappell
	  Patches: cut_fix.patch uploaded by chappell (license 8)
	  cut_clarify.patch uploaded by chappell (license 8)

2009-06-18 17:41 +0000 [r201717]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c: Added deadlock protection to
	  try_suggested_sip_codec in chan_sip.c. Review:
	  https://reviewboard.asterisk.org/r/285/

2009-06-18 16:37 +0000 [r201678]  David Vossel <dvossel@digium.com>

	* codecs/gsm/src/gsm_destroy.c, channels/h323/ast_h323.cxx,
	  main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
	  utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c,
	  res/res_config_ldap.c, apps/app_rpt.c, channels/misdn/isdn_lib.c,
	  main/asterisk.c, utils/conf2ael.c, main/ast_expr2.c,
	  utils/stereorize.c: fixes some memory leaks and redundant
	  conditions (closes issue #15269) Reported by: contactmayankjain
	  Patches: patch.txt uploaded by contactmayankjain (license 740)
	  memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
	  Tested by: contactmayankjain, dvossel

2009-06-18 15:27 +0000 [r201610]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 201600 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18
	  Jun 2009) | 29 lines Fix memory corruption and leakage related
	  reloads of non files mode MoH classes. For Music on Hold classes
	  that are not files mode, meaning that we are executing an
	  application that will feed us audio data, we use a thread to
	  monitor the external application and read audio from it. This
	  thread also makes use of the MoH class object. In the MoH class
	  destructor, we used pthread_cancel() to ask the thread to exit.
	  Unfortunately, the code did not wait to ensure that the thread
	  actually went away. What needed to be done is a pthread_join() to
	  ensure that the thread fully cleans up before we proceed. By
	  adding this one line, we resolve two significant problems: 1)
	  Since the thread was never joined, it never fully goes away. So,
	  on every reload of non-files mode MoH, an unused thread was
	  sticking around. 2) There was a race condition here where the
	  application monitoring thread could still try to access the MoH
	  class, even though the thread executing the MoH reload has
	  already destroyed it. (issue #15109) Reported by: jvandal (issue
	  #15123) Reported by: axisinternet (issue #15195) Reported by:
	  amorsen (issue AST-208) ........

2009-06-18 15:20 +0000 [r201583]  Mark Michelson <mmichelson@digium.com>

	* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
	  include/asterisk/rtp_engine.h: Trunk implementation of setting an
	  alternate RTP source. This contains the interface by which we can
	  let an rtp instance know that it might start receiving audio from
	  a new source. This is similar in nature to revision 197588 of
	  Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276

2009-06-18 15:16 +0000 [r201534-201570]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: parsing extension correctly from sip
	  register lines If a transport type was specified, but no
	  extension, parsing of the extension would return whatever was
	  after the transport rather than defaulting to 's'. (closes issue
	  #15111) Reported by: ffs Patches:
	  chan_sip.c_register-parser.patch uploaded by ffs (license 730)
	  Tested by: ffs, dvossel

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add
	  rtsavesysname to chan_iax chan_sip has an option to save the
	  sysname on rtupdate. This patch copies that same logic to
	  chan_iax. (closes issue #14837) Reported by: barthpbx Patches:
	  iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
	  rt_iax.diff uploaded by dvossel (license 671)

2009-06-17 21:31 +0000 [r201531]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Initialize additional variables, to prevent
	  a possible crash. (closes issue #15186) Reported by: ajohnson
	  Patches: 20090528__issue15186.diff.txt uploaded by tilghman
	  (license 14) Tested by: ajohnson

2009-06-17 20:10 +0000 [r201458-201462]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix problem with no audio due to ignoring
	  the SDP. A recent change to our SDP version comparison made audio
	  not function on some calls. This was because of a test wherein we
	  were trying to see if an unsigned value was less than 0. This is
	  a dumb comparison and arguably the compiler should have warned
	  about it. Alas, though, it slipped past. Now it's fixed by
	  changing the variable to be a signed type. Found by several
	  developers. Tested by mnicholson and dbrooks.

	* main/channel.c, /: Merged revisions 201450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
	  2009) | 9 lines Change the datastore traversal in
	  ast_do_masquerade to use a safe list traversal. It is possible
	  for datastore fixup functions to remove the datastore from the
	  list and free it. In particular, the queue_transfer_fixup in
	  app_queue does this. While I don't yet know of this causing any
	  crashes, it certainly could. Found while discussing a separate
	  issue with Brian Degenhardt. ........

2009-06-17 20:00 +0000 [r201445-201453]  David Vossel <dvossel@digium.com>

	* doc/datastores.txt: ast_channel_datastore_alloc is no longer
	  used. updating datastores.txt to reflect that.

	* /, apps/app_mixmonitor.c: Merged revisions 201423 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17
	  Jun 2009) | 19 lines StopMixMonitor race condition (not giving up
	  file immediately) StopMixMonitor only indicates to the MixMonitor
	  thread to stop writing to the file. It does not guarantee that
	  the recording's file handle is available to the dialplan
	  immediately after execution. This results in a race condition. To
	  resolve this, the filestream pointer is placed in a datastore on
	  the channel. When StopMixMonitor is called, the datastore is
	  retrieved from the channel and the filestream is closed
	  immediately before returning to the dialplan. Documentation
	  indicating the use of StopMixMonitor to free files has been
	  updated as well. (closes issue #15259) Reported by: travisghansen
	  Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/283/ ........

2009-06-17 19:15 +0000 [r201381]  David Brooks <dbrooks@digium.com>

	* /, channels/chan_sip.c: Merged revisions 201380 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
	  | 9 lines Checks for NULL sip_pvt pointer in
	  chan_sip.c->acf_channel_read() Zombie channels could be passed,
	  and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
	  checking for NULL pointer. (closes issue #15330) Reported by:
	  okrief Tested by: dbrooks ........

2009-06-17 15:20 +0000 [r201331-201344]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP registry ref count error During a sip
	  reload, the list of sip_registry objects are supposed to be
	  traversed, unlinked, and destroyed, but destruction never takes
	  place due to a ref counting error. This causes a memory leak when
	  registry items are removed from sip.conf and reloaded. While the
	  registries are removed from the global list, they are not removed
	  from the scheduler. Because of this, SIP register attempts
	  continue to be sent out for the item even though it may no longer
	  be in the .conf. (closes issue #15295) Reported by: amorsen
	  Review: https://reviewboard.asterisk.org/r/282/

	* channels/chan_iax2.c: update chan_iax to use 64bit feature flags.
	  (closes issue #15335) Reported by: lmadsen Review:
	  https://reviewboard.asterisk.org/r/284/

2009-06-17 12:04 +0000 [r201262]  Kevin P. Fleming <kpfleming@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 201261 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
	  2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
	  to be appended is empty. When the list to be appended is empty,
	  and the list to be appended to is *not*, AST_LIST_APPEND_LIST
	  would actually cause the target list to become broken, and no
	  longer have a pointer to its last entry. This patch fixes the
	  problem. (reported by Stanislaw Pitucha on the asterisk-dev
	  mailing list) ........

2009-06-16 22:29 +0000 [r201223]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: fix issue with build_contact introduced by
	  the "SIP trasnport type issues" commit

2009-06-16 22:11 +0000 [r201190]  Sean Bright <sean@malleable.com>

	* CREDITS: Update my e-mail address (thanks for the props, russell
	  :))

2009-06-16 21:10 +0000 [r200985-201139]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, channels/chan_sip.c, apps/app_fax.c,
	  include/asterisk/frame.h: Enable applications to enable/disable
	  digit and tone detection. Some applications (notably app_fax) do
	  not need digit detection nor FAX tone detection while they are
	  running, and if Asterisk is using software DSPs to provide the
	  detection, this consumes extra CPU cycles that could be better
	  spent on the actual application. This patch allows applications
	  to query and control the state of digit and tone detection on a
	  channel, and modifies app_fax to disable them while the FAX
	  operations are occurring (and re-enable digit detection
	  afterwards).

	* configure, configure.ac: Explicitly test for 'static weakref'
	  support. Since we use 'static' weakref symbols, and not all GCC
	  versions support them, test for that combination explicitly.

	* Makefile: When compiling in an Emacs-spawned shell, always print
	  directory names. This change ensures that Emacs can find the
	  proper source files when parsing compiler error messages, since
	  it uses the 'make' output including directory names to do it.

	* configure, autoconf/ast_gcc_attribute.m4, configure.ac: Another
	  minor fix to compiler attribute checking. Defaulting to 'static'
	  for the function scope was bad... so remove it.

	* main/channel.c, main/autoservice.c, main/frame.c, /,
	  apps/app_meetme.c, main/slinfactory.c,
	  include/asterisk/linkedlists.h, main/file.c,
	  include/asterisk/channel.h, include/asterisk/frame.h,
	  apps/app_chanspy.c, apps/app_mixmonitor.c: Merged revisions
	  200991 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
	  2009) | 11 lines Improve support for media paths that can
	  generate multiple frames at once. There are various media paths
	  in Asterisk (codec translators and UDPTL, primarily) that can
	  generate more than one frame to be generated when the application
	  calling them expects only a single frame. This patch addresses a
	  number of those cases, at least the primary ones to solve the
	  known problems. In addition it removes the broken TRACE_FRAMES
	  support, fixes a number of bugs in various frame-related API
	  functions, and cleans up various code paths affected by these
	  changes. https://reviewboard.asterisk.org/r/175/ ........

	* configure, autoconf/ast_gcc_attribute.m4, configure.ac: Fix
	  problems with new compiler attribute checking in configure
	  script. The last changes to ast_gcc_attribute.m4 caused some
	  problems checking for various attributes, because the scope of
	  the symbol the attribute is applied to can be important; this
	  patch allows the scope to be specified for the check.

2009-06-16 16:03 +0000 [r200946]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP transport type issues What this patch
	  addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP
	  address/port reguardless if the sip->pvt is of type UDP or not.
	  Now when no remapping is required, ast_sip_ouraddrfor() checks
	  the sip_pvt's transport type, attempting to set the address and
	  port to the correct TCP/TLS bindings if necessary. 2. It is not
	  necessary to send the port number in the Contact header unless
	  the port is non-standard for the transport type. This patch fixes
	  this and removes the todo note. 3. In sip_alloc(), the default
	  dialog built always uses transport type UDP. Now sip_alloc()
	  looks at the sip_request (if present) and determines what
	  transport type to use by default. 4. When changing the transport
	  type of a sip_socket, the file descriptor must be set to -1 and
	  in some cases the tcptls_session's ref count must be decremented
	  and set to NULL. I've encountered several issues associated with
	  this process and have created a function, set_socket_transport(),
	  to handle the setting of the socket type. (closes issue #13865)
	  Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
	  Kristijan (license 753) 13865.patch uploaded by mmichelson
	  (license 60) tls_port_v5.patch uploaded by vrban (license 756)
	  transport_issues.diff uploaded by dvossel (license 671) Tested
	  by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
	  https://reviewboard.asterisk.org/r/278/

2009-06-16 15:51 +0000 [r200943]  Michiel van Baak <michiel@vanbaak.info>

	* apps/app_voicemail.c: add FILE_STORAGE to Voicemail Build Options
	  Voicemail can only use one storage module at the moment. Because
	  it's unclear that selecting one of the storage modules in
	  menuselect will disable filesystem storage we now have a
	  FILE_STORAGE option that conflicts with the other modules.
	  (closes issue #15333)

2009-06-16 15:26 +0000 [r200942]  Russell Bryant <russell@digium.com>

	* CREDITS: Add Sean Bright to CREDITS - Thanks, Sean!

2009-06-16 14:12 +0000 [r200841-200878]  Eliel C. Sardanons <eliels@gmail.com>

	* /: Recorded merge of revisions 200875 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) |
	  5 lines Show the interface name on error, if it is not found. If
	  the smdiport specified is not found, show the interface name
	  instead of '(null)'. ........

	* res/res_smdi.c: Show the interface name on error, if it is not
	  found. If the smdiport specified is not found, show the interface
	  name instead of '(null)'.

2009-06-16 02:32 +0000 [r200805]  Russell Bryant <russell@digium.com>

	* main/manager.c: Don't claim a char * is a mansession object.
	  Since there was only 1 bucket, and no hash function was
	  specified, the code actually worked perfectly fine. However, in
	  theory, this was invalid use of the OBJ_POINTER flag, so remove
	  it so the code provides a better usage example.

2009-06-16 02:24 +0000 [r200799]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: keep
	  backwards compatible chan_dahdi with older openr2 versions by not
	  using the new skip category feature unless supported

2009-06-16 01:28 +0000 [r200764]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, autoconf/ast_gcc_attribute.m4: Ensure that
	  configure-script testing for compiler attributes actually works.
	  The configure script tests for compiler attributes didn't
	  actually enable enough warnings or provide a proper test harness
	  to determine whether the compiler supports the attribute in
	  question or not; this caused gcc 4.1 to report that it supports
	  'weakref', but it doesn't actually support it in the way that is
	  needed for our optional API mechanism. The new configure script
	  test will properly distinguish between full support and partial
	  support for this attribute, among others.

2009-06-16 01:26 +0000 [r200762]  Russell Bryant <russell@digium.com>

	* doc/tex/channelvariables.tex: Add missing closure of verbatim
	  environment.

2009-06-16 01:03 +0000 [r200519-200726]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: Document the new automatic 'ignoresdpversion' behavior.
	  Asterisk will now automatically ignore incorrect incoming SDP
	  version numbers when necessary to complete a T.38 re-INVITE
	  operation.

	* channels/chan_sip.c: Accept T.38 re-INVITE responses with invalid
	  SDP versions. This commit changes the 'incoming SDP version'
	  check logic a bit more; when 'ignoresdpversion' is *not* set for
	  a peer, if we initiate a re-INVITE to switch to T.38, we'll
	  always accept the peer's SDP response, even if they don't
	  properly increment the SDP version number as they should. If this
	  situation occurs, a warning message will be generated suggesting
	  that the peer's configuration be changed to include the
	  'ignoresdpversion' configuration option (although ideally they'd
	  fix their SIP implementation to be RFC compliant). AST-221

	* doc/CODING-GUIDELINES, apps/app_read.c, apps/app_page.c,
	  apps/app_fax.c, apps/app_readexten.c, apps/app_queue.c,
	  include/asterisk/app.h, apps/app_skel.c, apps/app_minivm.c,
	  apps/app_macro.c, apps/app_url.c, apps/app_sms.c,
	  apps/app_externalivr.c, apps/app_stack.c, apps/app_mixmonitor.c,
	  apps/app_voicemail.c: Last batch of 'static' qualifiers for
	  module-level global variables. Fix up modules in the 'apps'
	  directory, and also correct the bad example of enum definitions
	  in include/asterisk/app.h, which many developers followed (thanks
	  for reading the documentation!). In addition, add some basic
	  usage examples of the 'pahole' and 'pglobal' tools to the coding
	  guidelines.

	* res/res_snmp.c, main/devicestate.c, funcs/func_vmcount.c,
	  res/res_calendar_caldav.c, formats/format_wav_gsm.c,
	  res/res_jabber.c, main/loader.c, main/cli.c, funcs/func_enum.c,
	  main/manager.c, res/res_smdi.c, funcs/func_odbc.c,
	  main/features.c, main/logger.c, main/http.c, pbx/pbx_realtime.c,
	  main/image.c, main/db.c, cdr/cdr_manager.c,
	  res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
	  res/res_config_pgsql.c, funcs/func_lock.c, pbx/pbx_lua.c,
	  funcs/func_cut.c, include/asterisk/calendar.h,
	  funcs/func_realtime.c, funcs/func_curl.c, funcs/func_cdr.c,
	  funcs/func_channel.c, main/file.c, main/event.c, pbx/pbx_dundi.c,
	  main/xmldoc.c, res/res_calendar.c: More 'static' qualifiers on
	  module global variables. The 'pglobal' tool is quite handy indeed
	  :-)

	* channels/chan_dahdi.c, channels/chan_misdn.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_agent.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Convert a number of global module variables
	  to 'static'. These modules all contained variables that are
	  module-global but not system-global, but were not marked
	  'static'.

	* channels/chan_sip.c: Some minor structure size improvements in
	  sip_pvt and sip_peer. Using the 'pahole' tool, it is now quite
	  easy to see where structure fields could be organized differently
	  to keep the compiler from having to add padding to satisfy
	  alignment requirements. These changes reduced the sizes of
	  sip_pvt and sip_peer by a few bytes each (on 64-bit platforms),
	  and also fixed a spelling error in a field name.

	* include/asterisk/agi.h, main/Makefile,
	  include/asterisk/autoconfig.h.in, res/res_smdi.exports,
	  configure.ac, res/res_agi.exports, include/asterisk/compiler.h,
	  apps/app_queue.c, res/res_monitor.c,
	  include/asterisk/optional_api.h, Makefile, res/res_smdi.c,
	  configure, res/res_agi.c, include/asterisk/monitor.h,
	  apps/app_stack.c, include/asterisk/smdi.h,
	  res/res_monitor.exports, apps/app_voicemail.c: Redesigned
	  'optional API' support. This patch provides a new implementation
	  of the optional API support defined in asterisk/optional_api.h;
	  this new version provides solves compatibility issues with the
	  use of linker version scripts for suppressing global symbols. In
	  addition, there is now a functional (and tested!) implementation
	  for Mac OS/X, so module writers no longer need to use special
	  tests before calling optional API functions. All future
	  implementations must provide these same semantics, so that module
	  writers can rely on them.

2009-06-15 15:22 +0000 [r200514]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 200513 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
	  2009) | 5 lines Add INFO to our allowed methods so that endpoints
	  know they may send it to us. AST-223 ........

2009-06-14 06:13 +0000 [r200477]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  build_tools/menuselect-deps.in: added openr2 to
	  menuselect-deps.in, recent commit in menuselect made me realize
	  this was never done but was working anyways also added support
	  for skip category request feature of openr2 and updated
	  chan_dahdi.conf.sample

2009-06-12 19:46 +0000 [r200428-200430]  Sean Bright <sean@malleable.com>

	* contrib/upstart/asterisk.upstart-0.3.9: Include basic
	  installation and usage instructions for upstart script.

	* contrib/upstart/asterisk.upstart-0.3.9 (added), contrib/upstart
	  (added): First shot at an upstart script for asterisk on Ubuntu.
	  This works relatively well (assuming you are using
	  /var/run/asterisk) as your run directory and upstart 0.3.9. Needs
	  to be generalized and eventually added to the 'make install'
	  target for Ubuntu.

2009-06-12 19:07 +0000 [r200290-200361]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 200360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
	  2009) | 10 lines Suppress a warning message and give a better
	  return code when generating inband ringing after a call is
	  answered. (closes issue #15158) Reported by: madkins Patches:
	  15158.patch uploaded by mmichelson (license 60) Tested by:
	  madkins ........

	* channels/chan_local.c, apps/app_queue.c: Fix some bad locking
	  stemming from trying to forward a call to a non-existent
	  extension from a queue.

	* apps/app_queue.c: Fix a potential crash from trying to access a
	  NULL channel pointer.

2009-06-12 02:20 +0000 [r200254]  Sean Bright <sean@malleable.com>

	* contrib/init.d/rc.debian.asterisk: Call chgrp instead of chown
	  when setting run directory group ownership. (issue #13153)
	  Reported by: pabelanger

2009-06-11 21:17 +0000 [r200146]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a crash due to a potentially NULL
	  p->options. Thanks to mnicholson for pointing it out.

2009-06-11 15:40 +0000 [r200108]  Eliel C. Sardanons <eliels@gmail.com>

	* main/channel.c: Release the allocated channel decreasing the
	  reference counter. When allocating the channel use ao2_ref(-1) to
	  release it, instead of calling ast_free(). Also avoid freeing
	  structures inside that channel (on error) if they will be
	  released by the channel destructor being called if the reference
	  counter reachs 0.

2009-06-11 12:15 +0000 [r200039]  Leif Madsen <lmadsen@digium.com>

	* build_tools/make_version_c, build_tools/make_version_h: Fix path
	  for .flavor and .version (issue #14737) Reported by: davidw
	  Patches: flavor.patch uploaded by davidw (license 780) Tested by:
	  davidw

2009-06-10 20:40 +0000 [r200000]  Sean Bright <sean@malleable.com>

	* sample.call: Remove some trailing whitespace and steal revision
	  200000.

2009-06-10 20:15 +0000 [r199958]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Only try to use the invite_branch on
	  outgoing INVITEs with auth credentials. I have added a comment to
	  the code to help ease understanding of the logic here as well.

2009-06-10 20:00 +0000 [r199957]  David Brooks <dbrooks@digium.com>

	* main/pbx.c: Fixes the argument order in definition of
	  new_find_extension(). In the definition of new_find_extension(),
	  the arguments 'callerid' and 'label' were swapped. The prototype
	  declaration and all calls to the function are ordered 'callerid'
	  then 'label', but the function itself was ordered 'label' then
	  'callerid'. (closes issue #15303) Reported by: JimDickenson

2009-06-10 18:58 +0000 [r199923]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Use ast_channel_unref to instead of ast_free on a
	  newly created channel. Also I removed an unnecessary free of a
	  cid_name. This will be freed properly in the channel destructor.
	  Reported by mnicholson in #asterisk-dev.

2009-06-10 16:10 +0000 [r199857]  Sean Bright <sean@malleable.com>

	* include/asterisk/utils.h, /: Merged revisions 199856 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
	  10 Jun 2009) | 2 lines __WORDSIZE is not available on all
	  platforms, so use sizeof(void *) instead. ........

2009-06-09 20:47 +0000 [r199818]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: CLI NOTIFY sending wrong transport type.
	  SIP's cli NOTIFY command only used UDP rather than copying the
	  transport type from the peer. (closes issue #15283) Reported by:
	  jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by
	  jthurman (license 614) Tested by: jthurman, dvossel

2009-06-09 18:08 +0000 [r199781]  Sean Bright <sean@malleable.com>

	* Makefile: Fix all of the parallel build warnings issued when
	  running make -j#.

2009-06-09 16:22 +0000 [r199743]  David Vossel <dvossel@digium.com>

	* res/res_timing_pthread.c, include/asterisk/module.h,
	  res/res_timing_dahdi.c, res/res_timing_timerfd.c, main/loader.c:
	  module load priority This patch adds the option to give a module
	  a load priority. The value represents the order in which a
	  module's load() function is initialized. The lower the value, the
	  higher the priority. The value is only checked if the
	  AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
	  flag is not set, the value will never be read and the module will
	  be given the lowest possible priority on load. Since some modules
	  are reliant on a timing interface, the timing modules have been
	  given a high load priorty. (closes issue #15191) Reported by:
	  alecdavis Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/262/

2009-06-08 22:08 +0000 [r199696]  Tilghman Lesher <tlesher@digium.com>

	* doc/janitor-projects.txt: Add sigaction janitor

2009-06-08 19:33 +0000 [r199630]  Sean Bright <sean@malleable.com>

	* include/asterisk/utils.h, /: Merged revisions 199626,199628 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
	  2009) | 21 lines Increase the size of our thread stack on 64 bit
	  processors. We were setting the stack size for each thread to
	  240KB regardless of architecture, which meant that in some
	  scenarios we actually had less available stack space on 64 bit
	  processors (pointers use 8 bytes instead of 4). So now we
	  calculate the stack size we reserve based on the platform's
	  __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
	  bit -> 1008KB (that's right, we're ready for 128 bit processors)
	  Patch typed by me but written by several members of
	  #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
	  issue #14932) Reported by: jpiszcz Patches:
	  06052009_issue14932.patch uploaded by seanbright (license 71)
	  Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
	  15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
	  stack size calculation just introduced. ........

2009-06-08 17:32 +0000 [r199588]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix a deadlock that could occur when setting
	  rtp stats on SIP calls. (closes issue #15143) Reported by:
	  cristiandimache Patches: 15143.patch uploaded by mmichelson
	  (license 60) Tested by: cristiandimache

2009-06-07 19:15 +0000 [r199514-199547]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_osplookup.c: Move OSP* applications static documentation
	  to XML. Move OSP* applications static documentation to the new
	  AstXML form. (closes issue #15245) Reported by: eliel Patches:
	  app_osplookup_static_conversion.txt uploaded by lmadsen (license
	  10)

	* apps/app_externalivr.c: Move application ExternalIVR static
	  documentation to XML. Move application ExternalIVR static
	  documentation to the new AstXML form. (issue #15245) Reported by:
	  eliel Patches: app_externalivr.diff uploaded by eliel (license
	  64)

2009-06-07 14:55 +0000 [r199479]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, apps/app_dahdibarge.c, apps/app_dictate.c,
	  apps/app_authenticate.c, apps/app_echo.c, apps/app_fax.c,
	  apps/app_dahdiras.c, apps/app_disa.c, apps/app_alarmreceiver.c,
	  apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c,
	  apps/app_controlplayback.c, apps/app_channelredirect.c,
	  apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c,
	  apps/app_confbridge.c, apps/app_directory.c, apps/app_chanspy.c,
	  apps/app_adsiprog.c: Global var cleanup - constification and
	  removing unused vars.

2009-06-06 23:28 +0000 [r199374-199446]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_stack.c: Move AGI command 'gosub' static documentation
	  to XML. Move AGI command 'gosub' statis documentation to the new
	  AstXML form. (issue #15245) Reported by: eliel Patches:
	  app_stack_static_conversion.txt uploaded by lmadsen (license 10)
	  (with minor changes by me)

	* res/res_musiconhold.c: Move music on hold related applications
	  documentation to XML. Move MusicOnHold, SetMusicOnHold,
	  StartMusicOnHold, StopMusicOnHold static documentation to the new
	  AstXML form. (issue #15245) Reported by: eliel Patches:
	  res_musiconhold_static_conversion.txt uploaded by lmadsen
	  (license 10) (with some fixes and formatting by me)

	* res/res_phoneprov.c: Move function PP_EACH_USER and
	  PP_EACH_EXTENSION documentation to XML. Move function
	  PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
	  AstXML form. (issue #15245) Reported by: eliel Patches:
	  res_phoneprov_static_conversion.txt uploaded by lmadsen (license
	  10) (with PP_EACH_USER add by me)

	* apps/app_meetme.c: Move function MEETME_INFO documentation to
	  XML. Move function MEETME_INFO static documentation to the new
	  AstXML form. (issue #15245) Reported by: eliel Patches:
	  app_meetme_static_conversion.txt uploaded by lmadsen (license 10)

	* apps/app_minivm.c: Move function MINIVMACCOUNT and MINIVMCOUNTER
	  static documentation to XML. Move function MINIVMACCOUNT and
	  MINIVMCOUNTER statis documentation to the new AstXML form. (issue
	  #15245) Reported by: eliel Patches:
	  app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
	  (with minor changes by me)

	* funcs/func_sysinfo.c: Move function SYSINFO documentation to XML.
	  Move function SYSINFO static documentation to the new AstXML
	  form. (issue #15245) Reported by: eliel Patches:
	  func_sysinfo_static_conversion.txt uploaded by lmadsen (license
	  10)

2009-06-06 21:42 +0000 [r199368-199372]  Russell Bryant <russell@digium.com>

	* apps/app_jack.c: minor tweak

	* apps/app_jack.c: Constify a string and strip trailing whitespace.

	* Makefile: Switch from "echo -n" to printf. On my mac, the -n was
	  just getting printed out.

2009-06-05 21:21 +0000 [r199298]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, /, main/devicestate.c: Merged
	  revisions 199297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
	  | 14 lines Fixes issue with hints giving unexpected results.
	  Hints with two or more devices that include ONHOLD gave
	  unexpected results. (closes issue #15057) Reported by:
	  p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
	  (license 671) pbx.c.1.4.patch uploaded by p (license 558)
	  devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
	  p_lindheimer, dvossel Review:
	  https://reviewboard.asterisk.org/r/254/ ........

2009-06-05 13:51 +0000 [r199227]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c: Correct "dahdi show channels" output when
	  specifying a group. Since a DAHDI channel may belong to multiple
	  groups, we need to use a bitwise and instead of equivalence to
	  determine whether to display the channel information. (closes
	  issue #15248) Reported by: gentian Patches: 15248.patch uploaded
	  by mmichelson (license 60) Tested by: gentian

2009-06-04 19:10 +0000 [r199139]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 199138 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
	  Jun 2009) | 3 lines Additional updates to AST-2009-001 ........

2009-06-04 16:29 +0000 [r199091]  Eliel C. Sardanons <eliels@gmail.com>

	* res/res_smdi.c: Move static docs to the new AstXML form. Move
	  SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation to
	  XML. (issue #15245) Reported by: eliel Patches:
	  res_smdi_static_conversion.txt uploaded by lmadsen (license 10)

2009-06-04 14:31 +0000 [r199051]  Sean Bright <sean@malleable.com>

	* /, include/asterisk/_private.h, main/asterisk.c, main/loader.c:
	  Merged revisions 199022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
	  2009) | 40 lines Safely handle AMI connections/reload requests
	  that occur during startup. During asterisk startup, a lock on the
	  list of modules is obtained by the primary thread while each
	  module is initialized. Issue 13778 pointed out a problem with
	  this approach, however. Because the AMI is loaded before other
	  modules, it is possible for a module reload to be issued by a
	  connected client (via Action: Command), causing a deadlock. The
	  resolution for 13778 was to move initialization of the manager to
	  happen after the other modules had already been lodaded. While
	  this fixed this particular issue, it caused a problem for users
	  (like FreePBX) who call AMI scripts via an #exec in a
	  configuration file (See issue 15189). The solution I have come up
	  with is to defer any reload requests that come in until after the
	  server is fully booted. When a call comes in to ast_module_reload
	  (from wherever) before we are fully booted, the request is added
	  to a queue of pending requests. Once we are done booting up, we
	  then execute these deferred requests in turn. Note that I have
	  tried to make this a bit more intelligent in that it will not
	  queue up more than 1 request for the same module to be reloaded,
	  and if a general reload request comes in ('module reload') the
	  queue is flushed and we only issue a single deferred reload for
	  the entire system. As for how this will impact existing
	  installations - Before 13778, a reload issued before module
	  initialization was completed would result in a deadlock. After
	  13778, you simply couldn't connect to the manager during startup
	  (which causes problems with #exec-that-calls-AMI configuration
	  files). I believe this is a good general purpose solution that
	  won't negatively impact existing installations. (closes issue
	  #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
	  06032009_15189_deferred_reloads.diff uploaded by seanbright
	  (license 71) Tested by: p_lindheimer, seanbright Review:
	  https://reviewboard.asterisk.org/r/272/ ........

2009-06-03 20:30 +0000 [r198824-198954]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c, main/channel.c, apps/app_queue.c:
	  ast_call_forward() todo notes and originate flag copy.

	* main/channel.c, main/features.c, include/asterisk/channel.h:
	  Generic call forward api, ast_call_forward() The function
	  ast_call_forward() forwards a call to an extension specified in
	  an ast_channel's call_forward string. After an ast_channel is
	  called, if the channel's call_forward string is set this function
	  can be used to forward the call to a new channel and terminate
	  the original one. I have included this api call in both
	  channel.c's ast_request_and_dial() and feature.c's
	  feature_request_and_dial(). App_dial and app_queue already
	  contain call forward logic specific for their application and
	  options. (closes issue #13630) Reported by: festr Review:
	  https://reviewboard.asterisk.org/r/271/

	* channels/chan_iax2.c: fixes issue with channels not going down
	  after transfer Iax2 currently does not support native bridging if
	  the timeoutms value is set. We check for that in iax2_bridge, but
	  then set timeoutms to 0 by default. If the timeoutms is not
	  provided it is set to -1. By setting timeoutms to 0 it is
	  processed causing a bridging retry loop. (closes issue #15216)
	  Reported by: oxymoron Tested by: dvossel

2009-06-02 13:48 +0000 [r198762-198791]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Correct
	  documentation for the register line, specifically where the
	  domain should be specified. (closes issue #14367) Reported by:
	  Nick_Lewis

	* main/rtp_engine.c: Fix a bug where we were passing in address
	  information that should remain unmodified to a function that may
	  modify it. (closes issue #15243) Reported by: pj

2009-06-01 21:03 +0000 [r198729]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.c: Tell the IAX2 parser about more control
	  frame types.

2009-06-01 20:57 +0000 [r198727]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
	  main/dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
	  main/features.c, apps/app_macro.c, doc/tex/channelvariables.tex,
	  main/app.c, include/asterisk/channel.h, apps/app_queue.c: Add the
	  ability to execute connected line interception macros. When
	  connected line updates are received or generated in the middle of
	  an application call, it is now possible to execute a macro to
	  manipulate the connected line data. This way, phone numbers may
	  be manipulated to be more presentable to users, names may be
	  changed for...whatever reason, or whatever else needs to be done
	  may be. Review: https://reviewboard.asterisk.org/r/256 AST-165

2009-06-01 20:33 +0000 [r198725]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_math.c: Add INCrement and DECrement functions (closes
	  issue #15025) Reported by: greenfieldtech Patches:
	  func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
	  slightly modified by me Tested by: greenfieldtech, lmadsen

2009-06-01 20:17 +0000 [r198670]  Russell Bryant <russell@digium.com>

	* include/asterisk/frame.h: Minor whitespace fix.

2009-06-01 19:37 +0000 [r198661]  Eliel C. Sardanons <eliels@gmail.com>

	* res/res_monitor.c: Moved more static documentation to the new
	  AstXML form. Moved more static docs to XML (pplications and
	  manager actions): Monitor, StopMonitor, ChangeMonitor,
	  PauseMonitor, UnpauseMonitor.

2009-06-01 18:40 +0000 [r198626]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/meetme.sql: Add information for new meetme
	  realtime fields

2009-06-01 17:53 +0000 [r198561-198597]  Eliel C. Sardanons <eliels@gmail.com>

	* main/Makefile: Do not add say.o in a separate line.

	* res/res_jabber.c: Move JabberSend manager action from static docs
	  to the AstXML form.

	* res/res_agi.c: Move static documentation of E|Dead|AGI()
	  application and manager action to XML.

2009-06-01 15:23 +0000 [r198558]  David Vossel <dvossel@digium.com>

	* main/threadstorage.c: Fixed an issue in the threadstorage cli
	  functions resulting from the constification of struct
	  ast_cli_args in r196072.

2009-06-01 14:45 +0000 [r198500-198530]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Remove extra lock from app_queue.

	* channels/chan_local.c: Remove extra lock from local_indicate in
	  connected line case. Oh, and this fixes a deadlock I was seeing.

	* channels/chan_local.c: Add missing unlock of local pvt.

	* channels/chan_agent.c: Remove documentation for the 'exten'
	  argument to the AGENT function. Since AgentCallbackLogin has been
	  removed, this should not be documented any more.

2009-06-01 13:31 +0000 [r198498]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where the Event and Content-Type
	  headers were added twice to outgoing SIP NOTIFY messages. (closes
	  issue #15239) Reported by: pj

2009-05-31 17:52 +0000 [r198470]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_strings.c: Fix documentation for FIELDQTY.

2009-05-31 02:09 +0000 [r198442]  Eliel C. Sardanons <eliels@gmail.com>

	* main/Makefile: Filter the say.o object, it is being added later.

2009-05-31 01:40 +0000 [r198438]  Russell Bryant <russell@digium.com>

	* main/manager.c: Constification and remove some unused code.

2009-05-31 01:22 +0000 [r198437]  Eliel C. Sardanons <eliels@gmail.com>

	* res/res_timing_dahdi.c: Avoid a crash when res_timing_dahdi is
	  unloaded but wasn't properly loaded. if dahdi_test_timer() fails,
	  timing_funcs_handle remains NULL causing a crash when calling
	  ast_unregister_timing_interface() with a NULL pointer. (closes
	  issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
	  uploaded by eliel (license 64)

2009-05-31 01:19 +0000 [r198434]  Russell Bryant <russell@digium.com>

	* main/channel.c, include/asterisk/channel.h: Constify the
	  ast_frame arg to ast_queue_frame().

2009-05-30 20:11 +0000 [r198371-198375]  Sean Bright <sean@malleable.com>

	* res/res_jabber.c: Properly terminate the receive buffer before
	  sending to iksemel. aji_io_recv takes the maximum number of bytes
	  to read (instead of the total buffer size), so we have to
	  subtract 1 from our buffer size. Without this, when we receive
	  packets that are larger than our buffer, iksemel will choke and
	  things get wonky. (closes issue #15232) Reported by: lp0 Patches:
	  05302009_res_jabber.c.patch uploaded by seanbright (license 71)
	  Tested by: seanbright, lp0

	* /, res/res_jabber.c: Merged revisions 198370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
	  2009) | 12 lines Properly terminate AMI JabberSend response
	  messages. The response message (either Error or Success) needs an
	  extra trailing \r\n after the fields to inform the client that
	  the message is complete. (closes issue #14876) Reported by: srt
	  Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
	  (license 71) asterisk_14876.patch uploaded by srt (license 378)
	  trunk-14876-2.diff uploaded by phsultan (license 73) ........

2009-05-30 03:43 +0000 [r198312]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c, /: Merged revisions 198311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
	  | 5 lines Fix a crash that occurred when MWI SMDI messages
	  expired. (closes issue #14561) Reported by: cmoss28 ........

2009-05-30 03:26 +0000 [r198285]  Sean Bright <sean@malleable.com>

	* apps/app_dial.c, /: Merged revisions 198251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
	  2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
	  treat a missing one. (closes issue #15056) Reported by:
	  p_lindheimer Patches: 05292009_bug15056.diff uploaded by
	  seanbright (license 71) Tested by: p_lindheimer ........

2009-05-30 02:31 +0000 [r198248]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When removing all packets from a dialog we
	  also need to free the data if present.

2009-05-30 01:04 +0000 [r198217]  Eliel C. Sardanons <eliels@gmail.com>

	* configs/agents.conf.sample, channels/chan_agent.c: Remove not
	  used code in the Agent channel. This code was there because of
	  the AgentCallbackLogin() application. ->loginchan[] member was
	  only used by AgentCallbackLogin(). Agent where dumped to astdb if
	  they where logged in using AgentCallbacklogin() so they are not
	  being dumper anymore. Review:
	  https://reviewboard.asterisk.org/r/267/

2009-05-29 23:04 +0000 [r198183-198186]  Russell Bryant <russell@digium.com>

	* configs/modules.conf.sample: Suggesting that only a single timing
	  module be loaded is no longer necessary.

	* res/res_timing_pthread.c: Improve handling of trying to ACK too
	  many timer expirations.

2009-05-29 22:21 +0000 [r198182]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.c: Add a couple of TODO items so I don't forget

2009-05-29 20:06 +0000 [r198146]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c: Resolve issues with choppy sound when
	  using res_timing_pthread. The situation that caused this problem
	  was when continuous mode was being turned on and off while a rate
	  was set for a timing interface. A very easy way to replicate this
	  bug was to do a Playback() from behind a Local channel. In this
	  scenario, a rate gets set on the channel for doing file playback.
	  At the same time, continuous mode gets turned on and off about
	  every 20 ms as frames get queued on to the PBX side channel from
	  the other side of the Local channel. Essentially, this module
	  treated continuous mode and a set rate as mutually exclusive
	  states for the timer to be in. When I dug deep enough, I observed
	  the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
	  almost 20 ms ... 3) Continuous mode gets turned on for a queued
	  up frame 4) Continuous mode gets turned off 5) The timer goes
	  back to its tick per 20 ms. state but starts counting at 0 ms. 6)
	  Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
	  and produce a timer tick, but not most of the time. This is what
	  produced the choppy sound (or sometimes no sound at all). Now,
	  the module treats continuous mode and a set rate as completely
	  independent timer modes. They can be enabled and disabled
	  independently of each other and things work as expected. (closes
	  issue #14412) Reported by: dome Patches: issue14412.diff.txt
	  uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
	  uploaded by russell (license 2) Tested by: DennisD, russell

2009-05-29 19:46 +0000 [r198139]  Eliel C. Sardanons <eliels@gmail.com>

	* main/Makefile: Simplify the Makefile and avoid needing to specify
	  each object file. Instead of specifying every object file, use
	  make's magic to generate it. This will generate less conflicts in
	  team branches when a new file is added in trunk. (closes issue
	  #15226) Reported by: eliel Patches: makefile uploaded by eliel
	  (license 64) Review: http://reviewboard.asterisk.org/r/269/

2009-05-29 19:19 +0000 [r198088]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, channels/sig_analog.c (added),
	  channels/sig_analog.h (added), channels/Makefile: New signaling
	  module to handle analog operations in chan_dahdi This branch
	  splits all the analog signaling logic out of chan_dahdi.c into
	  sig_analog.c. Functionality in theory should not change at all.
	  As noted in the code, there is still some unused code remaining
	  that will be cleaned up in a later commit. Review:
	  https://reviewboard.asterisk.org/r/253/

2009-05-29 19:18 +0000 [r198083]  Eliel C. Sardanons <eliels@gmail.com>

	* CREDITS: Apply anti-spam obfuscation to an email address.

2009-05-29 19:04 +0000 [r198072]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
	  revisions 198068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
	  2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
	  the default CDR disposition. This change also involves the
	  addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
	  originated channels to distinguish: them from dialed channels.
	  (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
	  uploaded by mnicholson (license 96) Tested by: mnicholson,
	  dbrooks (closes issue #15122) Reported by: sum Tested by: sum
	  ........

2009-05-29 18:39 +0000 [r198064]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Fix a memory leak of the write buffer when writing a
	  file.

2009-05-29 18:15 +0000 [r198000]  Sean Bright <sean@malleable.com>

	* Makefile, /: Merged revisions 197998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
	  2009) | 8 lines Fix 'make config' target for Slackware. There was
	  a missing semi-colon after the echo statement in the Makefile
	  that was causing problems for some users. Fix suggested by
	  reporter. (closes issue #15225) Reported by: pdavis ........

2009-05-29 17:51 +0000 [r197996]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where the default setting did not
	  perform a remote bridge when it should have.

2009-05-29 16:15 +0000 [r197960]  Russell Bryant <russell@digium.com>

	* res/res_timing_pthread.c: Trim trailing whitespace so that I can
	  work on this bug without it bothering me. :-)

2009-05-29 15:48 +0000 [r197959]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: A few fixes to SIP with regards to connected
	  line updates during transfers. * Set the invitestate to
	  INV_CALLING when we send a connected line reinvite. This prevents
	  us from potentially rapid-firing reinvites to a single peer. *
	  Use the astdb to store a peer's allowed methods. This prevents us
	  from sending an UPDATE during the interval between startup and
	  the peer's first registration if the peer does not support the
	  UPDATE method. * Handle Polycom's method of indicating allowed
	  methods in REGISTER. Instead of using an Allow header, they place
	  the allowed methods in a methods= parameter in the Contact
	  header. ABE-1873

2009-05-29 05:15 +0000 [r197926]  Terry Wilson <twilson@digium.com>

	* doc/tex/asterisk.tex, doc/tex/calendaring.tex (added): Add some
	  TeX docs for calendaring. I still need to set up tests to make
	  sure my examples are completely correct, but I ran out of time
	  tonight and felt that they at least would give an idea as to how
	  to use calendaring. I will try to test the examples and do some
	  cleanup on the docs tomorrow night.

2009-05-28 22:42 +0000 [r197861]  Sean Bright <sean@malleable.com>

	* include/asterisk/doxygen/releases.h, sounds/Makefile: Update
	  references to downloads.digium.com to its new URL.

2009-05-28 22:04 +0000 [r197828]  Leif Madsen <lmadsen@digium.com>

	* apps/app_mixmonitor.c: Update documentation in MixMonitor.
	  Updated the MixMonitor documentation for the 'b' option so that
	  it is more obvious that you must not optimize away the Local
	  channel when using this option. (closes issue #14829) Reported
	  by: licedey Tested by: mmichelson, licedey, lmadsen

2009-05-28 21:50 +0000 [r197824]  Sean Bright <sean@malleable.com>

	* doc/CODING-GUIDELINES, doc/asterisk.8, BUGS, doc/backtrace.txt,
	  doc/tex/mp3.tex, channels/h323/README, main/enum.c,
	  doc/tex/misdn.tex, include/asterisk/doxyref.h,
	  contrib/scripts/ast_grab_core, doc/tex/backtrace.tex,
	  include/asterisk/doxygen/reviewboard.h,
	  include/asterisk/doxygen/commits.h,
	  contrib/scripts/asterisk.ldif,
	  contrib/scripts/asterisk.ldap-schema,
	  configs/extensions.conf.sample, doc/asterisk.sgml: Update
	  references to bugs.digium.com and reviewboard.digium.com to the
	  new URLs.

2009-05-28 20:43 +0000 [r197777]  Terry Wilson <twilson@digium.com>

	* configs/calendar.conf.sample: Make note of Exchange calendar
	  support limitations

2009-05-28 20:36 +0000 [r197775]  Kevin P. Fleming <kpfleming@digium.com>

	* main/utils.c: Ensure that accidental calls to
	  ast_string_field_free_memory() on embedded stringfield pools are
	  safe. It is possible for a stringfield manager structure (and
	  pool) structure to be allocated as part of a larger structure
	  allocation (using ast_calloc_with_strinfields()); when this is
	  done, the stringfield pool cannot be separately freed, but users
	  of the tructure may not be aware (and shouldn't have to be aware)
	  of whether the pool was embedded. This patch modifies the
	  behavior so that they can always call
	  ast_string_field_free_memory() and the function will do the right
	  thing for both embedded and non-embedded situations.

2009-05-28 20:17 +0000 [r197740]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Treat 405 responses the same way we would a
	  501. This makes sure that we mark a method as being unallowed if
	  we receive a 405 response so that we don't continue to try to
	  send that same type of message.

2009-05-28 19:57 +0000 [r197738]  Terry Wilson <twilson@digium.com>

	* res/res_calendar.exports (added), res/res_calendar_exchange.c
	  (added), res/res_calendar_icalendar.c (added),
	  build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  configs/calendar.conf.sample (added), res/res_calendar_caldav.c
	  (added), include/asterisk/calendar.h (added), makeopts.in,
	  res/res_calendar.c (added): Add Calendaring support for Asterisk
	  This commit add Calendaring support to Asterisk for iCalendar,
	  CalDAV, and MS Exchange calendars. Exchange support has only been
	  tested on Exchange Server 2k3 and does not support forms-based
	  authentication at this time (patches *very* welcome). Exchange
	  support is also currently missing the ability to return a list of
	  a meting's attendees (again, patches are very, very welcome).
	  Features include: Querying a calendar for events over a specific
	  time range Checking a calendar's busy status via the dialplan
	  Writing calendar events via the dialplan (CalDAV and Exchange
	  only) Handling calendar event notifications through the dialplan
	  (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash
	  Review: https://reviewboard.asterisk.org/r/58

2009-05-28 18:48 +0000 [r197701]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Add missing lock to local_indicate
	  function for connected line frames.

2009-05-28 18:45 +0000 [r197697]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Fix a bug where the trunkmtu setting was
	  not set to the default value of 1240 on load but was on reload.

2009-05-28 16:01 +0000 [r197621]  Eliel C. Sardanons <eliels@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 197562 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
	  13 lines Use the address we already know when reloading a peer
	  with nat=yes. If we already have an address for a peer, and we
	  are reloading the sip configuration, try to use that address to
	  contact the peer, instead of getting it from the Contact. (closes
	  issue #15194) Reported by: ibc Patches: sip.patch uploaded by
	  eliel (license 64) Tested by: manwe ........

2009-05-28 15:35 +0000 [r197616]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, channels/chan_console.c, apps/app_rpt.c,
	  main/astobj2.c, main/cli.c: Eliminate several needless checks and
	  fix a few memory leaks (closes issue #14833) Reported by:
	  contactmayankjain Patches: all_changes.patch uploaded by
	  contactmayankjain (license 740) slightly modified by me

2009-05-28 15:32 +0000 [r197606]  Mark Michelson <mmichelson@digium.com>

	* /: Recorded merge of revisions 197588 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May
	  2009) | 16 lines Allow for media to arrive from an alternate
	  source when responding to a reinvite with 491. When we receive a
	  SIP reinvite, it is possible that we may not be able to process
	  the reinvite immediately since we have also sent a reinvite out
	  ourselves. The problem is that whoever sent us the reinvite may
	  have also sent a reinvite out to another party, and that reinvite
	  may have succeeded. As a result, even though we are not going to
	  accept the reinvite we just received, it is important for us to
	  not have problems if we suddenly start receiving RTP from a new
	  source. The fix for this is to grab the media source information
	  from the SDP of the reinvite that we receive. This information is
	  passed to the RTP layer so that it will know about the alternate
	  source for media. Review: https://reviewboard.asterisk.org/r/252
	  ........

2009-05-28 15:23 +0000 [r197570]  Joshua Colp <jcolp@digium.com>

	* main/logger.c: Fix an incorrect call to
	  ast_string_field_free_memory which caused a crash in the logger.
	  Since the message structure is allocated using
	  ast_calloc_with_stringfields we do not need to free the memory
	  used for the stringfields as it will get freed when the message
	  structure is.

2009-05-28 14:58 +0000 [r197543]  Mark Michelson <mmichelson@digium.com>

	* /, include/asterisk/audiohook.h, main/audiohook.c,
	  apps/app_chanspy.c: Merged revisions 197537 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
	  2009) | 21 lines Add flags to chanspy audiohook so that audio
	  stays in sync. There are two flags being added to the chanspy
	  audiohook here. One is the pre-existing
	  AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
	  the read and write slinfactories on the audiohook do not skew
	  beyond a certain tolerance. In addition, there is a new audiohook
	  flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
	  we do not allow for a slinfactory to build up a substantial
	  amount of audio before flushing it. For this particular issue,
	  this means that the person spying on the call will hear the
	  conversations in real time with very little delay in the audio.
	  (closes issue #13745) Reported by: geoffs Patches: 13745.patch
	  uploaded by mmichelson (license 60) Tested by: snblitz ........

2009-05-28 14:51 +0000 [r197538]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Fix a bug in stringfields where it did not actually
	  free the pools of memory. (closes issue #15074) Reported by: pj

2009-05-28 14:39 +0000 [r197528-197535]  Sean Bright <sean@malleable.com>

	* configs/amd.conf.sample, configs/users.conf.sample,
	  configs/gtalk.conf.sample, configs/rpt.conf.sample,
	  configs/rtp.conf.sample, configs/cli_aliases.conf.sample,
	  configs/modules.conf.sample, configs/phone.conf.sample,
	  configs/extensions.ael.sample, configs/skinny.conf.sample,
	  configs/ais.conf.sample, configs/meetme.conf.sample,
	  configs/extensions_minivm.conf.sample, configs/telcordia-1.adsi,
	  configs/alsa.conf.sample, configs/iax.conf.sample,
	  configs/followme.conf.sample, configs/mgcp.conf.sample,
	  configs/sip.conf.sample, configs/extensions.lua.sample,
	  configs/say.conf.sample, configs/queuerules.conf.sample,
	  configs/minivm.conf.sample, configs/osp.conf.sample,
	  configs/chan_dahdi.conf.sample,
	  configs/cli_permissions.conf.sample, configs/console.conf.sample,
	  configs/dundi.conf.sample, configs/indications.conf.sample,
	  configs/oss.conf.sample, configs/queues.conf.sample,
	  configs/voicemail.conf.sample, configs/usbradio.conf.sample,
	  configs/cdr.conf.sample, configs/jingle.conf.sample,
	  configs/misdn.conf.sample, configs/manager.conf.sample,
	  configs/festival.conf.sample, configs/features.conf.sample,
	  configs/logger.conf.sample, configs/http.conf.sample,
	  configs/h323.conf.sample, configs/sla.conf.sample,
	  configs/phoneprov.conf.sample, configs/res_odbc.conf.sample,
	  configs/agents.conf.sample, configs/alarmreceiver.conf.sample,
	  configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
	  configs/jabber.conf.sample, configs/extconfig.conf.sample,
	  configs/res_snmp.conf.sample, configs/iaxprov.conf.sample,
	  configs/unistim.conf.sample, configs/dnsmgr.conf.sample,
	  configs/extensions.conf.sample, configs/asterisk.adsi: Remove a
	  bunch of trailing whitespace in preparation for
	  reformatting/cleanup. Let's try that again, this time removing
	  trailing whitespace and not leading whitespace. I can't believe
	  no one noticed.

	* configs/amd.conf.sample, configs/gtalk.conf.sample,
	  configs/rtp.conf.sample, configs/rpt.conf.sample,
	  configs/cli_aliases.conf.sample, configs/extensions.ael.sample,
	  configs/skinny.conf.sample, configs/meetme.conf.sample,
	  configs/telcordia-1.adsi, configs/alsa.conf.sample,
	  configs/iax.conf.sample, configs/mgcp.conf.sample,
	  configs/extensions.lua.sample, configs/sip.conf.sample,
	  configs/say.conf.sample, configs/minivm.conf.sample,
	  configs/console.conf.sample, configs/cli_permissions.conf.sample,
	  configs/chan_dahdi.conf.sample, configs/oss.conf.sample,
	  configs/queues.conf.sample, configs/jingle.conf.sample,
	  configs/usbradio.conf.sample, configs/voicemail.conf.sample,
	  configs/misdn.conf.sample, configs/manager.conf.sample,
	  configs/features.conf.sample, configs/h323.conf.sample,
	  configs/sla.conf.sample, configs/res_odbc.conf.sample,
	  configs/phoneprov.conf.sample, configs/alarmreceiver.conf.sample,
	  configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
	  configs/jabber.conf.sample, configs/unistim.conf.sample,
	  configs/dnsmgr.conf.sample, configs/extensions.conf.sample,
	  configs/asterisk.adsi: Remove a bunch of trailing whitespace in
	  preparation for reformatting/cleanup.

2009-05-28 13:47 +0000 [r197467]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 197466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
	  lines Fix a bug where the flag indicating the presence of rport
	  would get overwritten by the nat setting. The presence of rport
	  is now stored as a separate flag. Once the dialog is setup and
	  authenticated (or it passes through unauthenticated) the proper
	  nat flag is set. (closes issue #13823) Reported by: dimas
	  ........

2009-05-28 11:25 +0000 [r197406-197431]  Gavin Henry <ghenry@suretecsystems.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: Added AstVoicemailContext Added
	  AstVoicemailContext (closes issue #15155) Reported by: scramatte
	  Tested by: suretec

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: New objectclass AsteriskVoiceMail
	  and AstAccountCallLimit attribute Added new ObjectClass
	  AsteriskVoiceMail, and AstAccountCallLimit attribute and cleaned
	  up formatting and tested with OpenLDAP (closes issue #15155)
	  Reported by: scramatte Patches: asterisk.schema uploaded by
	  scramatte (license 796) Tested by: suretec Review: [full review
	  board URL with trailing slash]

	* doc/ldap.txt, configs/res_ldap.conf.sample,
	  contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif: closes issue #15156

2009-05-27 23:48 +0000 [r197374]  Tilghman Lesher <tlesher@digium.com>

	* main/xml.c: Revert commit 192032. This define is needed on Mac OS
	  X.

2009-05-27 22:42 +0000 [r197338]  Russell Bryant <russell@digium.com>

	* main/rtp_engine.c: Don't do a pointer comparison before setting
	  the remote address.

2009-05-27 22:21 +0000 [r197335]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/agi.h: Ensure that this header includes
	  xmldoc.h, since it depends on it.

2009-05-27 20:14 +0000 [r197266]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Adding some generic handling of error codes
	  sent to us in replys to requests. Previously they always set
	  hangupcause 0, which is generally wrong. With this change, we're
	  setting some generic hangup causes. For 5xx errors, which
	  indicate some sort of problem with the remote server, we're now
	  setting CONGESTION. EDVX002

2009-05-27 20:08 +0000 [r197260]  Sean Bright <sean@malleable.com>

	* Makefile: Use bash explicitly when calling
	  build_tools/mkpkgconfig from the Makefile. Since we use bashisms
	  in build_tools/mkpkgconfig, we should call on bash explicitly
	  when running from the Makefile, otherwise we get errors during a
	  'make install.' (closes issue #15209) Reported by: seandarcy

2009-05-27 19:20 +0000 [r197209]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_cut.c: Recorded merge of revisions 197194 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
	  | 5 lines Use a different determinator on whether to print the
	  delimiter, since leading fields may be blank. (closes issue
	  #15208) Reported by: ramonpeek Patch by me, though inspired in
	  part by a patch from ramonpeek ........

2009-05-27 18:25 +0000 [r196948-197189]  Sean Bright <sean@malleable.com>

	* configs/adtranvofr.conf.sample (removed): Remove a file sample
	  configuration file that is no longer used.

	* configs/chan_dahdi.conf.sample, configs/vpb.conf.sample,
	  configs/smdi.conf.sample, configs/extensions.conf.sample,
	  configs/sla.conf.sample: Fix references to /etc/dahdi/system.conf
	  and /etc/asterisk/chan_dahdi.conf in the sample configuration
	  files. (closes issue #15207) Reported by: seandarcy

	* channels/chan_alsa.c: Display an error message when chan_alsa
	  fails to load due to a missing or inaccessible configuration
	  file. Before this change, when chan_alsa failed to load due to a
	  missing or inaccessible configuration file, no message would be
	  displayed. With this change, when chan_alsa fails to load due to
	  a missing or inaccessible configuration file, a message will be
	  displayed. (closes issue #14760) Reported by: Nick_Lewis Patches:
	  chan_alsa.c-confload.patch uploaded by Nick (license 657)

	* main/xmldoc.c: Reset the terminal to the correct fg/bg after XML
	  documenation is rendered. (closes issue #15200) Reported by:
	  ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
	  (license 71) Tested by: ajohnson

2009-05-26 22:40 +0000 [r196946]  Russell Bryant <russell@digium.com>

	* autoconf/ast_check_osptk.m4 (added), configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Update configure
	  script to check for OSP toolkit 3.5.0. (closes issue #14988)
	  Reported by: tzafrir Patches: configure.ac.diff uploaded by
	  homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick
	  (license 91)

2009-05-26 22:38 +0000 [r196907-196945]  Sean Bright <sean@malleable.com>

	* main/manager.c: Add ActionID to CoreShowChannel event. There is
	  inconsistency in how we handle manager responses that are lists
	  of items and, unfortunately, third parties have come to rely on
	  ActionID being on every event within those lists instead of just
	  keeping track of the ActionID for the current response. This
	  change makes CoreShowChannels include the ActionID with each
	  CoreShowChannel event generated as a result of it being called.
	  (closes issue #15001) Reported by: sum Patches:
	  patchactionid2.patch uploaded by sum (license 766)

	* main/manager.c: Include startup and reload date in the CoreStatus
	  manager message. The CoreStartupTime and CoreReloadTime
	  name/value pairs in the CoreStatus response message only included
	  the time and not the date. This patch, inspired by the reporter's
	  patch, adds 2 new fields - CoreStartupDate and CoreReloadDate -
	  which contain the date portion of these values. (closes issue
	  #15000) Reported by: sum

2009-05-26 19:50 +0000 [r196893]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, apps/app_directed_pickup.c: Remove some
	  redundant or unnecessary connected line-related function calls.

2009-05-26 18:20 +0000 [r196843]  Russell Bryant <russell@digium.com>

	* /, res/res_convert.c: Merged revisions 196826 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
	  | 9 lines Resolve a file handle leak. The frames here should have
	  always been freed. However, out of luck, there was never any
	  memory leaked. However, after file streams became reference
	  counted, this code would leak the file stream for the file being
	  read. (closes issue #15181) Reported by: jkroon ........

2009-05-26 16:38 +0000 [r196725-196792]  Sean Bright <sean@malleable.com>

	* apps/app_queue.c: Add a missing unref for queues in
	  handle_statechange.

	* main/pbx.c, include/asterisk/pbx.h, res/res_clioriginate.c: Add
	  new ast_complete_applications function so that we can use it with
	  the 'channel originate ... application <app>' CLI command. (And
	  yeah, I cleaned up some whitespace in res_clioriginate.c... big
	  whoop, wanna fight about it!?)

	* cdr/cdr_sqlite3_custom.c: Use a properly allocated channel for
	  substitution in cdr_sqlite3_custom.

2009-05-26 13:43 +0000 [r196658-196721]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where the sip unregister CLI
	  command did not completely unregister the peer. (closes issue
	  #15118) Reported by: alecdavis Patches:
	  chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585)

	* /, contrib/scripts/safe_asterisk: Merged revisions 196657 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
	  lines Remove some bash specific stuff from safe_asterisk. (closes
	  issue #10812) Reported by: paravoid Patches:
	  safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
	  ........

2009-05-26 12:14 +0000 [r196622]  Sean Bright <sean@malleable.com>

	* cdr/cdr_manager.c: Use a properly allocated channel for
	  substitution in cdr_manager.

2009-05-24 16:17 +0000 [r196554-196585]  Eliel C. Sardanons <eliels@gmail.com>

	* res/res_agi.c: Move AGI static documentation to the new AstXML
	  form. Move AGI commands documentation to XML docs: 'set priority'
	  'set variable' 'stream file' 'control stream file' 'tdd mode'
	  'verbose' 'wait for digit' 'speech create' 'speech set' 'speech
	  destroy' 'speech load grammar' 'speech unload grammar' 'speech
	  activate grammar' 'speech deactivate grammar' 'speech recognize'

	* res/res_agi.c: Move static AGI commands documentation to XML.
	  Move AGI commands ('say datetime', 'send image', 'send text',
	  'set autohangup', 'set callerid', 'set context', 'set extension')
	  documentation to the AstXML form.

2009-05-23 15:16 +0000 [r196520]  Sean Bright <sean@malleable.com>

	* cdr/cdr_custom.c: Fix errors in cdr_custom that cause reference
	  errors when non-CDR variable substitution is done. cdr_custom was
	  creating a ast_channel struct directly and passing it into the
	  core for variable substition. This was fine as long as the format
	  string contained only calls to the CDR() function. Doing
	  something like ${EPOCH} on the other hand tried to lock the
	  channel, which would fail and throw an error because the passed
	  channel hadn't been allocated as an ao2 object. So now we create
	  the dummy channel with ast_channel_alloc, and everything works as
	  expected.

2009-05-23 13:31 +0000 [r196488]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/cli.h: Correct example for CLI autocompletion
	  (generation) Reported by Atis on #asterisk-dev

2009-05-23 04:27 +0000 [r196456]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c: set MFCR2_CATEGORY just when starting the
	  pbx

2009-05-22 21:11 +0000 [r196417]  Sean Bright <sean@malleable.com>

	* main/asterisk.c: Call ast_stun_init() when we're initializing to
	  get the 'stun debug set' commands.

2009-05-22 21:09 +0000 [r196416]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: SIP set outbound
	  transport type from Registration In sip.conf the transport option
	  allows for the configuration of what transport types (udp, tcp,
	  and tls) a peer will accept, but only the first type listed was
	  used for outbound connections. This patch changes this. Now the
	  default transport type is only used until the peer registers.
	  When registration takes place the transport type is parsed out of
	  the Contact header. If the Contact header's transport type is
	  equal to one that the peer supports, the peer's default transport
	  type for outbound connections is set to match the Contact
	  header's type. If the Contact header's transport type is not
	  present, then the peer's default transport type is set to match
	  the one the peer registered with. When a peer unregisters or the
	  registration expires, the default transport type for that peer is
	  reset. (closes issue #12282) Reported by: rjain Patches:
	  reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
	  dvossel (closes issue #14727) Reported by: pj Patches:
	  reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
	  dvossel Review: https://reviewboard.asterisk.org/r/249/

2009-05-22 20:01 +0000 [r196381]  Sean Bright <sean@malleable.com>

	* channels/chan_gtalk.c: Don't crash if an RTP instance can't be
	  created. This could occur when an invalid bindaddr was specified
	  in gtalk.conf.

2009-05-22 19:38 +0000 [r196308-196377]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_minivm.c: Unregister every registered application by
	  MiniVM. The MinivmMWI application was not being unregistered on
	  unload and we were not able to load again the module or reload
	  it. (closes issue #15174) Reported by: junky Patches:
	  unregister_minivm_mwi.diff uploaded by junky (license 177)

	* res/res_agi.c: Moved static documentation to the AstXML form.
	  Moved AGI commands static documentation to XML docs ('say alpha',
	  'say digits', 'say number', 'say phonetic', 'say date' and 'say
	  time').

	* main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
	  channels/chan_agent.c, apps/app_queue.c, channels/chan_iax2.c,
	  include/asterisk/manager.h, channels/chan_dahdi.c,
	  main/manager.c, channels/chan_skinny.c, main/features.c,
	  res/res_agi.c, include/asterisk/xmldoc.h, include/asterisk/pbx.h,
	  apps/app_senddtmf.c, doc/appdocsxml.dtd, main/db.c,
	  main/xmldoc.c, apps/app_voicemail.c: Implement a new element in
	  AstXML for AMI actions documentation. A new xml element was
	  created to manage the AMI actions documentation, using AstXML. To
	  register a manager action using XML documentation it is now
	  possible using ast_manager_register_xml(). The CLI command
	  'manager show command' can be used to show the parsed
	  documentation. Example manager xml documentation: <manager
	  name="ami action name" language="en_US"> <synopsis> AMI action
	  synopsis. </synopsis> <syntax> <xi:include
	  xpointer="xpointer(...)" /> <-- for ActionID <parameter
	  name="header1" required="true"> <para>Description</para>
	  </parameter> ... </syntax> <description> <para>AMI action
	  description</para> </description> <see-also> ... </see-also>
	  </manager>

2009-05-22 16:53 +0000 [r196272]  Tilghman Lesher <tlesher@digium.com>

	* main/astmm.c: Two more minor fixes due to constification

2009-05-22 16:51 +0000 [r196270]  Sean Bright <sean@malleable.com>

	* res/res_agi.c: Fix res_agi compilation after the const-ify the
	  world merge. Since we are dealing with a 'const char * const'
	  now, we have to create a temporary copy of the string to work on
	  rather than the original. Fix inspired by reporter. Reviewed by
	  everyone-and-their-mother in #asterisk-dev. (closes issue #15184)
	  Reported by: andrew

2009-05-22 16:50 +0000 [r196268]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: s/it's/its/

2009-05-22 16:20 +0000 [r196246]  Russell Bryant <russell@digium.com>

	* channels/chan_dahdi.c: resolve compiler warning

2009-05-22 16:10 +0000 [r196227]  Sean Bright <sean@malleable.com>

	* channels/chan_dahdi.c, main/pbx.c, res/res_jabber.c,
	  res/res_monitor.c: Fix build under dev mode and remove some casts
	  that are no longer necessary as a result of the const-ify the
	  world patch.

2009-05-22 15:07 +0000 [r196187-196188]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_mp3.c: Fix constify the world compile problem.

	* channels/chan_misdn.c: Make chan_misdn compile.

2009-05-22 13:56 +0000 [r196117]  Joshua Colp <jcolp@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 196116 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May
	  2009) | 5 lines Fix a bug where using immediate with mISDN caused
	  a cause code of 16 to get sent back instead of 1 if the 's'
	  extension did not exist. (closes issue #12286) Reported by:
	  lmamane ........

2009-05-22 13:34 +0000 [r196114]  Eliel C. Sardanons <eliels@gmail.com>

	* main/pbx.c: Avoid using prototypes when not necessary (it is
	  already defined in the header file). Make log_match_char_tree()
	  static to main/pbx.c (only used there).

2009-05-21 21:13 +0000 [r196072]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dahdibarge.c, main/frame.c, apps/app_record.c,
	  apps/app_playtones.c, funcs/func_strings.c,
	  include/asterisk/extconf.h, apps/app_alarmreceiver.c,
	  apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
	  channels/chan_iax2.c, main/astobj2.c, channels/chan_dahdi.c,
	  channels/chan_skinny.c, apps/app_dumpchan.c, pbx/pbx_ael.c,
	  main/pbx.c, channels/vcodecs.c, apps/app_softhangup.c,
	  apps/app_morsecode.c, apps/app_talkdetect.c,
	  channels/iax2-parser.c, apps/app_db.c, apps/app_speech_utils.c,
	  apps/app_sendtext.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
	  main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
	  apps/app_dictate.c, apps/app_authenticate.c,
	  apps/app_readexten.c, apps/app_userevent.c, res/res_jabber.c,
	  include/asterisk/abstract_jb.h, main/channel.c,
	  apps/app_setcallerid.c, apps/app_osplookup.c, funcs/func_odbc.c,
	  apps/app_mp3.c, apps/app_minivm.c, apps/app_directory.c,
	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
	  apps/app_read.c, channels/chan_sip.c,
	  include/asterisk/taskprocessor.h, include/asterisk/cli.h,
	  apps/app_originate.c, utils/conf2ael.c,
	  apps/app_channelredirect.c, apps/app_forkcdr.c,
	  main/abstract_jb.c, channels/misdn/chan_misdn_config.h,
	  apps/app_sms.c, utils/extconf.c, funcs/func_devstate.c,
	  apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
	  include/asterisk/agi.h, cdr/cdr_sqlite3_custom.c,
	  apps/app_readfile.c, apps/app_sayunixtime.c, apps/app_test.c,
	  include/asterisk/speech.h, cdr/cdr_adaptive_odbc.c,
	  apps/app_image.c, main/taskprocessor.c, main/loader.c,
	  main/cli.c, apps/app_skel.c, include/asterisk/module.h,
	  main/features.c, apps/app_amd.c, channels/chan_alsa.c,
	  apps/app_url.c, apps/app_externalivr.c, formats/format_gsm.c,
	  apps/app_milliwatt.c, res/res_speech.c, main/ast_expr2.fl,
	  apps/app_dial.c, include/asterisk/utils.h, apps/app_page.c,
	  apps/app_privacy.c, apps/app_fax.c, apps/app_echo.c,
	  channels/chan_agent.c, apps/app_dahdiras.c, apps/app_disa.c,
	  pbx/dundi-parser.c, apps/app_transfer.c, res/res_monitor.c,
	  apps/app_playback.c, include/asterisk/app.h,
	  channels/chan_misdn.c, apps/app_waitforring.c,
	  include/asterisk/image.h, apps/app_macro.c,
	  apps/app_zapateller.c, apps/app_chanspy.c, apps/app_cdr.c,
	  channels/chan_unistim.c, apps/app_meetme.c, main/utils.c,
	  res/res_musiconhold.c, apps/app_followme.c,
	  channels/misdn_config.c, apps/app_controlplayback.c, main/ulaw.c,
	  main/cdr.c, main/manager.c, channels/console_gui.c,
	  cdr/cdr_sqlite.c, res/res_agi.c, main/app.c,
	  apps/app_confbridge.c, main/image.c, apps/app_ivrdemo.c,
	  apps/app_parkandannounce.c, res/res_clioriginate.c,
	  apps/app_jack.c, apps/app_while.c, res/res_rtp_asterisk.c,
	  apps/app_nbscat.c, apps/app_festival.c, res/res_limit.c,
	  apps/app_waitforsilence.c, apps/app_waituntil.c,
	  channels/chan_console.c, apps/app_queue.c, apps/app_system.c,
	  apps/app_getcpeid.c, channels/chan_oss.c,
	  include/asterisk/features.h, apps/app_flash.c,
	  apps/app_directed_pickup.c, channels/chan_nbs.c,
	  include/asterisk/strings.h, include/asterisk/pbx.h,
	  apps/app_senddtmf.c: Const-ify the world (or at least a good part
	  of it) This patch adds 'const' tags to a number of Asterisk APIs
	  where they are appropriate (where the API already demanded that
	  the function argument not be modified, but the compiler was not
	  informed of that fact). The list includes: - CLI command handlers
	  - CLI command handler arguments - AGI command handlers - AGI
	  command handler arguments - Dialplan application handler
	  arguments - Speech engine API function arguments In addition,
	  various file-scope and function-scope constant arrays got 'const'
	  and/or 'static' qualifiers where they were missing. Review:
	  https://reviewboard.asterisk.org/r/251/

2009-05-21 19:11 +0000 [r195995]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 195991 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21
	  May 2009) | 14 lines Sign problem calculating timestamp for iax
	  frame leads to no audio on the receiving peer. There are rare
	  cases in which a frame's delivery timestamp is slightly less than
	  the iax2_pvt's offset. This causes the pvt's timestamp to be a
	  small negative number, but since the timestamp value is unsigned
	  it looks like a huge positive number. This patch checks for this
	  negative case and sets the ms to zero. A similar check is already
	  done right below this one in the 'else' statement. (closes issue
	  #15032) Reported by: guillecabeza Patches:
	  chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
	  380) Tested by: guillecabeza (closes issue #14216) Reported by:
	  Andrey Sofronov ........

2009-05-21 19:06 +0000 [r195992]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Pass connected line updates along during a
	  bridge.

2009-05-21 17:15 +0000 [r195949]  Sean Bright <sean@malleable.com>

	* configs/cdr_custom.conf.sample: Rework the cdr_custom.conf.sample
	  header a bit to reflect the changes in functionality (allowing
	  multiple mappings).

2009-05-21 15:33 +0000 [r195882]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195881
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
	  2009) | 13 lines This commit prevents cdr records with
	  AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
	  in certain cases. This is accomplished by adding two functions to
	  update the answer time and disposition of calls that checks for
	  the proper lock flags. These functions are used in the
	  ast_bridge_call() function so that ForkCDR(A) calls are
	  respected. This patch also modifies the way ast_bridge_call()
	  chooses the cdr record to base the bridged_cdr on. Previously the
	  first unlocked cdr record would be chosen, now instead the first
	  cdr record is chosen and forked cdr records are moved to the
	  bridge_cdr. This allows the original cdr record and any forked
	  cdr records to be properly updated with answer and end times.
	  (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
	  issue #14744) Reported by: deepesh ........

2009-05-20 23:30 +0000 [r195839]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: If a variable had a blank value upon the
	  initial setting, then it would do nothing. Identified by Dmitry
	  Andrianov via private email, fixed by me.

2009-05-20 20:45 +0000 [r195763-195798]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Get rid of some duplicated code and correct
	  a connected line error. When receiving a 200 OK response to an
	  INVITE, it was possible to transmit two connected line updates
	  instead of a single one. Furthermore, the second did not have the
	  proper information present. Now the two have been combined into a
	  single update and the correct information is presented.

	* apps/app_dial.c: Plug a memory leak in app_dial. Since we may
	  have copied connected line info into the chanlist struct prior to
	  placing an outbound call, we need to be sure to free the
	  allocated data when we hang the call up.

2009-05-20 17:33 +0000 [r195636-195698]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 195688 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
	  lines Fix some code that wrongly assumed a pointer would always
	  be non-NULL when dealing with CDRs after a bridge. (closes issue
	  #15079) Reported by: barryf ........

	* /, apps/app_meetme.c: Merged revisions 195635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
	  lines Fix a bug where the MeetMe option 'D' did not actually
	  prompt for the pin. (closes issue #15050) Reported by: pmhaddad
	  ........

2009-05-19 20:59 +0000 [r195589]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Add basic support
	  for handling connected line-related UPDATE requests. SIP purists
	  may want to look the other way... When COLP/CONP support for SIP
	  was committed, there was a condition under which Asterisk may
	  transmit a SIP UPDATE in order to communicate the change in
	  connected line information. The issue here is that while we could
	  send a SIP UPDATE message, we were not prepared to receive such
	  an UPDATE and would always responde with a 501 when we received
	  an UPDATE. The situation was a bit rough. We really want to be
	  able to receive UPDATEs having to do with connected line changes,
	  but the amount of effort involved in properly supporting RFC 3311
	  was staggering. This commit represents a compromise. First, it
	  was decided that it is important to only send a SIP UPDATE to an
	  endpoint that is able to handle one. So, now we have added
	  parsing of the Allow header into SIP. We store the allowed
	  methods on SIP peers so that when we communicate with them, we
	  already will know what we can and cannot send to them. We will
	  parse the peer's allowed methods when he registers with us. If
	  the peer is not the type to register with us, but the qualify
	  option is enabled, then we will use the response to the OPTIONS
	  request we send the peer to determine the peer's allowed methods.
	  When the peer's registration expires, or when qualify deems the
	  peer to be unreachable, we clear the allowed methods from the
	  peer. For an actual call, we will copy the peer's allowed methods
	  to the sip_pvt representing the call leg. If we are communicating
	  with an endpoint which is not a peer, then we will just parse the
	  Allow header from the first message we receive during the call
	  and store the information in the sip_pvt. If, during
	  communication with a peer, we receive a 501 response, then we
	  will make sure to save the fact that we cannot use that method
	  when communicating with that peer. Now, with all that
	  infrastructure in place, the only actual place we use this
	  information currently is when attempting to send a connected line
	  change using an UPDATE request. If we cannot send the change
	  immediately using an UPDATE, we will set the SIP_NEEDREINVITE
	  flag so that we can send a REINVITE as soon as it is allowed. The
	  second part of the changes here is for Asterisk to accept UPDATE
	  requests that have connected line changes. Since we are not fully
	  supporting RFC 3311, Asterisk will NOT place the UPDATE method in
	  Allow headers it sends. Instead, if you are communicating with
	  what you know to be another Asterisk box, you may set the
	  rpid_update parameter in sip.conf so that we will send UPDATEs to
	  that Asterisk box. When we send a connected line update, we set a
	  custom header called "X-Asterisk-rpid-update." On the receiving
	  end, if Asterisk receives an UPDATE that does not have the
	  "X-Asterisk-rpid-update" header present, then Asterisk will
	  respond with a 501 since media-changing UPDATEs are not
	  supported. We should never get such UPDATEs, since as was stated
	  earlier, Asterisk does not put UPDATE in its Allow header. If the
	  custom header is present in the received UPDATE, though, then we
	  will check the incoming request for connected line updates and
	  queue the update on the channel where the change occurred.
	  ABE-1840 ABE-1822

2009-05-19 20:16 +0000 [r195521]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 195520 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19
	  May 2009) | 7 lines Ensure thread keys are initialized before
	  attempting to access them. (closes issue #14889) Reported by:
	  jaroth Patches: app_voicemail.c.patch uploaded by msirota
	  (license 758) Tested by: msirota, BlargMaN ........

2009-05-19 14:43 +0000 [r195449]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 195448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
	  lines Fix a bug where direct RTP setup would partially occur even
	  when disabled if the calling channel was answered. (issue #13545)
	  Reported by: davidw (issue #14244) Reported by: mbnwa ........

2009-05-18 20:52 +0000 [r195370]  Tilghman Lesher <tlesher@digium.com>

	* res/res_smdi.c, /, include/asterisk/monitor.h, apps/app_queue.c,
	  include/asterisk/smdi.h, res/res_monitor.c, apps/app_voicemail.c:
	  Recorded merge of revisions 195366 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
	  | 8 lines Add a similar dependency on SMDI for voicemail as
	  already exists for ADSI. (closes issue #14846) Reported by: pj
	  Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
	  (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
	  tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
	  uploaded by tilghman (license 14) ........

2009-05-18 20:49 +0000 [r195365-195369]  Eliel C. Sardanons <eliels@gmail.com>

	* main/manager.c: Fix the CLI command 'manager show command'
	  documentation and functionality. The CLI command 'manager show
	  command' supports passing multiple action names in the same line,
	  but it was not allowing that because of a incorrect check in the
	  argumentes counter. Also the documentation was updated to show
	  that this usage of the command is possible.

	* main/manager.c: Rollback commit 195367. The CLI command 'manager
	  show command' supports passing multiple AMI actions at a time.
	  The issue with this command was in another place.

	* main/manager.c: Avoid autocompleting passed the action name
	  argument in the CLI command. When running the autocomplete of the
	  CLI command 'manager show command <action>' it was autocompleting
	  everything else after the <action> argument, giving an error,
	  because this command doesn't support multiple AMI action names at
	  a time.

	* res/res_agi.c: Move AGI documentation from static to the XML
	  form. Move the AGI commands 'receive text', 'receive char' and
	  'record' static documentation to XML docs.

2009-05-18 19:17 +0000 [r195320]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Move the spawn of astcanary down, until after
	  the call to daemon(3). This avoids possible conflicts with the
	  internal implementation of daemon(3). (closes issue #15093)
	  Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt
	  uploaded by tilghman (license 14) Tested by: tzafrir

2009-05-18 18:58 +0000 [r195316]  Mark Michelson <mmichelson@digium.com>

	* apps/app_externalivr.c: Fix externalivr's setvariable command so
	  that it properly sets multiple variables. The command had a for
	  loop that was guaranteed to only execute once since the
	  continuation operation of the loop would set the input buffer
	  NULL. I rewrote the loop so that its operation was more obvious,
	  and it would set multiple variables correctly. I also reduced
	  stack space required for the function, constified the input
	  string, and modified the function so that it would not modify the
	  input string while I was at it. (closes issue #15114) Reported
	  by: chris-mac Patches: 15114.patch uploaded by mmichelson
	  (license 60) Tested by: chris-mac

2009-05-18 17:08 +0000 [r195279]  Sean Bright <sean@malleable.com>

	* cdr/cdr_custom.c: Remove some unused code.

2009-05-18 16:29 +0000 [r195266]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: The facilityenable parameter does not have
	  anything to do with pritimer parameters.

2009-05-18 15:55 +0000 [r195210]  Sean Bright <sean@malleable.com>

	* cdr/cdr_custom.c: Const-ify a string, fix a log message, and use
	  the correct signature for the load_module function.

2009-05-18 15:53 +0000 [r195207]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 195206 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
	  lines Fix a typo which caused loss of audio when using G729 in
	  some scenarios with a smoother present. (closes issue #15105)
	  Reported by: bamby Patches: process-vad-correctly.diff uploaded
	  by bamby (license 430) ........

2009-05-18 14:54 +0000 [r195165]  Sean Bright <sean@malleable.com>

	* configs/cdr_custom.conf.sample, CHANGES, cdr/cdr_custom.c: Allow
	  cdr_custom to write to multiple files instead of just one. Up to
	  now, cdr_custom would only accept a single filename/format from
	  cdr_custom.conf. This change allows you to specify multiple
	  filename & format directives.

2009-05-18 14:45 +0000 [r195162]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_dial.c, main/pbx.c, apps/app_macro.c: Warn about the use
	  of the application WaitExten() within a Macro(). Update
	  applications documentation to warn the user about the use of the
	  WaitExten() application within a Macro(). Recommend the use of
	  Read() instead. (closes issue #14444) Reported by: ewieling

2009-05-18 13:56 +0000 [r195089-195096]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c, /: Merged revisions 195095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
	  lines Fix a bug where the codecs of the called party leg were not
	  properly sent back to the caller call leg when reinvited. (closes
	  issue #13569) Reported by: bkw918 ........

	* channels/chan_sip.c: Fix a bug where specifying an empty
	  outboundproxy would cause packets to get sent to ourself. (closes
	  issue #15106) Reported by: timeshell

2009-05-18 13:30 +0000 [r195075]  Eliel C. Sardanons <eliels@gmail.com>

	* main/xml.c: Do not avoid loading the XML documentation if not
	  XInclude substitution is done.

2009-05-18 12:59 +0000 [r195021]  Russell Bryant <russell@digium.com>

	* /: Recorded merge of revisions 195020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
	  | 5 lines Don't try to unlock a bogus channel. (closes issue
	  #15144) Reported by: cristiandimache ........

2009-05-16 20:01 +0000 [r194945-194982]  Eliel C. Sardanons <eliels@gmail.com>

	* Makefile, main/xml.c, doc/appdocsxml.dtd: Allow to include
	  sections of other parts of the xml documentation. Avoid
	  duplicating xml documentation by allowing to include other parts
	  of the xml documentation using XInclude. Example: <xi:include
	  xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
	  (Insert this line to include the synopsis of the CHANNEL function
	  xml documentation). It is also possible to include documentation
	  from other files in the 'documentation/' directory using the
	  href="" attribute inside a xinclude element. (closes issue
	  #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling

	* main/pbx.c: Fix a missing unlock in case of error, and a missing
	  free(). Always free the allocated memory for a string field,
	  because we are always using it (not only when xmldocs are
	  enabled). Also if there is an error allocating memory for the
	  string field remember to unlock the list of registered
	  applications, before returning.

2009-05-15 22:44 +0000 [r194833-194874]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 194873 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15
	  May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ
	  to terminate invalid registrations. Instead it sent another
	  REGAUTH if the authentication challenge failed. This caused a
	  loop of REGREQ and REGAUTH frames. (Related to Security fix
	  AST-2009-001) (closes issue #14867) Reported by: aragon Tested
	  by: dvossel (closes issue #14717) Reported by: mobeck Patches:
	  regauth_loop_update_patch.diff uploaded by dvossel (license 671)
	  Tested by: dvossel ........

	* channels/iax2-parser.h, /, channels/iax2.h, channels/chan_iax2.c,
	  channels/iax2-parser.c: Merged revisions 194557,194685 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
	  | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
	  where people are reporting "Ghost" channels in their 'iax2 show
	  channels' output. The confusion is caused by channels being
	  listed as "(NONE)" with format "unknown". These are not channels
	  of coarse. They are usually just pending registration or poke
	  requests, but it is confusing output. To help make sense of this
	  I have added two columns to 'iax2 show channels'. One shows the
	  first message which started the transaction, and the second shows
	  the last message sent by either side of the call. This helps
	  diagnose why the entry exists and why it may not go away. (closes
	  issue #14207) Reported by: clive18 Review:
	  https://reviewboard.asterisk.org/r/246/ ........ r194685 |
	  dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
	  Update to previous IAX2 "Ghost" Channels patch. Fixed some
	  comments made on reviewboard for the previous patch. (issue
	  #14207) ........

2009-05-15 18:43 +0000 [r194714-194765]  Russell Bryant <russell@digium.com>

	* /, configs/logger.conf.sample: Merged revisions 194764 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
	  | 2 lines Fix some spelling fail. ........

	* codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Shuttle
	  some bits around to address some gain issues with G.722. (closes
	  AST-209)

	* codecs/Makefile, codecs/g722/Makefile (removed): Further simplify
	  codec_g722 build.

	* codecs/Makefile: Actually force running make for g722.

2009-05-15 13:43 +0000 [r194649]  Michiel van Baak <michiel@vanbaak.info>

	* CREDITS: add eliel

2009-05-15 13:23 +0000 [r194635]  Eliel C. Sardanons <eliels@gmail.com>

	* doc/appdocsxml.dtd, main/xmldoc.c: Allow to specify an enumlist
	  inside an enum. It was not possible to use an enumlist inside an
	  enum: <enumlist> <enum name="aa"> <enumlist> ... </enumlist>
	  </enum> </enumlist> Now we will be able to insert as many levels
	  as we want. (closes issue #15112) Reported by: lmadsen

2009-05-15 13:13 +0000 [r194520-194610]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/logger.h, tests/test_logger.c (added),
	  main/logger.c: Add ability for modules to dynamically register
	  logger levels This patch adds the ability for modules to
	  dynamically create logger levels for their own use; these are
	  named levels just like the built-in levels, and can be directed
	  to any destination that the logger can send any level to, by
	  including their names in logger.conf. Review:
	  https://reviewboard.asterisk.org/r/244/

	* /: Merged revisions 194509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
	  2009) | 1 line Update URL to Reviewboard ........

2009-05-14 22:20 +0000 [r194496]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 194484 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
	  2009) | 24 lines Fix a race condition where a reinvite could
	  trigger a 482 response. The loop detection/spiral detection code
	  in chan_sip used the owner channel's state as a criterion for
	  determining if the incoming INVITE is a looped request. The
	  problem with this is that the INVITE-handling code happens in a
	  different thread than the thread that marks the owner channel as
	  being up. As a result, if a reinvite were to come in very
	  quickly, say from another Asterisk on the same LAN, it was
	  possible for the reinvite to arrive before the owner channel had
	  been set to the up state. This patch corrects the problem by
	  using the invitestate of the sip_pvt instead, since that can be
	  guaranteed to be set correctly by the time the reinvite arrives.
	  Since there is a switch statement further in the INVITE-handling
	  code, the AST_STATE_RINGING state also checks the invitestate of
	  the sip_pvt in case we should actually be treating the channel as
	  if it were up already. (closes issue #12215) Reported by: jpyle
	  Patches: 12215_confirmed.patch uploaded by mmichelson (license
	  60) Tested by: lmadsen ........

2009-05-14 22:03 +0000 [r194479]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h,
	  channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample,
	  CHANGES, channels/misdn/isdn_lib.c, channels/misdn_config.c: Add
	  outgoing_colp misdn.conf port parameter. Select what to do with
	  outgoing COLP information on this port. 0 - Send out COLP
	  information unaltered. (default) 1 - Force COLP to restricted on
	  all outgoing COLP information. 2 - Do not send COLP information.
	  outgoing_colp=0 Also fixed sending the EctInform message so it
	  always has the required redirectionNumber parameter when the
	  status is active. JIRA ABE-1853

2009-05-14 21:24 +0000 [r194477]  Russell Bryant <russell@digium.com>

	* main/features.c: Fix a typo where an equality check should be an
	  assignment. (closes issue #15103) Reported by: lmsteffan Patches:
	  transfer_crash.patch uploaded by lmsteffan (license 779)

2009-05-14 17:05 +0000 [r194434]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Fix a bug where the 'T' option to Meetme did
	  not work. (closes issue #15031) Reported by: Stochastic (closes
	  issue #13801) Reported by: justdave

2009-05-14 16:22 +0000 [r194430]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: If the timing ended on a zero, then we would loop
	  forever. (closes issue #14983) Reported by: teox Patches:
	  20090513__issue14983.diff.txt uploaded by tilghman (license 14)
	  Tested by: teox

2009-05-13 15:02 +0000 [r194283]  Eliel C. Sardanons <eliels@gmail.com>

	* main/manager.c: Do not lock the 'sessions' container, lock the
	  allocated 'session'. There was a typo in the structure being
	  locked, and we were locking the 'sessions' container instead of
	  the 'session' structure thar we are modifying. Reported by
	  seanbright on #asterisk-dev, thanks!

2009-05-13 13:39 +0000 [r194209]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, /: Merged revisions 194208 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May
	  2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated
	  and with duration wrapping over. (closes issue #14815) Reported
	  by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license
	  88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes
	  issue #14460) Reported by: moliveras Tested by: moliveras
	  ........

2009-05-13 00:52 +0000 [r194101-194138]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 194137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
	  | 7 lines Fix logic for how to proceed with a single digit
	  extension. (closes issue #15091) Reported by: andrew Patches:
	  20090512__issue15091.diff.txt uploaded by tilghman (license 14)
	  Tested by: andrew ........

	* main/pbx.c, main/logger.c: Two fixes found while debugging with
	  ast_backtrace(): 1) If MALLOC_DEBUG is used when concurrently
	  using ast_backtrace, the free() used in that routine will trigger
	  an error, because the memory was allocated internally to libc,
	  where we could not intercept that call to wrap it. Therefore,
	  it's not memory we knew about, and the free is reported as an
	  error. 2) Now that channels are objects, the old hack of
	  initializing a channel to all zeroes no longer works, since we
	  may try to call something like ast_channel_lock() within a
	  function on that reference. In that case, it's reported as an
	  error, because the pointer isn't an object reference.

2009-05-12 22:49 +0000 [r194060]  Eliel C. Sardanons <eliels@gmail.com>

	* main/manager.c: Fix a crash when logging out from the AMI and
	  avoid astobj2 warning messages. When the user logout the session
	  was being destroyed twice and the file descriptor was being
	  closed twice. The sessions reference counter wasn't used in a
	  proper way. The 'mansession' structure was being treated as an
	  astobj2 and we were calling ao2_lock/ao2_unlock causing astobj2
	  report a warning message and not locking the structure. Also we
	  were using an ugly naming convention 'destroy_session',
	  'session_destroy', 'free_session', ... all this "duplicated" code
	  was merged. (closes issue #14974) Reported by: pj Patches:
	  manager.diff2 uploaded by eliel (license 64) Tested by: dhubbard,
	  eliel, mnicholson (closes issue #15088) Reported by: eliel
	  Review: http://reviewboard.asterisk.org/r/248/

2009-05-12 22:32 +0000 [r194057]  Matthew Nicholson <mnicholson@digium.com>

	* /, apps/app_queue.c: Merged revisions 194028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
	  2009) | 16 lines This change modifies app_queue to properly
	  generate CDR records in failure situations. This involves setting
	  a proper cdr disposition coresponding to the given failure
	  condition and ensuring the proper information is stored in the
	  cdr record. (closes issue #13691) Reported by: dferrer Tested by:
	  mnicholson (closes issue #13637) Reported by: atis Tested by:
	  atis ........

2009-05-12 20:40 +0000 [r193956]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 193955 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12
	  May 2009) | 6 lines Avoid initializing routines if the
	  authentication fails. Fixes a crash (RR) issue. (closes issue
	  #14508) Reported by: tiziano Patches:
	  20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
	  377) ........

2009-05-12 20:28 +0000 [r193954]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Update spiral support in trunk and 1.6.X to
	  match what is in 1.4. In 1.4, a SIP spiral is treated the same
	  way as a call forward. This works much better than what is
	  currently in trunk and 1.6.X. The code in trunk and 1.6.X did not
	  create a new call to the recipient of the spiral, instead trying
	  to continue the same call. In addition to just being plain wrong,
	  this also had the side effect of only being able to spiral calls
	  to other SIP channels. With this in place, as long as call
	  forwards are honored, SIP spirals will work properly. This means
	  that it will work for outbound calls made by the Queue, Dial, and
	  Page applications. For originated calls and spool calls, however,
	  the spiral will not work properly until a generic call forward
	  mechanism is introduced into Asterisk. (relates to issue #13630)

2009-05-12 17:29 +0000 [r193870]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Convert a THREADSTORAGE object into a
	  simple malloc'd object (as suggested by Russell on -dev)

2009-05-12 13:59 +0000 [r193832]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c, main/pbx.c, apps/app_meetme.c, apps/app_page.c,
	  main/devicestate.c, apps/app_queue.c, apps/app_transfer.c,
	  apps/app_playback.c, apps/app_controlplayback.c, main/term.c,
	  channels/chan_dahdi.c, channels/chan_misdn.c, funcs/func_curl.c,
	  apps/app_sendtext.c, apps/app_directed_pickup.c,
	  channels/console_gui.c, main/features.c, apps/app_confbridge.c,
	  apps/app_externalivr.c, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, apps/app_stack.c, res/res_odbc.c,
	  apps/app_voicemail.c: add 'const' qualifiers in various places
	  where they should have been

2009-05-11 23:04 +0000 [r193756-193757]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Found and fixed a memory leak

	* /: Recorded merge of revisions 193755 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
	  | 18 lines Move 300 bytes around on the stack, to make more room
	  for an extension buffer. This allows more concurrent extensions
	  to be copied for a single voicemail, without creating a
	  possibility of upsetting existing users, where a dialplan could
	  run out of stack space where it had run fine before.
	  Alternatively, we could have allocated off the heap, but that is
	  a larger change and would have increased the chance for
	  instability introduced by this change. This is really solved
	  starting in 1.6.0.11, as the use of an ast_str buffer allows an
	  unlimited number of extensions (up to available memory). We
	  additionally create a new warning message when the buffer length
	  is exceeded, permitting administrators to see an issue after the
	  fact, whereas previously the list was silently truncated. (closes
	  issue #14739) Reported by: p_lindheimer Patches:
	  20090417__bug14739.diff.txt uploaded by tilghman (license 14)
	  Tested by: p_lindheimer ........

2009-05-11 22:04 +0000 [r193718]  Russell Bryant <russell@digium.com>

	* res/res_timing_timerfd.c: Fix some timer state corruption. In
	  res_timer_timerfd, handle the case that set_rate gets called
	  while a timer is still in continuous mode. In this case, we want
	  to remember the configured rate, but not actually set it until
	  continuous mode has been disabled. Thanks to dvossel for finding
	  and helping to debug the problem. (closes issue #15080) Reported
	  by: dvossel Tested by: dvossel

2009-05-11 19:32 +0000 [r193678]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Don't nullify an ast_str pointer. (closes
	  issue #15061) Reported by: alecdavis

2009-05-11 19:11 +0000 [r193614]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 193613 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11
	  May 2009) | 12 lines Sent wrong message to clear a call we
	  started if the other end has not responed yet. In the state
	  MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
	  yet), it is not allowed to clear the call with RELEASE_COMPLETE.
	  It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
	  allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
	  5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
	  JIRA ABE-1862 ........

2009-05-11 18:01 +0000 [r193545]  Leif Madsen <lmadsen@digium.com>

	* /, funcs/func_channel.c: Recorded merge of revisions 193544 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009)
	  | 7 lines Document CHANNEL(transfercapability) in CLI
	  documentation. (issue #15073) Reported by: pkempgen Patches:
	  20090511__issue15073.diff.txt uploaded by tilghman (license 14)
	  ........

2009-05-10 17:07 +0000 [r193502]  Joshua Colp <jcolp@digium.com>

	* main/bridging.c: Fix a bug where receiving a control frame of
	  subclass -1 would cause certain channels to get hung up.

2009-05-09 11:33 +0000 [r193459-193461]  Russell Bryant <russell@digium.com>

	* include/asterisk/event.h: Minor documentation update for
	  ast_event_queue().

	* main/channel.c: Declare private data as static.

2009-05-08 20:32 +0000 [r193387]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: TCP not matching valid peer. find_peer()
	  does not find a valid peer when using pvt->recv as the
	  sockaddr_in argument. Because of the way TCP works, the port
	  number in pvt->recv is not what we're looking for at all. There
	  is currently only one place that find_peer searches for a peer
	  using the sockaddr_in argument. If the peer is not found after
	  using pvt->recv (works for UDP since the port number will be
	  correct), a temp sockaddr_in struct is made using the Contact
	  header in the sip_request. This has the correct port number in
	  it. Review: http://reviewboard.digium.com/r/236/

2009-05-08 19:50 +0000 [r193349]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Reset the members' call counts when resetting
	  queue statistics. This helps to prevent odd scenarios where a
	  queue will claim to have taken 0 calls, but the members appear to
	  have taken a non-zero amount. (closes issue #15068) Reported by:
	  sum Patches: patchreset.patch uploaded by sum (license 766)
	  Tested by: sum

2009-05-08 15:18 +0000 [r193274]  Sean Bright <sean@malleable.com>

	* funcs/func_devstate.c: Fix the spelling of UNAVAILABLE in
	  func_devstate CLI completion.

2009-05-08 14:52 +0000 [r193263]  David Vossel <dvossel@digium.com>

	* /, channels/misdn_config.c: Merged revisions 193262 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08
	  May 2009) | 9 lines "misdn show config" segfaults asterisk, if no
	  MSN lists (closes issue #14976) Reported by: alecdavis Patches:
	  misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
	  by: alecdavis, FabienToune ........

2009-05-08 14:06 +0000 [r193194]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/logger.c, configs/logger.conf.sample: Merged revisions
	  193193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
	  2009) | 7 lines Make absolute paths for logger channels work
	  properly (Note: This is not a new feature, it was previously
	  undocumented and broken.) The Asterisk logger has a feature to
	  support absolute pathnames for logger channels, but the code
	  implementing the feature was broken. This has been fixed, and the
	  absolute path feature is now documented in the sample
	  logger.conf. ........

2009-05-07 23:42 +0000 [r193120]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 193119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
	  | 19 lines Fix Background within a Macro for FreePBX. If the
	  single digit DTMF is an extension in the specified context, then
	  go there and signal no DTMF. Otherwise, we should exit with that
	  DTMF. If we're in Macro, we'll exit and seek that DTMF as the
	  beginning of an extension in the Macro's calling context. If
	  we're not in Macro, then we'll simply seek that extension in the
	  calling context. Previously, someone complained about the
	  behavior as it related to the interior of a Gosub routine, and
	  the fix (#14011) inadvertently broke FreePBX (#14940). This
	  change should fix both of these situations, but with the possible
	  incompatibility that if a single digit extension does not exist
	  (but a longer extension COULD have matched), it would have
	  previously gone immediately to the "i" extension, but will now
	  need to wait for a timeout. (closes issue #14940) Reported by:
	  p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
	  tilghman (license 14) Tested by: p_lindheimer ........

2009-05-07 22:24 +0000 [r193077]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 193050 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07
	  May 2009) | 5 lines Give a more helpful message when an incoming
	  call's dialed extension does not match. Added the dialed
	  extension and context to the chan_misdn messages warning that the
	  dialed number cannot be matched in the dialplan. ........

2009-05-07 17:51 +0000 [r192933-193006]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: Second result should not contain data from the
	  first result. (closes issue #15039) Reported by: jims Patches:
	  20090506__issue15039.diff.txt uploaded by tilghman (license 14)
	  Tested by: jims

	* channels/chan_unistim.c: Send DTMF frame before playing back
	  audio. (closes issue #14858) Reported by: barryf Patches:
	  20090507__bug14858.diff.txt uploaded by tilghman (license 14)

	* /, channels/chan_sip.c: Merged revisions 192932 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
	  | 10 lines Eliminate repetition of fullcontact during
	  reconstruction. If the fullcontact field appears in both the
	  sippeers and the sipregs table, then during reconstruction of the
	  field, it will otherwise be doubled. (closes issue #14754)
	  Reported by: Alexei Gradinari Patches:
	  20090506__bug14754.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen ........

2009-05-06 22:17 +0000 [r192853-192861]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 192858 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
	  | 10 lines Make ParkedCall application stop execution of the
	  dialplan after hang up Just changed park_exec to always return
	  non-zero. I really wasn't entirely sure at first if this was a
	  bug. Decided it was since it would be surprising when not using
	  ParkedCall in the dialplan to hang up and have dialplan execution
	  continue. (closes issue #14555) Reported by: francesco_r ........

	* main/pbx.c: If no extension was found in the pattern tree, don't
	  crash.

2009-05-06 17:38 +0000 [r192808]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Fix a bug where a timer would be created
	  but not acknowledged. This scenario crept up if chan_iax2 was
	  loaded with no configuration file present. It would create a
	  timer and tell it to go at an interval but the thread that
	  normally acknowledges it would not be created because no
	  configuration file was present. The timer will now be closed if
	  no configuration file is present. (closes issue #15014) Reported
	  by: madkins

2009-05-06 16:28 +0000 [r192772]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c, doc/lang/urdu.ods (added): Add numbers in Urdu, the
	  national language of Pakistan (closes issue #15034) Reported by:
	  nasirq Patches: ast_say_number_full_ur-patch.c uploaded by nasirq
	  (license 772) urdu.ods uploaded by nasirq (license 772)

2009-05-06 16:09 +0000 [r192634-192736]  Joshua Colp <jcolp@digium.com>

	* res/res_clialiases.c: Make the code that prevents an infinite
	  loop from happening into a case insensitive check. (thanks eliel)

	* res/res_clialiases.c: Fix an infinite loop with tab completion of
	  CLI aliases that reference themselves. (closes issue #15020)
	  Reported by: junky

	* /, channels/chan_sip.c: Merged revisions 192633 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
	  lines Update some old logic to stop both begin and end DTMF
	  frames from reaching the core if rfc2833 is not enabled. (closes
	  issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
	  by dimas (license 88) ........

2009-05-05 20:54 +0000 [r192590]  Richard Mudgett <rmudgett@digium.com>

	* apps/app_dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
	  main/features.c, apps/app_queue.c: Fixed crashes from issue8824
	  review board channel locking changes. The local struct
	  ast_party_connected_line connected_caller variable was
	  uninitialized when the copy function was called.

2009-05-05 19:57 +0000 [r192525]  Sean Bright <sean@malleable.com>

	* /, static-http/astman.js: Merged revisions 192524 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue,
	  05 May 2009) | 11 lines Fix Javascript error when using astman.js
	  in Internet Explorer. Internet Explorer (tested with 7.0) does
	  not like trailing commas on constructs like object initializers,
	  so get rid of them to avoid some errors. (closes issue #15026)
	  Reported by: rajnishgiri Patches: bug15026.patch uploaded by
	  seanbright (license 71) Tested by: seanbright ........

2009-05-05 18:23 +0000 [r192430-192462]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 192454 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
	  lines Fix an incorrect assumption that certain values on the
	  channel will always exist when they may not. The CDR code
	  involved with bridges wrongly assumed that the currently
	  executing application and data values will always exist. It is
	  possible for this to be false when call forwarding is involved.
	  (closes issue #14984) Reported by: gincantalupo ........

	* /, apps/app_followme.c: Merged revisions 192429 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
	  lines Fix a bug where the followme application would continue
	  trying numbers after the caller hung up. (closes issue #13624)
	  Reported by: sgenyuk ........

2009-05-05 17:33 +0000 [r192427]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Revert CPC patch for now, until I decide
	  whether or not it all should be merged into libss7/1.0 (It's
	  still in the bug13495 branch and in libss7/trunk)

2009-05-05 14:22 +0000 [r192387]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug with setting t38pt_udptl at the
	  user or peer level. If an incoming call authenticated as a user
	  or peer and t38pt_udptl was not set to yes in general then no
	  UDPTL session would be present and any T38 related things would
	  fail. This commit changes it so that if after authenticating T38
	  is enabled but no UDPTL session is present one will be created.
	  (issue AST-215)

2009-05-05 14:17 +0000 [r192279-192362]  Kevin P. Fleming <kpfleming@digium.com>

	* main/utils.c, include/asterisk/stringfields.h: Add a more
	  efficient way of allocating structures that use stringfields This
	  commit adds an API call that can be used to allocate a structure
	  along with this stringfield storage in a single allocation.

	* main/utils.c, main/astobj2.c, include/asterisk/stringfields.h:
	  Correct some flaws in the memory accounting code for stringfields
	  and ao2 objects Under some conditions, the memory allocation for
	  stringfields and ao2 objects would not have supplied valid
	  file/function names for MALLOC_DEBUG tracking, so this commit
	  corrects that.

	* main/channel.c, include/asterisk/astobj2.h,
	  include/asterisk/datastore.h, include/asterisk/channel.h,
	  main/astobj2.c, main/datastore.c: Properly account for memory
	  allocated for channels and datastores As in previous commits,
	  when channels are allocated (with ast_channel_alloc) or
	  datastores are allocated (with ast_datastore_alloc) properly
	  account for the memory being owned by the caller, instead of the
	  allocator function itself.

	* main/utils.c, include/asterisk/stringfields.h: Ensure that string
	  pools allocated to hold stringfields are properly accounted in
	  MALLOC_DEBUG mode This commit modifies the stringfield pool
	  allocator to remember the 'owner' of the stringfield manager the
	  pool is being allocated for, and ensures that pools allocated in
	  the future when fields are populated are owned by that
	  file/function.

2009-05-04 22:44 +0000 [r192214]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 192213 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04
	  May 2009) | 11 lines global mohinterpret setting is ignored
	  mohinterpret and mohsuggest global variables were not copied over
	  during build_users and build_peers. (closes issue #14728)
	  Reported by: dimas Patches: v1-14728.patch uploaded by dimas
	  (license 88) Tested by: dimas, dvossel ........

2009-05-04 19:29 +0000 [r192132-192171]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/autoconfig.h.in, res/res_agi.c: Restore
	  'asyncagi break' command to 1.6.1 and higher. (closes issue
	  #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt
	  uploaded by tilghman (license 14)
	  20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
	  14) Tested by: nikkk

	* autoconf/ast_ext_tool_check.m4: Pass libraries in LIBS, not
	  LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches:
	  asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
	  Chainsaw (license 723)

2009-05-04 17:42 +0000 [r192096]  Leif Madsen <lmadsen@digium.com>

	* apps/app_forkcdr.c: Commit documentation changes related to issue
	  #14801. (issue #14801)

2009-05-04 16:24 +0000 [r192059]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/astobj2.h, main/astobj2.c: Ensure that astobj2
	  memory allocations are properly accounted for when MALLOC_DEBUG
	  is used This commit ensures that all astobj2 allocated objects
	  are properly accounted for in MALLOC_DEBUG mode by passing down
	  the file/function/line information from the module/function that
	  actually called the astobj2 allocation function.

2009-05-04 15:35 +0000 [r192032]  Eliel C. Sardanons <eliels@gmail.com>

	* main/xml.c: Do not re-define _POSIX_C_SOURCE if it was already
	  defined.

2009-05-04 12:52 +0000 [r191919-191997]  Kevin P. Fleming <kpfleming@digium.com>

	* tests/test_skel.c, tests/test_sched.c: Minor changes in test
	  modules Correct command description in test_sched.c and include
	  asterisk/cli.h in test_skel.c, since it's highly unlikely that a
	  test module will *not* want to provide CLI commands to execute
	  the tests

	* configs/modules.conf.sample: Ensure that by default only one
	  console channel driver is loaded This configuration file was
	  changed to ensure that only one console channel driver (chan_oss)
	  is loaded by default, but the change would only work if
	  chan_console was not built. Now it will work as expected; if
	  chan_alsa or chan_console are built and installed, they will not
	  be loaded unless explicity requested.

	* include/asterisk/event.h, include/asterisk/event_defs.h,
	  main/event.c: Add 'bitflags'-style information elements to event
	  framework This patch add a new payload type for information
	  elements, a set of bit flags. The payload is transported as a
	  32-bit unsigned integer but when matching is performed between
	  events and subscribers, the matching is done by using a bitwise
	  AND instead of numeric value comparison. Review:
	  http://reviewboard.asterisk.org/r/242/

2009-05-03 14:05 +0000 [r191848-191884]  Russell Bryant <russell@digium.com>

	* Makefile: Remove unnecessary compiler flag

	* main/event.c: Do a bit of code cleanup. - convert handling of IE
	  PLTYPEs to switch statements - add braces to various small blocks
	  - remove a bit of trailing whitespace - remove a couple of
	  unnecessary ast_strdupa() uses

2009-05-02 19:02 +0000 [r191775-191785]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/logger.h, main/manager.c, pbx/pbx_spool.c,
	  main/logger.c, apps/app_sms.c, CHANGES, apps/app_verbose.c,
	  configs/logger.conf.sample: Remove rarely-used
	  event_log/LOG_EVENT support In discussions today at the Europe
	  Asterisk Developer Meet-Up, we determined that the event_log was
	  used in only 9 places in the entire tree, and really was not
	  needed at all. The users have been converted to use LOG_NOTICE,
	  or the messages have been removed since other messages were
	  already in place that provided the same information.

	* main/logger.c: Fix an error in queue_log file rotation
	  optimization code This code was copy-and-pasted without properly
	  changing references to event_rotate into queue_rotate, so under
	  some conditions the log rotation would rotate queue_log even
	  though it was not necessary.

2009-05-02 16:43 +0000 [r191700-191739]  Sean Bright <sean@malleable.com>

	* channels/chan_dahdi.c: Conditional include ioctl's to change EC
	  policy based on DAHDI caps. This feels like a sane change
	  (wouldn't compile without this addition), but I'm not intimately
	  familiar with this code.

	* main/asterisk.c: Update copyright year to 2009

2009-05-01 20:01 +0000 [r191494-191560]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 191559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
	  | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
	  (closes issue #14993) Reported by: BigJimmy Patches: causepatch
	  uploaded by BigJimmy (license 371) ........

	* channels/chan_iax2.c: Set debug message back to DEBUG level.
	  (closes issue #15007) Reported by: hulber

2009-05-01 18:09 +0000 [r191489]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 191488 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
	  | 9 lines Fix DTMF not being sent to other side after a partial
	  feature match This fixes a regression from commit 176701. The
	  issue was that ast_generic_bridge never exited after the feature
	  digit timeout had elapsed, which prevented the queued DTMF from
	  being sent to the other side. This issue was reported to me
	  directly. ........

2009-05-01 14:58 +0000 [r191419]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c: Drop my IRC nickname.

2009-05-01 09:50 +0000 [r191418]  TransNexus OSP Development <support@transnexus.com>

	* configs/osp.conf.sample, apps/app_osplookup.c: Made security
	  features optional.

2009-04-30 21:42 +0000 [r191411]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
	  buffer and echo canceller control to CHANNEL() dialplan function
	  for DAHDI channels Adds ability for CHANNEL() dialplan function,
	  when used on DAHDI channels, to temporarily change the number of
	  buffers and/or the buffer policy, and also to enable, disable, or
	  switch the echo canceller between FAX/data and voice modes.

2009-04-30 17:40 +0000 [r191367]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/asterisk.c: Detect eaccess (or euidaccess) before using it.
	  Reported by Andrew Lindh via the -dev list.

2009-04-30 09:11 +0000 [r191300-191332]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Added routing number support.

	* apps/app_osplookup.c: Fixed not report source network ID and not
	  export destination network ID issues.

2009-04-30 06:47 +0000 [r191219-191283]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Change working directory to / under certain
	  conditions. If backgrounding and no core will be produced, then
	  changing the directory won't break anything; likewise, if the CWD
	  isn't accessible by the current user, then a core wasn't possible
	  anyway. (closes issue #14831) Reported by: chris-mac Patches:
	  20090428__bug14831.diff.txt uploaded by tilghman (license 14)
	  20090430__bug14831.diff.txt uploaded by tilghman (license 14)
	  Tested by: chris-mac

	* /: Recorded merge of revisions 191220 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r191220 | tilghman | 2009-04-29 18:10:54 -0500 (Wed, 29 Apr 2009)
	  | 2 lines Allow H.323 to compile with FDLEAK checking enabled.
	  ........

	* channels/h323/ast_h323.cxx, channels/chan_h323.c: Make H.323
	  compile with FDLEAK detection code enabled

2009-04-29 22:56 +0000 [r191213]  Jeff Peeler <jpeeler@digium.com>

	* res/res_phoneprov.c: fix typos

2009-04-29 22:23 +0000 [r191211]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Part of the merge did not happen correctly, which
	  resulted in a compile error

2009-04-29 21:13 +0000 [r191177]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, configs/sip.conf.sample,
	  include/asterisk/tcptls.h, CHANGES: SIP option to specify
	  outbound TLS/SSL client protocol. chan_sip allows for outbound
	  TLS connections, but does not allow the user to specify what
	  protocol to use (default was SSLv2, and still is if this new
	  option is not specified). This patch lets the user pick the
	  SSL/TLS client method for outbound connections in sip. (closes
	  issue #14770) Reported by: TheOldSaint (closes issue #14768)
	  Reported by: TheOldSaint Review:
	  http://reviewboard.digium.com/r/240/

2009-04-29 21:07 +0000 [r191175]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, CHANGES: Outgoing PTP redirected calls did
	  not wait for the COLR from the redirected-to party. For outgoing
	  PTP redirected calls, you now need to use the inhibit(i) option
	  on all of the REDIRECTING statements before dialing the
	  redirected-to party. You still have to set the
	  REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The
	  PTP call will update the redirecting-to presentation when it
	  becomes available and queue the redirecting update to the calling
	  channel.

2009-04-29 18:53 +0000 [r191140]  Tilghman Lesher <tlesher@digium.com>

	* tests/test_substitution.c (added), funcs/func_base64.c,
	  funcs/func_rand.c, funcs/func_speex.c, funcs/func_md5.c,
	  funcs/func_module.c, include/asterisk/autoconfig.h.in,
	  funcs/func_env.c, funcs/func_strings.c, res/res_phoneprov.c,
	  funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
	  funcs/func_logic.c, apps/app_exec.c, funcs/func_groupcount.c,
	  configure, funcs/func_aes.c, main/ast_expr2f.c, res/res_agi.c,
	  apps/app_minivm.c, include/asterisk/ast_expr.h, cdr/cdr_custom.c,
	  main/strings.c, main/pbx.c, funcs/func_dialplan.c,
	  funcs/func_db.c, funcs/func_timeout.c, funcs/func_lock.c,
	  funcs/func_cut.c, funcs/func_extstate.c, res/res_config_curl.c,
	  funcs/func_curl.c, funcs/func_blacklist.c, apps/app_macro.c,
	  include/asterisk/pbx.h, funcs/func_callerid.c,
	  apps/app_voicemail.c: Merge str_substitution branch. This branch
	  adds additional methods to dialplan functions, whereby the result
	  buffers are now dynamic buffers, which can be expanded to the
	  size of any result. No longer are variable substitutions limited
	  to 4095 bytes of data. In addition, the common case of needing
	  buffers much smaller than that will enable substitution to only
	  take up the amount of memory actually needed. The existing
	  variable substitution routines are still available, but users of
	  those API calls should transition to using the dynamic-buffer
	  APIs. Reviewboard: http://reviewboard.digium.com/r/174/

2009-04-29 18:32 +0000 [r191136]  David Brooks <dbrooks@digium.com>

	* pbx/pbx_config.c: Removing crufty code that is no longer
	  necessary. Code cleanup.

2009-04-29 14:39 +0000 [r191028]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
	  configs/manager.conf.sample, include/asterisk/tcptls.h, CHANGES,
	  configs/http.conf.sample: Consistent SSL/TLS options across conf
	  files ast_tls_read_conf() is a new api call for handling SSL/TLS
	  options across all conf files. Before this change, SSL/TLS
	  options were not consistent. http.conf and manager.conf required
	  the 'ssl' prefix while sip.conf used options with the 'tls'
	  prefix. While the options had different names in different conf
	  files, they all did the exact same thing. Now, instead of mixing
	  'ssl' or 'tls' prefixes to do the same thing depending on what
	  conf file you're in, all SSL/TLS options use the 'tls' prefix.
	  For example. 'sslenable' in http.conf and manager.conf is now
	  'tlsenable' which matches what already existed in sip.conf. Since
	  this has the potential to break backwards compatibility, previous
	  options containing the 'ssl' prefix still work, but they are no
	  longer documented in the sample.conf files. The change is noted
	  in the CHANGES file though. Review:
	  http://reviewboard.digium.com/r/237/

2009-04-29 08:58 +0000 [r190989-190993]  Russell Bryant <russell@digium.com>

	* main/indications.c: Log an error message if indications.conf is
	  not found. (closes issue #14990) Reported by: tzafrir Patches:
	  indications_err.diff uploaded by tzafrir (license 46)

	* apps/app_queue.c: Fix app_queue XML documentation. I think it
	  would behoove us to force "make validate-docs" to be run after
	  the XML documentation has been generated if dev-mode is enabled.
	  (closes issue #14989) Reported by: tzafrir Patches:
	  app_queue_xml.diff uploaded by tzafrir (license 46)

	* main/rtp_engine.c, include/asterisk/channel.h: Resolve Solaris
	  build issues and add some API documentation. (issue #14981)
	  Reported by: snuffy

2009-04-28 22:07 +0000 [r190946-190947]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Add support setting CPC from channel
	  variable

	* channels/chan_dahdi.c: Make sure that we do not clear the down
	  flag on the BRI during PTMP link transients

2009-04-28 17:31 +0000 [r190904]  Tilghman Lesher <tlesher@digium.com>

	* doc/tex/cdrdriver.tex: UniqueID column has a maximum size of 150

2009-04-28 14:15 +0000 [r190861-190865]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: Build XML documention from *only* the source files that
	  have docs in them Change the build process so that
	  doc/core-en_US.xml is dependent solely on the source files that
	  have documentation in them, not on all source files.

	* Makefile.rules: Remove Makefile rules for bison and flex sources
	  We never, ever want these files to processed automatically,
	  because we store the output files in Subversion and users should
	  never need to rebuild them.

2009-04-28 09:10 +0000 [r190830]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Updated for OSP Toolkit 3.5.

2009-04-27 21:22 +0000 [r190735-190797]  Richard Mudgett <rmudgett@digium.com>

	* main/channel.c: Fix a small memory leak on error in
	  ast_channel_alloc().

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c, CHANGES,
	  channels/misdn/isdn_lib.c, funcs/func_redirecting.c: Make PTP
	  DivertingLegInformation3 message behavior closer to the
	  specifications. * Wait for a DivertingLegInformation3 message
	  after receiving a DivertingLegInformation1 message to complete
	  the redirecting-to information before queuing a redirecting
	  update to the other channel. * A DivertingLegInformation2 message
	  should be responded to with a DivertingLegInformation3 when the
	  COLR is determined. If the call could or does experience another
	  redirection, you should manually determine the COLR to send to
	  the switch by setting REDIRECTING(to-pres) to the COLR and
	  setting REDIRECTING(to-num) = ${EXTEN}. * A
	  DivertingLegInformation2 message must have an original called
	  number if the redirection count is greater than one. Since
	  Asterisk does not keep track of this information, we can only
	  indicate that the number is not available due to interworking.

2009-04-27 19:34 +0000 [r190726]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Don't warn on pipe in the System call. (closes issue
	  #14979) Reported by: pj

2009-04-27 19:30 +0000 [r190725]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in: Merged revisions
	  190721 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
	  2009) | 7 lines Fix 'inconsistent line endings' when autoconf
	  2.63 is used Attempt to make configure script regeneration 'safe'
	  using autoconf 2.63, which embeds a bare CR into the script, thus
	  making Subversion complain about inconsistent line endings This
	  commit changes the MIME type of the configure script to be
	  'binary' thus making Subversion no longer inspect line endings,
	  and as a bonus 'svn diff' will no longer try to generate diff
	  output for it, which is not generally useful anyway. ........

2009-04-27 19:08 +0000 [r190663]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c, /: Merged revisions 190661-190662 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27
	  Apr 2009) | 9 lines Resolve a crash in res_smdi when used with
	  chan_dahdi. When chan_dahdi goes to get an SMDI message, it
	  provides no search criteria. It just grabs the next message that
	  arrives. This code was written with the SMDI dialplan functions
	  in mind, since that is now the preferred method of using SMDI.
	  However, this broke support of it being used from chan_dahdi.
	  (closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59
	  -0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661.
	  ........

2009-04-27 16:37 +0000 [r190622-190626]  Mark Michelson <mmichelson@digium.com>

	* doc/tex/channelvariables.tex, apps/app_queue.c: Allow for a
	  position to be specified when entering a queue. This would allow
	  for one to add a caller to a specific place in the queue instead
	  of just placing the caller in the back every time. To help
	  facilitate some interesting manipulations, a new channel variable
	  called QUEUEPOSITION has been added. When a caller is removed
	  from a queue, his position in that queue is stored in the
	  QUEUEPOSITION variable. One such strategy an administrator can
	  employ is to allow for the removal of a caller from one queue
	  followed by the insertion of the same caller into a separate
	  queue in the same position. Review:
	  http://reviewboard.digium.com/r/189

	* apps/app_queue.c: Update warning message to not have pipes and
	  contain all options.

2009-04-27 15:18 +0000 [r190586]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Fix a bug where we tried to send events out when
	  no sessions container was present. This commit stops a warning
	  message (user_data is NULL) from getting output when manager
	  events get sent before manager is initialized. This happens
	  because manager is initialized *after* modules are loaded and the
	  act of loading modules triggers manager events. (issue #14974)
	  Reported by: pj

2009-04-27 14:46 +0000 [r190577]  Mark Michelson <mmichelson@digium.com>

	* configs/sip.conf.sample: Remove nonexistent option from
	  sip.conf.sample. The option to choose which connected line header
	  to use is not 'rpid_header' but 'sendrpid'

2009-04-24 21:22 +0000 [r190545]  David Vossel <dvossel@digium.com>

	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
	  configs/manager.conf.sample, configs/sip.conf.sample,
	  include/asterisk/tcptls.h, CHANGES, configs/http.conf.sample:
	  TLS/SSL private key option Adds option to specify a private key
	  .pem file when configuring TLS or SSL in AMI, HTTP, and SIP.
	  Before this, the certificate file was used for both the public
	  and private key. It is possible for this file to hold both, but
	  most configurations allow for a separate private key file to be
	  specified. Clarified in .conf files how these options are to be
	  used. The current conf files do not explain how the private key
	  is handled at all, so without knowledge of Asterisk's TLS
	  implementation, it would be hard to know for sure what was going
	  on or how to set it up. Review:
	  http://reviewboard.digium.com/r/234/

2009-04-24 17:59 +0000 [r190516-190517]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, funcs/func_connectedline.c: There is no
	  need to use the struct ast_party_connected_line.source update
	  values. The messages sent by a technology when a connected line
	  update is received are best determined by the current call state
	  of the channel. The struct ast_party_connected_line.source value
	  is really only useful as a possible tracing aid.

	* include/asterisk/channel.h: Update comment.

2009-04-24 15:26 +0000 [r190423-190484]  Russell Bryant <russell@digium.com>

	* include/asterisk/channel.h: Add \since tag for new API calls.

	* channels/chan_misdn.c: Fix a build error.

	* channels/chan_unistim.c, channels/chan_local.c,
	  apps/app_dahdiscan.c (removed), main/devicestate.c,
	  main/autochan.c (added), funcs/func_logic.c,
	  channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
	  main/channel.c, build_tools/cflags.xml, channels/chan_dahdi.c,
	  main/manager.c, funcs/func_odbc.c, apps/app_minivm.c,
	  main/features.c, res/res_agi.c, main/logger.c,
	  channels/chan_mgcp.c, res/res_clioriginate.c, main/pbx.c,
	  channels/chan_sip.c, include/asterisk/autochan.h (added),
	  channels/chan_bridge.c, main/Makefile, apps/app_softhangup.c,
	  channels/chan_agent.c, UPGRADE.txt, include/asterisk/channel.h,
	  CHANGES, funcs/func_global.c, res/res_monitor.c,
	  apps/app_channelredirect.c, channels/chan_misdn.c,
	  apps/app_directed_pickup.c, funcs/func_channel.c,
	  res/snmp/agent.c, include/asterisk/lock.h, apps/app_senddtmf.c,
	  apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
	  Convert the ast_channel data structure over to the astobj2
	  framework. There is a lot that could be said about this, but the
	  patch is a big improvement for performance, stability, code
	  maintainability, and ease of future code development. The channel
	  list is no longer an unsorted linked list. The main container for
	  channels is an astobj2 hash table. All of the code related to
	  searching for channels or iterating active channels has been
	  rewritten. Let n be the number of active channels. Iterating the
	  channel list has gone from O(n^2) to O(n). Searching for a
	  channel by name went from O(n) to O(1). Searching for a channel
	  by extension is still O(n), but uses a new method for doing so,
	  which is more efficient. The ast_channel object is now a
	  reference counted object. The benefits here are plentiful. Some
	  benefits directly related to issues in the previous code include:
	  1) When threads other than the channel thread owning a channel
	  wanted access to a channel, it had to hold the lock on it to
	  ensure that it didn't go away. This is no longer a requirement.
	  Holding a reference is sufficient. 2) There are places that now
	  require less dealing with channel locks. 3) There are places
	  where channel locks are held for much shorter periods of time. 4)
	  There are places where dealing with more than one channel at a
	  time becomes _MUCH_ easier. ChanSpy is a great example of this.
	  Writing code in the future that deals with multiple channels will
	  be much easier. Some additional information regarding channel
	  locking and reference count handling can be found in channel.h,
	  where a new section has been added that discusses some of the
	  rules associated with it. Mark Michelson also assisted with the
	  development of this patch. He did the conversion of ChanSpy and
	  introduced a new API, ast_autochan, which makes it much easier to
	  deal with holding on to a channel pointer for an extended period
	  of time and having it get automatically updated if the channel
	  gets masqueraded. Mark was also a huge help in the code review
	  process. Thanks to David Vossel for his assistance with this
	  branch, as well. David did the conversion of the DAHDIScan
	  application by making it become a wrapper for ChanSpy internally.
	  The changes come from the
	  svn/asterisk/team/russell/ast_channel_ao2 branch. Review:
	  http://reviewboard.digium.com/r/203/

2009-04-24 13:49 +0000 [r190421]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix nat setting on RTP instances. (closes
	  issue #14827) Reported by: pj

2009-04-23 21:13 +0000 [r190357]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 190356 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009)
	  | 2 lines Remove a bogus ast_channel_unlock(). ........

2009-04-23 20:42 +0000 [r190349-190352]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Labels are sometimes (most of the time?) NULL for
	  extensions. (closes issue #14895) Reported by: chris-mac Patches:
	  20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen

	* include/asterisk/http.h, include/asterisk/utils.h,
	  main/manager.c, res/res_phoneprov.c, main/http.c, main/utils.c,
	  res/res_http_post.c, main/astobj2.c: Support HTTP digest
	  authentication for the http manager interface. (closes issue
	  #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff
	  uploaded by ys (license 281) SVN branch
	  http://svn.digium.com/svn/asterisk/team/group/manager_http_auth
	  Tested by: ys, twilson, tilghman Review:
	  http://reviewboard.digium.com/r/223/ Reviewed by:
	  tilghman,russellb,mmichelson

2009-04-23 19:15 +0000 [r190287]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 190286 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr
	  2009) | 6 lines Fix a bug in chan_local glare hangup detection.
	  If both sides of a Local channel were hung up at around the same
	  time it was possible for one thread to destroy the local private
	  structure and have the other thread immediately try to remove the
	  already freed structure from the local channel list. ........

2009-04-23 17:45 +0000 [r190250]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix reversed behavior of leavewhenempty option
	  in queues.conf. (closes issue #14650) Reported by: alecdavis
	  Patches: 14650.patch uploaded by mmichelson (license 60) Tested
	  by: mmichelson, lmadsen

2009-04-23 16:55 +0000 [r190217]  Joshua Colp <jcolp@digium.com>

	* apps/app_directed_pickup.c: Fix a double free issue with the
	  Pickup dialplan application. As part of the pickup process the
	  connected line information is updated. Part of this process does
	  a shallow copy of the target channel's connected line information
	  to a local structure. Once complete the structure contents are
	  freed. As a result any information in the target channel's
	  connected line information structure is no longer valid. This
	  change will now set the contents back to a clean state so that
	  the freeing of the target channel's connected line information
	  structure when the channel is destroyed will no longer try to
	  double free things. (closes issue #14839) Reported by: lmsteffan

2009-04-23 00:44 +0000 [r190154]  Terry Wilson <twilson@digium.com>

	* funcs/func_strings.c: Fix example that could fail in certain
	  circumstances

2009-04-22 21:38 +0000 [r190093]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Merged revisions 190092 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22
	  Apr 2009) | 7 lines Detect availability of
	  pthread_rwlock_timedwrlock() before using it. (closes issue
	  #14930) Reported by: tilghman Patches:
	  20090420__bug14930.diff.txt uploaded by tilghman (license 14)
	  Tested by: mvanbaak, tilghman ........

2009-04-22 21:15 +0000 [r190057]  Jeff Peeler <jpeeler@digium.com>

	* funcs/func_groupcount.c, main/app.c, include/asterisk/channel.h,
	  main/cli.c: Fix building of chan_h323 with gcc-3.3 There seems to
	  be a bug with old versions of g++ that doesn't allow a structure
	  member to use the name list. Rename list member to group_list in
	  ast_group_info and change the few places it is used. (closes
	  issue #14790) Reported by: stuarth

2009-04-22 20:07 +0000 [r190000]  Terry Wilson <twilson@digium.com>

	* funcs/func_strings.c: Add funcs for manipulating delimited lists
	  in the dialplan Adds PUSH and POP for appending to and
	  retrieving/removing from the end of a list and UNSHIFT and SHIFT
	  for insert to and retrieiving/ removing from the beginning of a
	  list. Review: http://reviewboard.digium.com/r/230

2009-04-22 19:23 +0000 [r189993]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/chan_h323.h: Make chan_h323 respect packetization
	  settings and fix small reload issue. Previously, packetization
	  settings were ignored and now they are not. A new config option
	  'autoframing' has been added to mirror the way chan_sip handles
	  it. Turning on the autoframing option (available both as a global
	  option or per peer) overrides the local settings with the remote
	  packetization settings. Testing was performed with varying
	  packetization levels with the following codecs: ulaw, alaw, gsm,
	  and g729. Also, an unrelated config reload issue has been fixed
	  in the case of the config file not changing. (closes issue
	  #12415) Reported by: pj Patches:
	  2009012200_h323packetization.diff.txt uploaded by mvanbaak
	  (license 7), modified by me

2009-04-22 16:56 +0000 [r189951]  Russell Bryant <russell@digium.com>

	* main/features.c: Fix call parking callback. Pipes -> Commas.

2009-04-22 16:01 +0000 [r189911]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_unistim.c: Do not continue to receive DTMF, when
	  the channel is hungup and about to be destroyed. (closes issue
	  #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt
	  uploaded by tilghman (license 14) Tested by: barryf

2009-04-22 14:30 +0000 [r189850]  Michiel van Baak <michiel@vanbaak.info>

	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 189849
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009)
	  | 12 lines replace sed with tr to remove \r from downloaded file
	  On some systems, sed does not recognize \r in the pattern the way
	  it was used here. Use tr instead because this works the same
	  across systems. (closes issue #14936) Reported by: leobrown
	  Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license
	  7) Tested by: leobrown, mvanbaak ........

2009-04-22 06:33 +0000 [r189813]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Detect liblua on SuSE, and add libm for
	  linking for Fedora. (Reported via the -dev list, Subject:
	  Compiling Asterisk with LUA)

2009-04-21 20:28 +0000 [r189771]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: Fixes segfault when switching UDP to TCP in
	  sip.conf after reload. If transport in sip.conf is switched from
	  UDP to TCP, Asterisk segfaults right after issuing a sip reload.
	  The problem is the socket type is changed to TCP but the fd may
	  still be present for UDP. Later, when the TCP session should be
	  created or set using an existing one, it isn't because the old
	  file descriptor is still present. Now every time transport is
	  changed during a sip.conf reload, the file descriptor is set to
	  -1, signifying it must be created or found. (closes issue #14727)
	  Reported by: pj Tested by: dvossel Review:
	  http://reviewboard.digium.com/r/229/

2009-04-21 17:44 +0000 [r189735]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  configs/misdn.conf.sample, CHANGES, channels/misdn/isdn_lib.c,
	  channels/misdn_config.c: Added CCBS/CCNR Party A support and
	  enhanced COLP support. This change adds the following features to
	  chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. *
	  Enhances COLP support for call diversion and explicit call
	  transfer. These enhanced features require a modified version of
	  mISDN. The latest modified mISDN v1.1.x based version is
	  available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk
	  http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged
	  versions of the modified mISDN code are available under:
	  http://svn.digium.com/svn/thirdparty/mISDN/tags
	  http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review:
	  http://reviewboard.digium.com/r/218/ Merged from
	  team/rmudgett/misdn_facility branch.

2009-04-21 15:54 +0000 [r189629-189665]  Doug Bailey <dbailey@digium.com>

	* utils/muted.c, /: Merged revisions 189664 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189664 | dbailey | 2009-04-21 10:52:13 -0500 (Tue, 21 Apr 2009)
	  | 2 lines Remove daemon call on systems that do not support
	  forking. ........

	* /, configure, include/asterisk/autoconfig.h.in,
	  include/asterisk/compat.h, configure.ac: Merged revisions 189601
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009)
	  | 3 lines Add check in configure script to check for GLOB_NOMAGIC
	  and GLOB_BRACE in glob.h This allows config.c to compile when
	  linked against uclibc that does not support these parameters
	  ........

2009-04-20 22:10 +0000 [r189539]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Use nanosleep instead of poll. This is
	  not just because mmichelson suggested it, but also because Mac OS
	  X puked on my poll().

2009-04-20 21:29 +0000 [r189495-189516]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, /: Merged revisions 189465 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
	  | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
	  set ........

	* apps/app_dial.c, /: Merged revisions 189463 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009)
	  | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........

2009-04-20 21:09 +0000 [r189464]  Sean Bright <sean@malleable.com>

	* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189462 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
	  2009) | 13 lines Properly handle @s within hints in AEL. AEL was
	  not handling the case of a device hint containing an @ symbol,
	  which caused parking hints (e.g. hint(park:exten@context)) to
	  error out the parser. This patch makes AEL treat the @ the same
	  way it treats colon and ampersand now, meaning the characters are
	  included in verbatim. (closes issue #14941) Reported by: bpgoldsb
	  Patches: bug14941.patch uploaded by seanbright (license 71)
	  Tested by: bpgoldsb ........

2009-04-20 19:28 +0000 [r189419]  Doug Bailey <dbailey@digium.com>

	* main/manager.c, /, main/db1-ast/recno/rec_open.c,
	  channels/chan_iax2.c: Merged revisions 189391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009)
	  | 4 lines Clean up problem with manager implementation of mmap
	  where it was not testing against MAP_FAILED response. Got rid of
	  shadowed variable used in processign the mmap results. Change
	  test of mmap results to compare against MAP_FAILED ........

2009-04-20 17:05 +0000 [r189350]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug with non-UDP connections that
	  caused dialogs to not get freed. This issue crept up because of a
	  reference count issue on non-UDP based dialogs. The dialog
	  reference count was increased when transmitting a packet reliably
	  but never decreased. This caused the dialog structure to hang
	  around despite being unlinked from the dialogs container. (closes
	  issue #14919) Reported by: vrban

2009-04-20 14:05 +0000 [r189278]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 189277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
	  2009) | 12 lines Move the check for chan->fdno == -1 to after the
	  zombie/hangup check. Many users were finding that their hung up
	  channels were staying up and causing 100% CPU usage. (issue
	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
	  uploaded by mmichelson (license 60) Tested by: falves11, bamby
	  ........

2009-04-18 01:28 +0000 [r189204]  David Vossel <dvossel@digium.com>

	* /, channels/chan_agent.c: Merged revisions 189203 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17
	  Apr 2009) | 12 lines Fixed autologoff in agents.conf not working
	  when agent logs in via AgentLogin app An agent logs in by calling
	  an extension that calls the AgentLogin app. In agents.conf
	  ackcall=always is set, so when they get a call they have the
	  choice to either acknowledge it or ignore it. autologoff=10 is
	  set as well, so if the agent ignores the call over 10sec one may
	  assume that the agent should be logged out (and in this case
	  hungup on as well), but this was not happening. (closes issue
	  #14091) Reported by: evandro Patches: autologoff.diff uploaded by
	  dvossel (license 671) Review:
	  http://reviewboard.digium.com/r/225/ ........

2009-04-17 21:48 +0000 [r189137]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
	  revisions 188833,189134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
	  | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
	  Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
	  rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
	  Modifed/added some debug messages. JIRA ABE-1835 ........

2009-04-17 20:20 +0000 [r189097]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Prevent a crash when SIP blonde transferring
	  an unbridged call. If one attempts to use the attended transfer
	  button on a SIP phone to transfer an unbridged call (such as a
	  call to an IVR) but hangs up while the target of the transfer is
	  still ringing, we need to not crash. The problem was that
	  ast_hangup was called from outside the channel thread. AST-211

2009-04-17 19:36 +0000 [r189077]  Sean Bright <sean@malleable.com>

	* main/asterisk.c: Fix copy/paste error with 'transmit silence'
	  flag.

2009-04-17 15:44 +0000 [r189010]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 189009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
	  2009) | 5 lines Make Busy() application set the CDR disposition
	  to BUSY. (closes issue #14306) Reported by: cristiandimache
	  ........

2009-04-17 14:44 +0000 [r188947]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 188946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
	  15 lines Fix a bug where a value used to create the channel name
	  was bogus. This commit fixes the scenario where an incoming call
	  is authenticated using a peer entry. Previously the channel name
	  was created using either the username setting from the sip.conf
	  entry or the IP address that the call came from. Now the channel
	  name will be created using the peer name itself. This commit will
	  not change the way the channel name is generated for users or
	  friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
	  chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
	  Nick_Lewis, file ........

2009-04-17 14:33 +0000 [r188942]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Fix a spacing issue that I claimed I would when I
	  committed this code. Nothing major though.

2009-04-17 14:26 +0000 [r188938]  Joshua Colp <jcolp@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 188937 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr
	  2009) | 4 lines Fix a situation where the DAHDI channel private
	  structure lock was not unlocked when it should have been. (issue
	  AST-210) ........

2009-04-17 13:29 +0000 [r188901]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Several fixes to the extenpatternmatchnew logic. 1.
	  Differentiate between literal characters in an extension and
	  characters that should be treated as a pattern match. Prior to
	  these fixes, an extension such as NNN would be treated as a
	  pattern, rather than a literal string of N's. 2. Fixed the logic
	  used when matching an extension with a bracketed expression, such
	  as 2[5-7]6. 3. Removed all areas of code that were executed when
	  NOT_NOW was #defined. The code in these areas had the potential
	  to crash, for one thing, and the actual intent of these blocks
	  seemed counterproductive. 4. Fixed many many coding guidelines
	  problems I encountered while looking through the corresponding
	  code. 5. Added failure cases and warning messages for when
	  duplicate extensions are encountered. 6. Miscellaneous fixes to
	  incorrect or redundant statements. (closes issue #14615) Reported
	  by: steinwej Tested by: mmichelson Review:
	  http://reviewboard.digium.com/r/194/

2009-04-16 21:57 +0000 [r188774-188836]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 188835 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
	  | 7 lines Only update realtime, if global option rtupdate !=
	  false (closes issue #14885) Reported by: deepesh Patches:
	  20090413__bug14885.diff.txt uploaded by tilghman (license 14)
	  Tested by: deepesh ........

	* /, apps/app_voicemail.c: Merged revisions 188773 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16
	  Apr 2009) | 4 lines Umask should not be exported into global
	  namespace. (closes issue #14912) Reported by: jcapp ........

2009-04-16 19:30 +0000 [r188742]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c: SIP state notify reorganization What I've
	  done here is simply break up how a state NOTIFY is built.
	  Originally both the XML and sip header information were built
	  within the same function. While this does work, it does not allow
	  for the creation of multipart/related message bodies that can
	  contain multiple XML entries with only one sip header. Now a
	  separate function builds the XML for each notify. This patch also
	  makes maintaining and modifying state notifications in the future
	  much less of a pain. Review: http://reviewboard.digium.com/r/224/

2009-04-16 13:42 +0000 [r188705]  Joshua Colp <jcolp@digium.com>

	* channels/chan_dahdi.c: Fix a bug with the dahdi_setoption
	  callback in chan_dahdi. This function incorrectly reported
	  success even if the option was unsupported. This was exposed by
	  the options to change the underlying channel format. The function
	  now returns a failure if the option is unsupported.

2009-04-15 22:10 +0000 [r188647]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 188646 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15
	  Apr 2009) | 12 lines National prefix inserted even when caller ID
	  not available When the caller ID is restricted, the expected
	  behavior is for the caller id to be blank. In chan_dahdi, the
	  national prefix is placed onto the callers number even if its
	  restricted (empty) causing the caller id to be the national
	  prefix rather than blank. (closes issue #13207) Reported by:
	  shawkris Patches: national_prefix.diff uploaded by dvossel
	  (license 671) Review: http://reviewboard.digium.com/r/220/
	  ........

2009-04-15 20:17 +0000 [r188544-188585]  Mark Michelson <mmichelson@digium.com>

	* /, main/file.c: Merged revisions 188582 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
	  2009) | 7 lines Update ast_readvideo_callback to match
	  ast_readaudio_callback. This fixes potential refcount errors that
	  may occur on ast_filestreams. AST-208 ........

	* apps/app_dial.c: Make the cancellation of the dial timeout on a
	  call forward optional. This introduces the 'z' option to
	  app_dial. With it set, a call forward will cancel any timeout
	  originally set for this instance of the Dial application. AST-207

2009-04-15 14:57 +0000 [r188515]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c: Don't try to do anything in
	  pri_check_restart with service messages unless libpri supports
	  it.

2009-04-14 23:28 +0000 [r188470]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a couple of queue member reference leaks.

2009-04-14 17:40 +0000 [r188413]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Fix an incorrect clock rate when sending
	  T140 text. (closes issue #14029) Reported by: epicac

2009-04-14 16:49 +0000 [r188342-188378]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_dahdi.c, CHANGES: change some capitalization

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
	  service maintenance message support This is the companion commit
	  to libpri r732. Service messages are now supported for switch
	  types 4ess/5ess. A new option service_message_support has been
	  added to chan_dahdi.conf and is noted in the sample config file.
	  The service message support is turned off by default. The current
	  implementation relies on AstDB to keep track of channel state,
	  which allows the statuses to be preserved across Asterisk
	  restarts. Below is a description of the storage format. The state
	  and reason for the service state are in the form
	  <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of
	  Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No
	  reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3'
	  – both NEAR and FAR END The new CLI commands to handle channel
	  service state are: pri service disable channel <chan> pri service
	  enable channel <chan> Many people contributed to the development
	  of this functionality. Because I entered at the very end I do not
	  know the exact history. Special thanks to all who moved the bug
	  forward one way or another: cmaj, PCadach, markster, mattf,
	  drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman,
	  lmadsen, and especially dhubbard (he answered lots of my
	  questions and did a large portion of the work) (closes issue
	  #3450) Reported by: cmaj

2009-04-14 14:22 +0000 [r188283-188284]  Olle Johansson <oej@edvina.net>

	* doc/manager_1_1.txt: New actions should go under "New Actions",
	  not "new events"

	* main/xmldoc.c, apps/app_jack.c: Making sure we have references to
	  external libraries. Note: Update h.323 with the recent changes
	  too

2009-04-14 13:14 +0000 [r188247]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug with the change I made yesterday
	  to outbound proxy support. Per discussion with oej on IRC we need
	  the actual IP address, not the outbound proxy IP address, in the
	  sa field. This change matches the already existing code for all
	  other uses of the outbound proxy setting.

2009-04-14 05:45 +0000 [r188206-188210]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: As suggested by Russell, warn users when their
	  dialplan arguments contain pipes, but not commas.

	* utils/smsq.c: Application delimiter is ',', not '|'. (closes
	  issue #14881) Reported by: stegro Patches: smsq.patch uploaded by
	  stegro (license 752)

2009-04-13 19:31 +0000 [r188102]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c: Fix another crash related to cached
	  realtime music on hold. This was another off-by-one problem
	  caused by moh_register.

2009-04-13 16:28 +0000 [r188067]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix a bug where using an outbound proxy
	  would cause the local address to be 127.0.0.1. Copy the outbound
	  proxy IP address into the SIP dialog structure as the IP address
	  we will be sending to. This has to be done because the logic that
	  determines what local IP address to use in the SIP messages is
	  not aware of an outbound proxy being in place. It only knows what
	  IP address we are sending to. (closes issue #12006) Reported by:
	  mnicholson

2009-04-13 14:17 +0000 [r188032]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Set all queue variables on both the caller and
	  member channels. This allows for the variables to be accessed if
	  a member macro is run. Thanks to Grigoriy Puzankin for bringing
	  this up on the -dev list.

2009-04-10 20:26 +0000 [r187906]  Jeff Peeler <jpeeler@digium.com>

	* channels/Makefile: Fix module embedding for chan_h323. Include
	  libchanh323.a in the modules.link file so that all the symbols
	  can be resolved at link time. (closes issue #11966) Reported by:
	  dome Patches: issue_11966.patch uploaded by kpfleming (license
	  421) Tested by: jpeeler

2009-04-10 18:56 +0000 [r187830]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Indicating connected line or redirecting
	  updates were missing a call to lock the local_pvt.

2009-04-10 18:14 +0000 [r187772-187773]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, main/rtp_engine.c: Change how we set the
	  local and remote address. The code will now only change the
	  address and port. It will not overwrite any other values.

	* channels/chan_jingle.c, channels/chan_unistim.c,
	  res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_gtalk.c, channels/chan_mgcp.c: Fix some
	  uninitialized memory notices that appeared under valgrind.

2009-04-10 17:32 +0000 [r187770]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Make sure tc is unlocked before calling ast_call
	  since calling a Local channel could result in a deadlock.

2009-04-10 17:29 +0000 [r187764]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/realtime_pgsql.sql, /,
	  contrib/scripts/sip-friends.sql: Merged revisions 187763 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009)
	  | 2 lines Add lastms column to the contributed table designs
	  ........

2009-04-10 16:51 +0000 [r187721]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/embed_modules.xml: clean up some patterns for files
	  to remove add embedding support for bridge and test modules

2009-04-10 16:26 +0000 [r187680-187714]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: ast_strdup failures aren't really failures
	  if the original value was NULL.

	* main/channel.c: Don't let ast_channel_alloc fail if explicitly
	  passed NULL cid_name or cid_number. This also fixes a small
	  memory leak.

2009-04-10 16:00 +0000 [r187675]  Russell Bryant <russell@digium.com>

	* tests/test_heap.c, tests/test_sched.c: Disable test modules by
	  default.

2009-04-10 15:59 +0000 [r187674]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Ensure pvt is not NULL before dereferencing
	  it. (closes issue #14784) Reported by: pj

2009-04-10 15:49 +0000 [r187673]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c: Even more changes concerning r187426. Revised
	  where locks are placed yet once again. ast_call() should not be
	  called with a channel locked. could cause deadlock issues with
	  local channels.

2009-04-10 15:11 +0000 [r187636]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
	  configs/logger.conf.sample: revert addition of LOG_SECURITY log
	  channel; after further discussion, a much better solution will be
	  used

2009-04-10 14:53 +0000 [r187634-187635]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: Miscellaneous minor changes to
	  chan_misdn. * Miscellaneous spacing and comment changes. * Minor
	  code rearangements. * Miscellaneous doxygen comments.

	* channels/chan_misdn.c: Make chan_misdn_log() avoid generating the
	  log message if logging is disabled.

2009-04-10 03:55 +0000 [r187599]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, main/pbx.c, main/manager.c,
	  include/asterisk/linkedlists.h, main/features.c, main/http.c,
	  main/app.c, include/asterisk/lock.h, main/audiohook.c,
	  main/bridging.c: Modify headers and macros, according to
	  Russell's suggestions on the -dev list

2009-04-09 21:06 +0000 [r187560]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Add a new option,
	  mwi_from, to sip.conf. This allows for you to change the From
	  header for outgoing MWI NOTIFY requests. Prior to this, the best
	  you could do was to set a callerid in the general section of
	  sip.conf. The problem was that this was used for all outbound
	  requests, not just MWI NOTIFY requests. AST-201

2009-04-09 20:40 +0000 [r187556]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c: More changes concerning r187426. Revised where
	  locks are placed.

2009-04-09 19:10 +0000 [r187491]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
	  ability for dialplan execution to continue when caller hangs up.
	  The F option to app_dial has been modified to accept no
	  parameters and perform the above functionality. I don't see
	  anywhere else that is doing function overloading, but this really
	  is the best place for this operation because: - It makes it close
	  to the 'g' option in the argument list which provides similar
	  functionality. - The existing code to support the current F
	  option provides a very convienient location to add this new
	  feature. (closes issue #12381) Reported by: michael-fig

2009-04-09 18:58 +0000 [r187488]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 187484 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
	  2009) | 18 lines Handle a SIP race condition (reinvite before an
	  ACK) properly. RFC 5047 explains the proper course of action to
	  take if a reINVITE is received before the ACK from a previous
	  invite transaction. What we are to do is to treat the reINVITE as
	  if it were both an ACK and a reINVITE and process it normally.
	  Later, when we receive the ACK we had been expecting, we will
	  ignore it since its CSeq is less than the current iseqno of the
	  sip_pvt representing this dialog. (closes issue #13849) Reported
	  by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
	  (license 60) Tested by: mmichelson, klaus3000 ........

2009-04-09 18:40 +0000 [r187483]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /, include/asterisk/linkedlists.h,
	  include/asterisk/lock.h: Merged revisions 187428 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09
	  Apr 2009) | 8 lines Race condition between ast_cli_command() and
	  'module unload' could cause a deadlock. Add lock timeouts to
	  avoid this potential deadlock. (closes issue #14705) Reported by:
	  jamessan Patches: 20090320__bug14705.diff.txt uploaded by
	  tilghman (license 14) Tested by: jamessan ........

2009-04-09 17:39 +0000 [r187426]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c: Fixes deadlock caused by calling get_cid_name
	  with chan locked. get_cid_name should not be called with a
	  channel lock. get_cid_name calls ast_get_hint which eventually
	  calls pbx_find_extension. pbx_find_extension starts and stops
	  autoservice which should not be done with a channel lock, so
	  get_cid_name should not be called with one.

2009-04-09 17:34 +0000 [r187421-187424]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c: Use safe macro practices even though they
	  really aren't necessary.

	* res/res_musiconhold.c: Fix a crash in res_musiconhold when using
	  cached realtime moh. The moh_register function links an mohclass
	  and then immediately unrefs the class since the container now has
	  a reference. The problem with using realtime music on hold is
	  that the class is allocated, registered, and started in one fell
	  swoop. The refcounting logic resulted in the count being off by
	  one. The same problem did not happen when using a static config
	  because the allocation and registration of an mohclass is a
	  separate operation from starting moh. This also did not affect
	  non-cached realtime moh because the classes are not registered at
	  all. I also have modified res_musiconhold to use the _t_ variants
	  of the ao2_ functions so that more info can be gleaned when
	  attempting to trace the refcounts. I found this to be incredibly
	  helpful for debugging this issue and there's no good reason to
	  remove it. (closes issue #14661) Reported by: sum

2009-04-09 17:20 +0000 [r187363-187381]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Allow '/' in username portion of register;
	  this is a regression. (closes issue #14668) Reported by: Netview

	* /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
	  187362 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
	  | 3 lines Permit zero-length text messages in SIP. (Related to an
	  issue posted to the -users list, subject "AEL2, BASE64_DECODE and
	  hexadecimal") ........

2009-04-09 16:27 +0000 [r187360-187361]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Do not try to send the format read/format
	  write/make compatible options over IAX2.

	* main/channel.c, channels/chan_sip.c, include/asterisk/frame.h:
	  Add support for allowing the channel driver to handle
	  transcoding. This was accomplished using a set of options and the
	  setoption channel callback. The core calls into the channel
	  driver using these options and the channel driver either returns
	  success or failure.

2009-04-09 04:59 +0000 [r187302]  Tilghman Lesher <tlesher@digium.com>

	* agi/Makefile, build_tools/cflags.xml, utils/Makefile,
	  include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c
	  (added), main/asterisk.c: Merged revisions 187300-187301 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
	  | 3 lines Add debugging mode for diagnosing file descriptor
	  leaks. (Related to issue #14625) ........ r187301 | tilghman |
	  2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
	  missed this file in the last commit. ........

2009-04-09 02:44 +0000 [r187269]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
	  configs/logger.conf.sample: add a dedicated log channel for
	  modules to be able report security-related events, so that they
	  can be fed into external processes for analysis and possible
	  mitigation efforts (inspired by this evening's Toronto Asterisk
	  Users Group meeting and previous dicussions amongst various
	  community members)

2009-04-08 21:00 +0000 [r187211]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, main/features.c, include/asterisk/channel.h: Add
	  timer for features so that backup bridge config can go away The
	  biggest change done here was elimination of the backup_config for
	  use with features. Previously, the bridging code upon detecting a
	  feature would set the start time of the bridge to the start time
	  of the feature. Then after the feature had either expired or
	  timed out the start time would be reset to the true bridge start
	  time from the backup_config. Now, the time differences are
	  calculated with respect to the newly added feature_start_time
	  timeval instead. There should be no behavior changes from the
	  previous functionality aside from the bridge timing being
	  unaffected by either valid or partial feature matches. Previously
	  the timing would be increased by the length of time configured
	  for featuredigittimeout, which was probably never noticed.
	  (closes issue #14503) Reported by: KNK Tested by: jpeeler Review:
	  http://reviewboard.digium.com/r/179/

2009-04-08 20:39 +0000 [r187210]  Tilghman Lesher <tlesher@digium.com>

	* /: Recorded merge of revisions 187209 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009)
	  | 4 lines Backport resolution for file descriptor leak in 1.6.0
	  to 1.4. This fixes short reads in http manager sessions, such as
	  those done by the ast-gui branch. (Fixes AST-198) ........

2009-04-08 19:59 +0000 [r187179]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxyref.h,
	  include/asterisk/doxygen/reviewboard.h (added): Add documentation
	  for reviewboard usage and guidelines.

2009-04-08 18:12 +0000 [r187108]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c: Fix a bug where we would native bridge when we
	  did not want to.

2009-04-08 17:51 +0000 [r187105]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove duplicate prototype for temp_peer().

2009-04-08 17:08 +0000 [r187050]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c: If the first column is empty, output a
	  delimiter anyway. (closes issue #14848) Reported by: john8675309
	  Patches: 20090408__bug14848.diff.txt uploaded by tilghman
	  (license 14) Tested by: john8675309

2009-04-08 16:52 +0000 [r187046]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 187045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed,
	  08 Apr 2009) | 10 lines Fix a small logical error when loading
	  moh classes. We were unconditionally incrementing the number of
	  mohclasses registered. However, we should actually only increment
	  if the call to moh_register was successful. While this probably
	  has never caused problems, I noticed it and decided to fix it
	  anyway. ........

2009-04-08 16:27 +0000 [r187036]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c, main/rtp_engine.c: Turn a warning message
	  into a debug message and do not treat two situations as errors
	  when they are not.

2009-04-08 15:27 +0000 [r186985]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 186984 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
	  2009) | 24 lines Make a couple of changes with regards to a new
	  message printed in ast_read(). "ast_read() called with no
	  recorded file descriptor" is a new message added after a bug was
	  discovered. Unfortunately, it seems there are a bunch of places
	  that potentially make such calls to ast_read() and trigger this
	  error message to be displayed. This commit does two things to
	  help to make this message appear less. First, the message has
	  been downgraded to a debug level message if dev mode is not
	  enabled. The message means a lot more to developers than it does
	  to end users, and so developers should take an effort to be sure
	  to call ast_read only when a channel is ready to be read from.
	  However, since this doesn't actually cause an error in operation
	  and is not something a user can easily fix, we should not spam
	  their console with these messages. Second, the message has been
	  moved to after the check for any pending masquerades. ast_read()
	  being called with no recorded file descriptor should not
	  interfere with a masquerade taking place. This could be seen as a
	  simple way of resolving issue #14723. However, I still want to
	  try to clear out the existing ways of triggering this message,
	  since I feel that would be a better resolution for the issue.
	  ........

2009-04-08 13:38 +0000 [r186928-186957]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxygen/releases.h: Add some additional notes on
	  release numbering.

	* Makefile, include/asterisk/doxygen/releases.h (added),
	  include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
	  include/asterisk/doxygen (added),
	  include/asterisk/doxygen/commits.h (added),
	  include/asterisk/doxygen/licensing.h (added), main/asterisk.c:
	  Start splitting up miscellaneous doxygen documentation into
	  separate files. doxyref.h was created to hold miscellaneous
	  documentation that was not specific to a part of the code. This
	  file has grown quite a bit so I decided to start splitting parts
	  of it out into new files. Now, you can drop a new file into
	  include/asterisk/doxygen/ and it will be processed by doxygen.

	* channels/chan_sip.c: Update some comments and resolve potential
	  memory corruption in chan_sip. While browsing chan_sip the other
	  day, I noticed this dangerous code in dialog_needdestroy(). This
	  function is an ao2_callback. It is absolutely _not_ okay to
	  unlock the container from within this function. It's also not
	  clear why it was useful. Given that it could cause memory
	  corruption, I have removed it. There was also a TODO comment left
	  describing a potential implementation of an improvement to the
	  needdestroy handling. I'm not convinced that what was described
	  is the best choice here, so I have briefly described the way that
	  this function is used today that could be improved.

2009-04-08 05:06 +0000 [r186899]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Add lastms to the require API call.

2009-04-08 00:09 +0000 [r186833-186842]  Mark Michelson <mmichelson@digium.com>

	* /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
	  revisions 186841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
	  2009) | 8 lines Fix a few typos of the word "frequency." (closes
	  issue #14842) Reported by: jvandal Patches: frequency-typo.diff
	  uploaded by jvandal (license 413) ........

	* channels/chan_sip.c: Fix bad merge from fix for issue 13867.
	  (closes issue #14686) Reported by: davidw

	* main/channel.c, /: Merged revisions 186832 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
	  2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
	  p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
	  warning sounds will not be properly played to either party of the
	  bridge. (closes issue #14845) Reported by: adomjan ........

2009-04-07 22:23 +0000 [r186799]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 186775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
	  | 3 lines Fix Macro documentation to match current (and intended)
	  behavior. (See -dev mailing list) ........

2009-04-07 20:46 +0000 [r186720]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 186719 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
	  2009) | 6 lines Ensure that \r\n is printed after the ActionID in
	  an OriginateResponse. (closes issue #14847) Reported by: kobaz
	  ........

2009-04-06 23:11 +0000 [r186624-186687]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c: Fix a log message getting output when it
	  should not have been.

	* channels/chan_sip.c: Fix problem when authenticating a non-RTP
	  dialog.

	* channels/chan_sip.c, doc/tex/channelvariables.tex, CHANGES: Add
	  support for changing the outbound codec on a SIP call using a
	  dialplan variable. This adds a dialplan variable
	  (SIP_CODEC_OUTBOUND) which controls the codec offered for an
	  outgoing SIP call. This is much like the SIP_CODEC dialplan
	  variable and has the same restrictions. The codec set must be one
	  that is configured for the call. (closes issue #13243) Reported
	  by: samdell3 Patches: 13243.diff uploaded by file (license 11)

2009-04-06 16:06 +0000 [r186620]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_connectedline.c (added), funcs/func_redirecting.c
	  (added): Silly svn. These files didn't get merged over in the
	  merge of the issue8824 branch.

2009-04-06 13:23 +0000 [r186563]  Joshua Colp <jcolp@digium.com>

	* main/rtp_engine.c: Pass the correct value to sizeof when copying
	  address information. (issue #14827) Reported by: pj Patches:
	  14827.diff uploaded by file (license 11) Tested by: pj

2009-04-04 00:13 +0000 [r186537]  Richard Mudgett <rmudgett@digium.com>

	* /: Remove merged branch properties accidentally merged to trunk.

2009-04-03 22:41 +0000 [r186525]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_unistim.c, channels/misdn/isdn_lib_intern.h,
	  channels/chan_local.c, main/rtp_engine.c, /,
	  channels/misdn/isdn_msg_parser.c, channels/chan_iax2.c,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  include/asterisk/callerid.h, main/channel.c, main/dial.c,
	  channels/misdn/isdn_lib.h, channels/chan_dahdi.c,
	  channels/chan_phone.c, channels/chan_skinny.c, main/features.c,
	  configs/sip.conf.sample, include/asterisk/frame.h,
	  include/asterisk/rtp_engine.h, channels/chan_mgcp.c,
	  apps/app_dial.c, res/res_rtp_asterisk.c, main/stun.c,
	  channels/chan_sip.c, channels/chan_agent.c,
	  configs/misdn.conf.sample, include/asterisk/channel.h, CHANGES,
	  apps/app_queue.c, channels/chan_misdn.c,
	  apps/app_directed_pickup.c, channels/misdn/chan_misdn_config.h,
	  channels/chan_h323.c, main/callerid.c, include/asterisk/stun.h:
	  This commit introduces COLP/CONP and Redirecting party
	  information into Asterisk. The channel drivers which have been
	  most heavily tested with these enhancements are chan_sip and
	  chan_misdn. Further work is being done to add Q.SIG support and
	  will be introduced in a later commit. chan_skinny has code added
	  to it here, but according to user pj, the support on chan_skinny
	  is not working as of now. This will be fixed in a later commit. A
	  special thanks goes out to bugtracker user gareth for getting the
	  ball rolling and providing the initial support for this work.
	  Without his initial work on this, this would not have been nearly
	  as painless as it was. This functionality has been tested by
	  Digium's product quality department, as well as a customer site
	  running thousands of calls every day. In addition, many many many
	  many bugtracker users have tested this, too. (closes issue #8824)
	  Reported by: gareth Review: http://reviewboard.digium.com/r/201

2009-04-03 20:20 +0000 [r186461]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 186458 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03
	  Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would
	  not properly switch formats when requested Don't offer
	  AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
	  provide a slight performance benefit, the translation core in
	  Asterisk has some flaws when a channel driver offers multiple raw
	  formats. this fix is much simpler than fixing the translation
	  core to solve that issue (although that will be done later).
	  ........

2009-04-03 19:59 +0000 [r186444-186447]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 186445 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03
	  Apr 2009) | 2 lines Found a conflict in the last commit, due to
	  multiple targets ........

	* /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
	  revisions 186415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
	  | 7 lines Distinguish in a sent email between simple sends and
	  forwards. (closes issue #11678) Reported by: jamessan Patches:
	  20090330__bug11678.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman, lmadsen ........

2009-04-03 16:47 +0000 [r186382]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, channels/chan_sip.c, channels/chan_iax2.c,
	  include/asterisk/frame.h: Add better support for relaying success
	  or failure of the ast_transfer() API call. This API call now
	  waits for a special frame from the underlying channel driver to
	  indicate success or failure. This allows the return value to
	  truly convey whether the transfer worked or not. In the case of
	  the Transfer() dialplan application this means the value of the
	  TRANSFERSTATUS dialplan variable is actually true. (closes issue
	  #12713) Reported by: davidw Tested by: file

2009-04-03 16:29 +0000 [r186379]  David Vossel <dvossel@digium.com>

	* main/audiohook.c: audio_audiohook_write_list() did not correctly
	  update sample size after ast_translate.
	  audio_audiohook_write_list() did not take into account that the
	  sample size may change after translation depending on if the
	  original frame is is 8khz or 16khz. the sample size is now
	  updated after translating to reflect this possibility. This
	  caused the audio on the receiving end to sound terrible. Thanks
	  to jcolp and mmichelson for helping me work this out. (issue
	  AST-197)

2009-04-03 15:52 +0000 [r186321]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/crypto.h, /: Merged revisions 186320 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
	  lines Fix a problem with the crypto variable definitions not
	  actually being defined properly. (closes issue #14804) Reported
	  by: jvandal ........

2009-04-03 15:18 +0000 [r186297]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Compatibility fix for glibc 2.4 (Closes
	  issue #14820) Reported by: phsultan

2009-04-03 14:32 +0000 [r186286]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix the ability to retrieve voicemail
	  messages from IMAP. A recent change made interactive vm_states no
	  longer get added to the list of vm_states and instead get stored
	  in thread-local storage. In trunk and all the 1.6.X branches, the
	  problem is that when we search for messages in a voicemail box,
	  we would attempt to update the appropriate vm_state struct by
	  directly searching in the list of vm_states instead of using the
	  get_vm_state_by_imap_user function. This meant we could not find
	  the interactive vm_state that we wanted. (closes issue #14685)
	  Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
	  (license 60) Tested by: BlargMaN, qualleyiv, mmichelson

2009-04-03 02:03 +0000 [r186230]  Russell Bryant <russell@digium.com>

	* /, cdr/cdr_radius.c: Merged revisions 186229 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
	  | 21 lines Fix a memory leak in cdr_radius. I came across this
	  while doing some testing of my ast_channel_ao2 branch. After
	  running a test overnight that generated over 5 million calls,
	  Asterisk had taken up about 1 GB of my system memory. So, I
	  re-ran the test with MALLOC_DEBUG turned on. However, it showed
	  no leaks in Asterisk during the test, even though Asterisk was
	  still consuming it somehow. Instead, I turned to valgrind, which
	  when run with --leak-check=full, told me exactly where the leak
	  came from, which was from allocations inside the radiusclient-ng
	  library. This explains why MALLOC_DEBUG did not report it. After
	  a bit of analysis, I found that we were leaking a little bit of
	  memory every time a CDR record was passed to cdr_radius. I don't
	  actually have a radius server set up to receive CDR records.
	  However, I always have my development systems compile and install
	  all modules. In addition to making sure there are not build
	  errors across modules, always loading modules helps find bugs
	  like this, too, so it is strongly recommend for all developers.
	  ........

2009-04-02 21:56 +0000 [r186175]  Mark Michelson <mmichelson@digium.com>

	* /, configs/features.conf.sample: Merged revisions 186174 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
	  2009) | 5 lines Fix instructions in one-step parking comment to
	  make more sense. Changed a capital K to a lowercase k. ........

2009-04-02 17:26 +0000 [r186101]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 186081 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
	  Apr 2009) | 3 lines ensure that the buffer passed to
	  DAHDI_SET_BUFINFO is fully initialized ........

2009-04-02 17:20 +0000 [r186078]  Joshua Colp <jcolp@digium.com>

	* res/res_rtp_asterisk.c (added), channels/chan_unistim.c,
	  apps/app_dial.c, main/stun.c (added), main/rtp_engine.c (added),
	  channels/chan_local.c, channels/chan_sip.c,
	  channels/chan_bridge.c, main/Makefile, channels/chan_agent.c,
	  include/asterisk/rtp.h (removed), UPGRADE.txt,
	  channels/chan_gtalk.c, include/asterisk/_private.h, main/rtp.c
	  (removed), main/loader.c, channels/chan_jingle.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  configs/sip.conf.sample, include/asterisk/stun.h (added),
	  include/asterisk/rtp_engine.h (added), main/asterisk.c,
	  channels/chan_mgcp.c: Merge in the RTP engine API. This API
	  provides a generic way for multiple RTP stacks to be integrated
	  into Asterisk. Right now there is only one present,
	  res_rtp_asterisk, which is the existing Asterisk RTP stack.
	  Functionality wise this commit performs the same as previously.
	  API documentation can be viewed in the rtp_engine.h header file.
	  Review: http://reviewboard.digium.com/r/209/

2009-04-02 17:10 +0000 [r186021-186060]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
	  186059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
	  Apr 2009) | 2 lines Fix for AST-2009-003 ........
	  ................

	* main/strings.c: Missed a common case for needing to extend the
	  buffer. (closes issue #14716) Reported by: sum Patches:
	  20090402__bug14716.diff.txt uploaded by tilghman (license 14)
	  Tested by: sum

2009-04-02 13:51 +0000 [r185953]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 185952 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02
	  Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
	  DAHDI_GET_PARAMS ioctls were recently corrected to show that they
	  do, in fact, read data from userspace as part of their work. due
	  to this fix, valgrind now reports a number of cases where
	  chan_dahdi passed an uninitialized (or partially) buffer to these
	  ioctls, which could lead to unexpected behavior. this patch
	  corrects chan_dahdi to ensure that buffers passed to these ioctls
	  are always fully initialized. ........

2009-04-01 20:13 +0000 [r185912]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
	  main/manager.c, main/tdd.c, include/asterisk/astobj2.h,
	  main/ast_expr2f.c, include/asterisk/pbx.h,
	  include/asterisk/strings.h, main/taskprocessor.c, res/res_odbc.c:
	  Merge changes from str_substitution that are unrelated to that
	  branch. Included is a small bugfix to an ast_str helper, but most
	  of these changes are simply doxygen fixes.

2009-04-01 19:03 +0000 [r185846]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 185845 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
	  | 10 lines Fixes issue with dropped calles due to re-Invite glare
	  and re-Invites never executing after a 491 Acknowledgement for
	  491 responses were never being processed because it didn't match
	  our pending invite's seqno. Since the ACK was never processed,
	  the 491 frame would continue to be retransmitted until eventually
	  the call was dropped due to max retries. Now during a pending
	  invite, if we receive another invite, we send an 491 and hold on
	  to that glare invite's seqno in the "glareinvite" variable for
	  that sip_pvt struct. When ACK's are received, we first check to
	  see if it is in response to our pending invite, if not we check
	  to see if it is in response to a glare invite. In this case, it
	  is in response to the glare invite and must be dealt with or the
	  call is dropped. I've changed the wait time for resending the
	  re-Invite after receving a 491 response to comply with RFC 3261.
	  Before this patch the scheduled re-Invite would only change a
	  flag indicating that the re-Invite should be sent out, now it
	  actually sends it out as well. (closes issue #12013) Reported by:
	  alx Review: http://reviewboard.digium.com/r/213/ ........

2009-04-01 13:59 +0000 [r185777]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Address Russell's comments regarding rev 185704.
	  Use ast_debug and ast_softhangup_nolock.

2009-04-01 13:48 +0000 [r185741-185772]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 185771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
	  | 6 lines Fix a case where DTMF could bypass audiohooks. This
	  change fixes a situation where an audiohook that wants DTMF would
	  not actually get it. This is in the code path where we end DTMF
	  digit length emulation while handling a NULL frame. ........

	* include/asterisk/stringfields.h: Fix dev-mode build on my box.

2009-04-01 00:39 +0000 [r185704]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, CHANGES: Allow the AMI Hangup command to accept a
	  Cause header. (closes issue #14695) Reported by: mneuhauser
	  Patches: cause-for-hangup-manager-action.patch uploaded by
	  mneuhauser (license 425)

2009-03-31 22:35 +0000 [r185664]  Kevin P. Fleming <kpfleming@digium.com>

	* utils: ignore copied (generated) file

2009-03-31 22:12 +0000 [r185600-185604]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix trunk's compilation.

	* /, apps/app_queue.c: Merged revisions 185599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
	  2009) | 6 lines Fix crash that would occur if an empty member was
	  specified in queues.conf. (closes issue #14796) Reported by: pida
	  ........

2009-03-31 21:29 +0000 [r185581]  Kevin P. Fleming <kpfleming@digium.com>

	* main/utils.c, include/asterisk/stringfields.h: Optimizations to
	  the stringfields API This patch provides a number of
	  optimizations to the stringfields API, focused around saving (not
	  wasting) memory whenever possible. Thanks to Mark Michelson for
	  inspiring this work and coming up with the first two
	  optimizations that are represented here: Changes: - Cleanup of
	  some code, fix incorrect doxygen comments - When a field is
	  emptied or replaced with a new allocation, decrease the amount of
	  'active' space in the pool it was held in; if that pool reaches
	  zero active space, and is not the current pool, then free it as
	  it is no longer in use - When allocating a pool, try to allocate
	  a size that will fit in a 'standard' malloc() allocation without
	  wasting space - When allocating space for a field, store the
	  amount of space in the two bytes immediately preceding the field;
	  this eliminates the need to call strlen() on the field when
	  overwriting it, and more importantly it 'remembers' the amount of
	  space the field has available, even if a shorter string has been
	  stored in it since it was allocated - Don't automatically double
	  the size of each successive pool allocated; it's wasteful
	  http://reviewboard.digium.com/r/165/

2009-03-31 19:46 +0000 [r185469]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 185468 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue,
	  31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the
	  word "messages" properly. (closes issue #14736) Reported by:
	  chappell Patches: voicemail_no_messages.diff uploaded by chappell
	  (license 8) ........

2009-03-31 19:07 +0000 [r185432]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Improve performance of the code handling
	  the frame queue in chan_iax2. In my tests that exercised full
	  frame handling in chan_iax2, the version with these changes took
	  30% to 40% of the CPU time compared to the same test of Asterisk
	  trunk before these modifications. While doing some profiling for
	  <http://reviewboard.digium.com/r/205/>, one function that caught
	  my eye was network_thread() in chan_iax2.c. After the things that
	  I was working on there, it was the next target for analysis and
	  optimization. I used oprofile's source annotation functionality
	  and found that the loop traversing the frame queue in
	  network_thread() was to blame for the excessive CPU cycle
	  consumption. The frame_queue in chan_iax2 previously held all
	  frames that either were pending transmission or had been
	  transmitted and are still pending acknowledgment. In
	  network_thread(), the previous code would go back through the
	  main for loop after reading a single incoming frame or after
	  being signaled because a frame had been queued up for initial
	  transmission. In each iteration of the loop, it traverses the
	  entire frame queue looking for frames that need to be
	  transmitted. On a busy server, this could easily be quite a few
	  entries. This patch is actually quite simple. The frame_queue has
	  become only a list of frames pending acknowledgment. Frames that
	  need to be transmitted are queued up to a dedicated transmit
	  thread via the taskprocessor API. As a result, the code in
	  network_thread() becomes much simpler, as its only job is to read
	  incoming frames. In addition to the previously described changes,
	  this patch includes some additional changes to the frame_queue.
	  Instead of one big frame_queue, now there is a list per call
	  number to further reduce wasted list traversals. The biggest
	  impact of this change is in socket_process(). For additional
	  details on testing and test results, see the review request.
	  Review: http://reviewboard.digium.com/r/212/

2009-03-31 16:46 +0000 [r185363]  David Brooks <dbrooks@digium.com>

	* /, channels/chan_gtalk.c: Merged revisions 185362 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31
	  Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when
	  xmpp contains extra whitespaces To drill into the xmpp to find
	  the capabilities between channels, chan_gtalk calls iks_child()
	  and iks_next(). iks_child() and iks_next() are functions in the
	  iksemel xml parsing library that traverse xml nodes. The bug here
	  is that both iks_child() and iks_next() will return the next
	  iks_struct node *regardless* of type. chan_gtalk expects the next
	  node to be of type IKS_TAG, which in most cases, it is, but in
	  this case (a call being made from the Empathy IM client), there
	  exists iks_struct nodes which are not IKS_TAG data (they are
	  extraneous whitespaces), and chan_gtalk doesn't handle that case,
	  so capabilities don't match, and a call cannot be made.
	  iks_first_tag() and iks_next_tag(), on the other hand, will not
	  return the very next iks_struct, but will check to see if the
	  next iks_struct is of type IKS_TAG. If it isn't, it will be
	  skipped, and the next struct of type IKS_TAG it finds will be
	  returned. This assures that chan_gtalk will find the iks_struct
	  it is looking for. This fix simply changes all calls to
	  iks_child() and iks_next() to become calls to iks_first_tag() and
	  iks_next_tag(), which resolves the capability matching. The
	  following is a payload listing from Empathy, which, due to the
	  extraneous whitespace, will not be parsed correctly by iksemel:
	  <iq from='dbrooksjab@235-22-24-10/Telepathy'
	  to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
	  <session xmlns='http://www.google.com/session'
	  initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
	  id='1837267342'> <description
	  xmlns='http://www.google.com/session/phone'> <payload-type
	  clockrate='16000' name='speex' id='96'/> <payload-type
	  clockrate='8000' name='PCMA' id='8'/> <payload-type
	  clockrate='8000' name='PCMU' id='0'/> <payload-type
	  clockrate='90000' name='MPA' id='97'/> <payload-type
	  clockrate='16000' name='SIREN' id='98'/> <payload-type
	  clockrate='8000' name='telephone-event' id='99'/> </description>
	  </session> </iq> Review: http://reviewboard.digium.com/r/181/
	  ........

2009-03-31 14:53 +0000 [r185261]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Don't free() an astobj2 object. (closes issue
	  #14672) Reported by: makoto

2009-03-31 14:07 +0000 [r185197]  Joshua Colp <jcolp@digium.com>

	* /, main/audiohook.c: Merged revisions 185196 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
	  lines Fix crash when moving audiohooks between channels. Handle
	  the scenario where we are called to move audiohooks between
	  channels and the source channel does not actually have any on it.
	  (closes issue #14734) Reported by: corruptor ........

2009-03-30 20:42 +0000 [r185122-185123]  Richard Mudgett <rmudgett@digium.com>

	* /, configs/misdn.conf.sample, channels/misdn_config.c: Merged
	  revisions 185121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
	  | 1 line Update the channel allocation method documentation.
	  ........

	* /, channels/misdn/isdn_lib.c: Merged revisions 185120 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
	  | 19 lines Make chan_misdn BRI TE side normally defer channel
	  selection to the NT side. Channel allocation collisions are not
	  handled by chan_misdn very well. This patch simply avoids the
	  problem for BRI only. For PRI, allocation collisions are still
	  possible but less likely since there are simply more channels
	  available and each end could use a different allocation strategy.
	  misdn.conf options available: te_choose_channel - Use to force
	  the TE side to allocate channels. method - Specify the channel
	  allocation strategy. (closes issue #13488) Reported by:
	  Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
	  Tested by: crich, siepkes, festr ........

2009-03-30 16:26 +0000 [r185072]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 185031 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
	  2009) | 39 lines Fix queue weight behavior so that calls in
	  low-weight queues are not inappropriately blocked. (This is
	  copied and pasted from the review request I made for this patch)
	  Asterisk has some odd behavior when queue weights are used. The
	  current logic used when potentially calling a queue member is: If
	  the member we are going to call is part of another queue and
	  _that other queue has any callers in it_ and has a higher weight
	  than the queue we are calling from, then don't try to contact
	  that member. The issue here is what I have marked with
	  underscores. If the higher-weighted queue has any callers in it
	  at all, then the queue member will be unreachable from the
	  lower-weighted queue. This has the potential to be really really
	  bad if using a queue strategy, such as leastrecent or
	  fewestcalls, with the potential to call the same member
	  repeatedly. The fix proposed by garychen on issue 13220 is very
	  simple and, as far as I can see, works well for this situation.
	  With this set of changes, the logic used becomes: If the member
	  we are going to call is part of another queue, the other queue
	  has a higher weight than the queue we are calling from, and the
	  higher weight queue has at least as many callers as available
	  members, then do not try to contact the queue member. If the
	  higher weighted queue has fewer callers than available members,
	  then there is no reason to deny the call to this member since the
	  other queue can afford to spare a member. Since the fix involved
	  writing a generic function for determining the number of
	  available members in the queue, I also modified the is_our_turn
	  function to make use of the new num_available_members function to
	  determine if it is our turn to try calling a member. There is one
	  small behavior change. Before writing this patch, if you had
	  autofill disabled, then if you were the head caller in a queue,
	  you would automatically be told that it was your turn to try
	  calling a member. This did not take into account whether there
	  were actually any queue members available to take the call. Now
	  we actually make sure there is at least one member available to
	  take the call if autofill is disabled. (closes issue #13220)
	  Reported by: garychen Review:
	  http://reviewboard.digium.com/r/202/ ........

2009-03-30 14:37 +0000 [r184948]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 184947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
	  14 lines Improve our handling of T38 in the initial INVITE from a
	  device. We now answer with matching media streams to what is
	  requested. If an INVITE is received with both a T38 and RTP media
	  stream this means we answer with both. For any outgoing calls
	  created as a result of this inbound one no T38 is requested in
	  the initial INVITE. Instead if we start receiving udptl packets
	  we trigger a reinvite on the outbound side. (closes issue #12437)
	  Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
	  Review: http://reviewboard.digium.com/r/208/ ........

2009-03-30 13:55 +0000 [r184910]  Russell Bryant <russell@digium.com>

	* channels/h323/Makefile.in: Fix build error when chan_h323 is not
	  being built. (reported by cai1982 in #asterisk-dev)

2009-03-29 05:52 +0000 [r184838-184843]  Russell Bryant <russell@digium.com>

	* /, apps/app_followme.c: Merged revisions 184842 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
	  | 5 lines Ensure targs variable is fully initialized. (closes
	  issue #14758) Reported by: tim_ringenbach ........

	* channels/Makefile: Simplify chan_h323 build to not require a
	  second run of "make". (closes issue #14715) Reported by: jthurman
	  Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman
	  (license 614) Tested by: tzafrir, russell

2009-03-27 20:08 +0000 [r184798-184801]  Leif Madsen <lmadsen@digium.com>

	* apps/app_ices.c: Fix a typo in app_ices. (closes issue #14765)
	  Reported by: timeshell Patches: app_ices.svn-1.6.0.diff uploaded
	  by timeshell (license 399)

	* include/asterisk/doxyref.h: Update commit message guidelines in
	  re: to punctuation. The doxygen documentation has now been
	  updated to state explicitly that I want punctuation atthe end of
	  the first sentence in a commit message. :).

2009-03-27 19:10 +0000 [r184762]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, bridges/bridge_softmix.c,
	  include/asterisk/timing.h, include/asterisk/channel.h,
	  channels/chan_iax2.c, main/timing.c: Improve timing interface to
	  remember which provider provided a timer The ability to
	  load/unload timing interfaces is nice, but it means that when a
	  timer is allocated, it may come from provider A, but later
	  provider B becomes the 'preferred' provider. If this happens, all
	  timer API calls on the timer that was provided by provider A will
	  actually be handed to provider B, which will say WTF and return
	  an error. This patch changes the timer API to include a pointer
	  to the provider of the timer handle so that future operations on
	  the timer will be forwarded to the proper provider. (closes issue
	  #14697) Reported by: moy Review:
	  http://reviewboard.digium.com/r/211/

2009-03-27 18:04 +0000 [r184693-184726]  Russell Bryant <russell@digium.com>

	* main/manager.c, apps/app_minivm.c: Use ast_random() instead of
	  rand() to ensure we use the best RNG available.

	* include/asterisk/app.h, apps/app_dumpchan.c, main/app.c,
	  apps/app_queue.c, apps/app_voicemail.c, main/cli.c: Change
	  global_app_buf to ast_str_thread_global_buf.

2009-03-27 15:57 +0000 [r184639-184677]  Joshua Colp <jcolp@digium.com>

	* bridges/bridge_softmix.c: Fix a potential timer leak in
	  bridge_softmix. It is possible for a bridge to be created without
	  actually being used. In that scenario a timing file descriptor
	  would be opened and not closed. To fix this the timing file
	  descriptor is now closed in the destroy callback, not the thread
	  function.

	* res/res_agi.c: Fix speech structure leak in the AGI speech
	  recognition integration. The AGI dialplan applications did not
	  destroy the speech structure automatically if it was not
	  destroyed by the running AGI script. They will now do this.
	  (issue LUMENVOX-15)

	* bridges/bridge_softmix.c: Remove a cast that is not needed.

2009-03-27 14:00 +0000 [r184630]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h, main/pbx.c, res/ais/evt.c,
	  main/event.c, pbx/pbx_dundi.c, main/asterisk.c: Change g_eid to
	  ast_eid_default.

2009-03-27 13:57 +0000 [r184566-184628]  Joshua Colp <jcolp@digium.com>

	* bridges/bridge_softmix.c: Fix a potential race condition when
	  creating a software based mixing bridge. It was possible for no
	  timer to become available between creating the bridge and
	  starting it. We now open a timer when creating it and keep it
	  open until the bridge is destroyed.

	* /, channels/chan_sip.c: Merged revisions 184565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
	  lines Fix an issue where nat=yes would not always take effect for
	  the RTP session on outgoing calls. If calls were placed using an
	  IP address or hostname the global nat setting was copied over but
	  was not set on the RTP session itself. This caused the RTP stack
	  to not perform symmetric RTP actions. (closes issue #14546)
	  Reported by: acunningham ........

2009-03-27 02:20 +0000 [r184512-184531]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Fix some issues with rwlock corruption
	  that caused deadlock like symptoms. When dvossel and I were doing
	  some load testing last week, we noticed that we could make
	  Asterisk trunk lock up instantly when we started generating a
	  bunch of calls. The backtraces of locked threads were bizarre,
	  and many were stuck on an _unlock_ of an rwlock. The changes are:
	  1) Fix a number of places where a backtrace would be loaded into
	  an invalid index of the backtrace array. It's an off by one
	  error, which ends up writing over the rwlock itself. 2) Ensure
	  that in the array of held locks, we NULL out an index once it is
	  not being used so that it's not confusing when analyzing its
	  contents. 3) Remove a bunch of logging referring to an rwlock
	  operating being done with "deep reentrancy". It is normal for
	  _many_ threads to hold a read lock on an rwlock.

	* main/file.c: Don't act surprised if we get a -1 indication.

	* main/heap.c, include/asterisk/heap.h: Pass more useful
	  information through to lock tracking when DEBUG_THREADS is on.

2009-03-26 22:18 +0000 [r184448]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 184447 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
	  2009) | 3 lines use new, improved 8kHz prompts ........

2009-03-26 21:09 +0000 [r184389]  David Vossel <dvossel@digium.com>

	* /, apps/app_test.c: Merged revisions 184388 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
	  | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
	  8 app_test was failing when sending the last DTMF digit, 8,
	  because of the 100ms pause issued after DTMF is sent. During this
	  pause the other side would hang up causing the test to look like
	  it failed. Now the other side waits a second before hanging up.
	  (closes issue #12442) Reported by: tzafrir ........

2009-03-25 22:11 +0000 [r184339-184344]  Russell Bryant <russell@digium.com>

	* main/event.c: Remove unneeded AST_LIST_ENTRY() and comment on the
	  purpose of ast_event_ref.

	* channels/chan_unistim.c, channels/chan_dahdi.c,
	  include/asterisk/devicestate.h, include/asterisk/event.h,
	  channels/chan_sip.c, apps/app_minivm.c, res/ais/evt.c,
	  main/devicestate.c, main/event.c, include/asterisk/_private.h,
	  include/asterisk/strings.h, channels/chan_iax2.c,
	  main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c:
	  Improve performance of the ast_event cache functionality. This
	  code comes from svn/asterisk/team/russell/event_performance/.
	  Here is a summary of the changes that have been made, in order of
	  both invasiveness and performance impact, from smallest to
	  largest. 1) Asterisk 1.6.1 introduces some additional logic to be
	  able to handle distributed device state. This functionality comes
	  at a cost. One relatively minor change in this patch is that the
	  extra processing required for distributed device state is now
	  completely bypassed if it's not needed. 2) One of the things that
	  I noticed when profiling this code was that a _lot_ of time was
	  spent doing string comparisons. I changed the way strings are
	  represented in an event to include a hash value at the front. So,
	  before doing a string comparison, we do an integer comparison on
	  the hash. 3) Finally, the code that handles the event cache has
	  been re-written. I tried to do this in a such a way that it had
	  minimal impact on the API. I did have to change one API call,
	  though - ast_event_queue_and_cache(). However, the way it works
	  now is nicer, IMO. Each type of event that can be cached (MWI,
	  device state) has its own hash table and rules for hashing and
	  comparing objects. This by far made the biggest impact on
	  performance. For additional details regarding this code and how
	  it was tested, please see the review request. (closes issue
	  #14738) Reported by: russell Review:
	  http://reviewboard.digium.com/r/205/

2009-03-25 19:22 +0000 [r184280]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue with a T38 reinvite being sent
	  even if not configured to do so. If we receive a T38 request
	  negotiate control frame we should only attempt to do so if the
	  option is enabled on the dialog.

2009-03-25 14:38 +0000 [r184220]  Eliel C. Sardanons <eliels@gmail.com>

	* /, main/asterisk.c: Merged revisions 184188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
	  13 lines Avoid destroying the CLI line when moving the cursor
	  backward and trying to autocomplete. When moving the cursor
	  backward and pressing TAB to autocomplete, a NULL is put in the
	  line and we are loosing what we have already wrote after the
	  actual cursor position. (closes issue #14373) Reported by: eliel
	  Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
	  by: lmadsen ........

2009-03-25 14:33 +0000 [r184147-184219]  Russell Bryant <russell@digium.com>

	* main/timing.c: Include poll-compat.h

	* main/timing.c: Change poll() to ast_poll().

	* utils/Makefile, include/asterisk/compat.h: Fix build issues on
	  Mac OSX. (closes issue #14714) Reported by: ygor

2009-03-24 22:40 +0000 [r184079]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_senddtmf.c: Merged revisions 184078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
	  2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
	  The 'digit' variable is guaranteed to be non-NULL, so the if
	  statement could never evaluate true. Changing to ast_strlen_zero
	  makes the logic correct. This was found while reviewing
	  ast_channel_ao2 code review. ........

2009-03-24 22:00 +0000 [r184037-184043]  Russell Bryant <russell@digium.com>

	* main/channel.c: Put siren7 and siren14 in ast_best_codec() just
	  so they're in there somewhere.

	* channels/chan_iax2.c: Exclude slin16, siren7, and siren14 from
	  bandwidth=low and =medium The default codec configuration for
	  chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as
	  the codec in some test calls, but that no longer happens after
	  this change.

2009-03-24 20:01 +0000 [r183995]  David Vossel <dvossel@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: SIP
	  preferred codec only feature Added an option to respond to a SIP
	  invite with only the single most preferred joint codec. This
	  limits the options of what codecs the other side can use. (closes
	  issue #12485) Reported by: bamby Review:
	  http://reviewboard.digium.com/r/206/

2009-03-24 15:26 +0000 [r183865-183914]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 183913 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
	  | 3 lines Additionally note that the operator option needs an 'o'
	  extension. (Related to issue #14731) ........

	* main/http.c: Allow browsers to cache images and other static
	  content.

2009-03-23 22:35 +0000 [r183831]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/portinfo.c,
	  channels/misdn/isdn_lib.c, channels/misdn_config.c: Removed
	  trailing whitespace in chan_misdn files.

2009-03-23 18:58 +0000 [r183766]  Mark Michelson <mmichelson@digium.com>

	* /, res/res_monitor.c: Merged revisions 183700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
	  2009) | 7 lines Fix a memory leak in res_monitor.c The only way
	  that this leak would occur is if Monitor were started using the
	  Manager interface and no File: header were given. Discovered
	  while reviewing the ast_channel_ao2 review request. ........

2009-03-23 18:06 +0000 [r183701]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_dahdi.c: Fixes a documentation error introduced
	  during the CLI cleanup at AstriDevCon 2008. (closes issue #14655)
	  Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic
	  (license 728) Tested by: lmadsen

2009-03-22 21:00 +0000 [r183652]  Joshua Colp <jcolp@digium.com>

	* main/bridging.c: Fix a minor logic flaw with the bridge generic
	  thread. We only want to move the channel pointers that are
	  actually present.

2009-03-20 17:00 +0000 [r183560]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 183559 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20
	  Mar 2009) | 2 lines Fix a crash in IAX2 registration handling
	  found during load testing with dvossel. ........

2009-03-20 16:25 +0000 [r183553-183555]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix chan_sip so it builds.

	* include/asterisk/rtp.h, main/rtp.c, main/asterisk.exports: Remove
	  symbols I just added to main/asterisk.exports and instead rename
	  the functions.

	* main/asterisk.exports: Add some missing symbols to
	  main/asterisk.exports Hey! chan_sip.so loads now!

2009-03-20 12:12 +0000 [r183511]  Eliel C. Sardanons <eliels@gmail.com>

	* channels/chan_dahdi.c: Remove duplicate <description> inside the
	  xml documentation.

2009-03-19 20:30 +0000 [r183436]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
	  Merged revisions 183386 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
	  | 6 lines Cleaning up a few things in detect disconnect patch
	  Initialized ast_call_feature in detect_disconnect to avoid
	  accessing uninitialized memory. Cleaned up /param tags in
	  features.h. No longer send dynamic features in
	  ast_feature_detect. issue #11583 ........

2009-03-19 19:22 +0000 [r183321-183345]  Tilghman Lesher <tlesher@digium.com>

	* /: Recorded merge of revisions 183342 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009)
	  | 2 lines Reordering, to change prior to unlocking ........

	* channels/chan_dahdi.c, /: Merged revisions 183319 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19
	  Mar 2009) | 8 lines Delay signalling progress until a PRI channel
	  really signals progress. (closes issue #13034) Reported by:
	  klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
	  tilghman (license 14) patch_trunk_183progress_klaus3000.txt
	  uploaded by klaus3000 (license 65) Tested by: klaus3000 ........

2009-03-19 18:34 +0000 [r183312]  Jason Parker <jparker@digium.com>

	* /, main/asterisk.exports: Merged revisions 183291 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar
	  2009) | 1 line Export some more required symbols. ........

2009-03-19 18:10 +0000 [r183244]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a memory leak associated with queues. For
	  every attempt that app_queue made to place an outbound call to a
	  queue member, we would allocate a queue_end_bridge structure.
	  When the bridge for the call had completed, we would free the
	  structure. Unfortunately not all call attempts actually end up
	  bridged to a member, so we need to be more selective of when to
	  allocate the structure. With this change, the allocation occurs
	  in an area where we can guarantee that the call will be bridged.
	  (closes issue #14680) Reported by: caspy Patches: 14680.patch
	  uploaded by mmichelson (license 60) Tested by: caspy

2009-03-19 18:00 +0000 [r183239-183242]  Russell Bryant <russell@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/loader.c: Merged revisions 183241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
	  | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
	  like expected. ........

	* /, main/asterisk.exports: Merged revisions 183238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19
	  Mar 2009) | 1 line Allow the AES API to work. ........

2009-03-19 17:00 +0000 [r183196]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.exports: 2 symbols defined when DEBUG_THREADS

2009-03-19 16:28 +0000 [r183172]  David Vossel <dvossel@digium.com>

	* apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
	  Merged revisions 183126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009)
	  | 17 lines Allow disconnect feature before a call is bridged
	  feature.conf has a disconnect option. By default this option is
	  set to '*', but it could be anything. If a user wishes to
	  disconnect a call before the other side answers, only '*' will
	  work, regardless if the disconnect option is set to something
	  else. This is because features are unavailable until bridging
	  takes place. The default disconnect option, '*', was hardcoded in
	  app_dial, which doesn't make any sense from a user perspective
	  since they may expect it to be something different. This patch
	  allows features to be detected from outside of the bridge, but
	  not operated on. In this case, the disconnect feature can be
	  detected before briding and handled outside of features.c.
	  (closes issue #11583) Reported by: sobomax Patches:
	  patch-apps__app_dial.c uploaded by sobomax (license 359)
	  11583.latest-patch uploaded by murf (license 17)
	  detect_disconnect.diff uploaded by dvossel (license 671) Tested
	  by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
	  ........

2009-03-19 16:22 +0000 [r183124-183148]  Russell Bryant <russell@digium.com>

	* /, main/asterisk.exports: Merged revisions 183145 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183145 | russell | 2009-03-19 11:21:56 -0500 (Thu, 19
	  Mar 2009) | 1 line Add missing semicolon in exports script.
	  ........

	* /, main/asterisk.exports: Merged revisions 183123 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r183123 | russell | 2009-03-19 11:13:18 -0500 (Thu, 19
	  Mar 2009) | 2 lines Allow the CallerID API to work again.
	  ........

2009-03-19 16:07 +0000 [r183117]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 183115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
	  2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
	  would erroneously report the device as "in use." A user was
	  having an issue where if an outgoing SIP call was canceled, the
	  SIP device would remain in use if we had not received any
	  response to the initial INVITE we sent out. The SIP device would
	  remain in use until the autocongestion timer was exhausted. I
	  tracked down the cause of this to be the section of code I am
	  removing here. I asked several people what the purpose of this
	  code was meant to be, but no one could give me any sort of answer
	  as to why this was here. The person who was having this issue has
	  been using this patch for several months and it has stopped the
	  problems they have had. AST-196 ........

2009-03-19 15:37 +0000 [r183057-183108]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Improve our triggering of a T38 switchover
	  internally when triggered by a received reinvite. Previously we
	  reached across the channel bridge to get the other party's SIP
	  dialog structure in order to trigger an outgoing reinvite. This
	  is extremely dangerous to do and only works if bridged to another
	  SIP channel. This patch changes this to use the T38 control frame
	  method of requesting a switchover. This change also causes the
	  SIP channel driver to propogate back whether the switchover
	  worked or not instead of blindly accepting the incoming T38
	  reinvite. Review: http://reviewboard.digium.com/r/200/

	* main/channel.c: Fix an issue where a T38 control frame would get
	  dropped. If two channels were bridged together using a generic
	  bridge the T38 control frame would get passed up instead of being
	  indicated on the other channel.

2009-03-18 21:28 +0000 [r183032]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_ael_share.exports (added): allow this module to export
	  everything for now

2009-03-18 21:18 +0000 [r183028]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/ast_h323.cxx: Add some code removed by mistake from
	  commit 182722 that works around a file descriptor leak in
	  versions of PWLib prior to 1.12.0.

2009-03-18 19:41 +0000 [r182960]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.exports: Fixing a lost symbol in manager.c

2009-03-18 11:40 +0000 [r182848-182883]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/callerid.h, channels/chan_dahdi.c, /,
	  main/callerid.c: Merged revisions 182882 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182882 | kpfleming | 2009-03-18 06:31:41 -0500 (Wed, 18 Mar
	  2009) | 3 lines fix another symbol namespace issue (reported by
	  Andrew on asterisk-dev) ........

	* res/res_phoneprov.c, res/res_config_ldap.c, res/res_curl.c,
	  res/res_config_sqlite.c, res/res_jabber.exports, res/res_odbc.c,
	  res/res_odbc.exports: a few more namespace updates...
	  res_ael_share still needs some work before this can be merged to
	  other release branches

2009-03-18 02:28 +0000 [r182847]  Russell Bryant <russell@digium.com>

	* apps/app_nbscat.c, /, main/Makefile,
	  include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
	  include/asterisk/io.h, include/asterisk/channel.h, main/poll.c,
	  main/io.c, main/channel.c, channels/chan_skinny.c, configure,
	  apps/app_mp3.c, res/res_agi.c, channels/chan_alsa.c,
	  include/asterisk/poll-compat.h, main/asterisk.c: Merged revisions
	  182810 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
	  | 44 lines Fix cases where the internal poll() was not being used
	  when it needed to be. We have seen a number of problems caused by
	  poll() not working properly on Mac OSX. If you search around,
	  you'll find a number of references to using select() instead of
	  poll() to work around these issues. In Asterisk, we've had poll.c
	  which implements poll() using select() internally. However, we
	  were still getting reports of problems. vadim investigated a bit
	  and realized that at least on his system, even though we were
	  compiling in poll.o, the system poll() was still being used. So,
	  the primary purpose of this patch is to ensure that we're using
	  the internal poll() when we want it to be used. The changes are:
	  1) Remove logic for when internal poll should be used from the
	  Makefile. Instead, put it in the configure script. The logic in
	  the configure script is the same as it was in the Makefile.
	  Ideally, we would have a functionality test for the problem, but
	  that's not actually possible, since we would have to be able to
	  run an application on the _target_ system to test poll()
	  behavior. 2) Always include poll.o in the build, but it will be
	  empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
	  throughout the source tree to ast_poll(). I feel that it is good
	  practice to give the API call a new name when we are changing its
	  behavior and not using the system version directly in all cases.
	  So, normally, ast_poll() is just redefined to poll(). On systems
	  where AST_POLL_COMPAT is defined, ast_poll() is redefined to
	  ast_internal_poll(). 4) Change poll() in main/poll.c to be
	  ast_internal_poll(). It's worth noting that any code that still
	  uses poll() directly will work fine (if they worked fine before).
	  So, for example, out of tree modules that are using poll() will
	  not stop working or anything. However, for modules to work
	  properly on Mac OSX, ast_poll() needs to be used. (closes issue
	  #13404) Reported by: agalbraith Tested by: russell, vadim
	  http://reviewboard.digium.com/r/198/ ........

2009-03-18 02:21 +0000 [r182826]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_pgsql.c, /, res/res_snmp.c, res/res_smdi.exports
	  (added), main/Makefile, include/asterisk/astobj2.h,
	  res/res_agi.exports (added), Makefile.rules, main/astobj2.c,
	  main/asterisk.exports (added), res/res_odbc.exports (added),
	  res/res_speech.exports (added), res/res_config_odbc.c,
	  res/res_features.exports (added), build_tools/strip_nonapi
	  (removed), res/res_adsi.exports (added), default.exports (added),
	  makeopts.in, res/res_jabber.exports (added),
	  res/res_monitor.exports (added): Merged revisions 182808 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar
	  2009) | 5 lines Improve the build system to *properly* remove
	  unnecessary symbols from the runtime global namespace. Along the
	  way, change the prefixes on some internal-only API calls to use a
	  common prefix. With these changes, for a module to export symbols
	  into the global namespace, it must have *both* the
	  AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
	  the linker to leave the symbols exposed in the module's .so file
	  (see res_odbc.exports for an example). ........

2009-03-17 21:28 +0000 [r182762]  Russell Bryant <russell@digium.com>

	* funcs/func_channel.c, CHANGES: Add support for the "name" option
	  in the CHANNEL() function. Review:
	  http://reviewboard.digium.com/r/199/

2009-03-17 20:47 +0000 [r182722]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
	  configure, autoconf/ast_check_openh323.m4,
	  channels/h323/compat_h323.h, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
	  Plus library to be used in addition to the OpenH323 library
	  Chan_h323 can now be compiled against both the previously
	  supported versions of OpenH323 as well as the current H.323 Plus
	  (version 1.20.2). The configure script has been modified to look
	  in the default install location of h323 to hopefully help avoid
	  using the environment variables OPENH323DIR and PWLIBDIR. Also,
	  the CLI command "h323 show version" has been added which
	  indicates which version of h323 is in use. (closes issue #11261)
	  Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
	  uploaded by jthurman (license 614)

2009-03-17 18:06 +0000 [r182596-182607]  David Vossel <dvossel@digium.com>

	* CHANGES: Fixing CHANGES in rev 182596. Progress DTMF was added
	  into app_dial's D() option. In CHANGES it should have been
	  updated under 1.6.3 rather than 1.6.2.

	* apps/app_dial.c, CHANGES: Option to send DTMF when receiving
	  PROGRESS status The D() option in app_dial is only able to send
	  DTMF after the call has been answered. A progress option has been
	  added to D() to allow DTMF to be sent upon receiving PROGRESS.
	  This allows DTMF to be sent before the call is answered. (closes
	  issue #12123) Reported by: VoipForces Patches:
	  app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419)
	  dtmf_progress.patch uploaded by dvossel (license 671) Tested by:
	  VoipForces, dvossel

2009-03-17 15:22 +0000 [r182553]  Russell Bryant <russell@digium.com>

	* main/channel.c: Tweak the handling of the frame list inside of
	  ast_answer(). This does not change any behavior, but moves the
	  frames from the local frame list back to the channel read queue
	  using an O(n) algorithm instead of O(n^2).

2009-03-17 14:59 +0000 [r182525-182530]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c: correct logic flaw in ast_answer() changes in
	  r182525

	* main/channel.c, main/features.c, include/asterisk/channel.h:
	  Improve behavior of ast_answer() to not lose incoming frames
	  ast_answer(), when supplied a delay before returning to the
	  caller, use ast_safe_sleep() to implement the delay.
	  Unfortunately during this time any incoming frames are discarded,
	  which is problematic for T.38 re-INVITES and other sorts of
	  channel operations. When a delay is not passed to ast_answer(),
	  it still delays for up to 500 milliseconds, waiting for media to
	  arrive. Again, though, it discards any control frames, or
	  non-voice media frames. This patch rectifies this situation, by
	  storing all incoming frames during the delay period on a list,
	  and then requeuing them onto the channel before returning to the
	  caller. http://reviewboard.digium.com/r/196/

2009-03-17 14:24 +0000 [r182521]  Sean Bright <sean@malleable.com>

	* autoconf/ast_ext_lib.m4: Don't include a space before the
	  optional extra text that may follow a help string.

2009-03-17 05:51 +0000 [r182450]  Tilghman Lesher <tlesher@digium.com>

	* /, main/db.c: Merged revisions 182449 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
	  | 7 lines Fix race in astdb The underlying db1 implementation
	  does not fully isolate the pages retrieved from astdb, so the
	  lock protecting accesses needs to be extended until the copy from
	  the shared memory structure is done. (closes issue #14682)
	  Reported by: makoto ........

2009-03-17 01:54 +0000 [r182408]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: OPENR2 uses an incorrect string value if
	  the extension delimiter is not present. * Fixed OPENR2 using an
	  incorrect string value if the extension delimiter is not present
	  in the Dial() function. This was fixed for SS7 and PRI in trunk
	  -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
	  PRI, and others. * Removed trailing whitespace that appeared with
	  OPENR2.

2009-03-16 20:53 +0000 [r182362]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGES for 1.6.3

