Asterisk/astguiclient install from scratch.    v.1.1.12               2006-06-22
   By the astGUIclient group                              astguiclient@eflo.net

**** IMPORTANT - In order for vicidial/astguiclient to function correctly please
read the REQUIREMENTS.txt for a minimum requirements list. ***


This document is meant to be a very in-depth step-by-step explanation of 
installing the Asterisk open-source PBX on a Linux system and also installing 
the astGUIclient suite. The instructions will assume starting from nothing and 
will try to give several side step instructions to account for some differences 
in choices of hardware and software.

The actual installation that I am doing as I write these instructions will be on 
the following hardware:

 - Pentium 3 500MHz
 - Intel motherboard D815BN
 - 256MB PC133 RAM
 - 80GB IBM deskstar 7200RPM Hard Drive
 - Digium Wildcard Single Span T1 Card T100P
 - 2U rackmount case with 250W power supply
 - Phone hardware will be a Grandstream BT102 and a Sipura SPA-2000 because they 
are so cheap and readily available

All of these parts, aside from the Digium card and the two SIP VOIP devices, 
were purchased from ebay and the entire package(with the two VOIP devices and 
all server hardware included) cost me about $1100 to put together including the 
phones and Digium adapter.

We have many other Asterisk servers at our main office, but this one can be 
experimented with easily because it was so cheap to make and has a relatively 
small capacity when compared with a multi-processor server with a quad span T1 
card.

This is our test Asterisk server and functions well for a dozen or so extensions 
in use if it were to be used in production. A size that is optimal for many 
small offices operating with a fractional data/voice T1 for instance.

For hardware you can use almost any Pentium-class processor(PII, PIII, Athlon, 
Xeon, etc), and you can use any digium telco interface card. Both of these 
choices will determine what the capacity of your Asterisk server will be. If you
want to do simple IVR or conference calling and a few extensions, then a PIII 
with a single Digium T1 card will work just fine for you. If you want to use the
VICIDIAL application, you will want to get as high-powered of a machine as you 
can afford and get a digium quad-span T1 card.

The following is assumed for these installation procedures:
- You have access to a CD burner and 3 blank CDs
- You have some sort of broadband internet connection
- You understand basic linux commands and can use a file editor like vi
- You have all of the necessary hardware:
	- a pentium-class computer
	- a digium telco interface card with appropriate telco lines
	- at least 1 SIP VOIP device
	- a Local Area Network(LAN) with extra ports enough for the new server 
and the number of phones you want





PHASE 1: INSTALLING AN OPERATING SYSTEM


This installation will be using Slackware 10.2 for the linux distribution, 
Slackware 10.0will also work with these instructions. There are several easier 
linux distributions and there are others that are more popular, but Slackware is 
a nice non-commercial distro that has been around for a long time and proven 
itself to be a very uncluttered and stable platform for development.

1.	Go to http://www.ultimatebootcd.com/ , download the latest bootcd and 
burn it to a CD. This will be needed to partition the hard drive prior to 
installation of Slackware linux. The latest version as of this writing is 3.3
(If you have problems with your hardware booting some of the utilities with 3.3
I suggest trying 1.7, that version has older utilities, but still gets the job 
done and works on every machine I've tried it on).

2.	Insert the ultimatebootcd you just burned into your CDROM drive and boot 
to it. You will select "filesystem utilities" and then "XFDISK"

3.	Select any old partitions and delete them and then create 2 new 
partitions:
   - 70000 MB, select yes to validate, change partition type to "Linux Native"
   - 3332 MB, select yes to validate, change partition type to "Linux Swap"
   - press F3 to exit and let it do it's thing, this will take an hour or so.

4.	Go to http://www.slackware.com/getslack/ to download Slackware linux. 
The most recent release as of this writing is 10.2. This release fits on 2 CDs. 
Download both installation disks from any close server listed on the download 
page and burn them both to CDs.

5.	Insert Disk 1 of the Slackware installation CD and boot your computer. 
If you have a simple computer with just an IDE drive just hit enter at the boot: 
prompt. If you have other hard drive adapters(SCSI/RAID/SATA/etc..) you will 
need to look at the Slackware installation help page to determine what boot 
image you will need to use to install Slackware correctly.

6.	Login as root and type "setup" at the prompt to go to the setup menu.
7.	Go to ADDSWAP and hit Enter
8.	Select the swap partition you just created and hit Yes, The swap 
partition will then be formatted
9.	Select the root partiton you just created as Linux Native and hit 
Select, then select "ext3" for the file system, then select 4096 for the inodes 
and the root partition will then be formatted
10.	Select "Install Slackware from a CD" and hit OK
11.	Select "auto" installation and hit OK
12.	Select every package except for "KDEI" and hit OK
13.	Select "full" installation and hit OK
14.	Insert the next Slackware installation disk (disk 2) when it prompts 
you, and hit OK to continue
15.	Now you will select the boot kernel that you will use from now on. If 
you have a simple system with IDE drives you can probably just select "skip" and 
go to the next step. If not then you should probably select "cdrom" and select 
the kernel from the list that you selected to boot into the installation.
16.	You can make a bootdisk if you like, but you don't have to.
17.	For Modem you can select "no modem" and hit OK to continue
18.	Enable hotplug, hit Yes to continue
19.	Install lilo "simple" and hit OK to continue
20.	lilo frame buffer console 640x480 is safe choice if you're not sure
21.	Optional Lilo append, leave blank and hit OK to continue
22.	Lilo destination, I usually choose MBR but root works most of the time
23.	Mouse, select the mouse type that you have hooked up, or select ps/2
24.	Load GPM at boot time, Hit Yes to continue
25.	select Yes to configure your network
26.	Hostname, we are typing "phone"
27.	network, we are typing our local domain name
28.	IP address, we are selecting Static IP, here's what we enter for 
network, you should enter a network setup that will work with your local LAN:
	 - IP address: 10.10.10.15
	 - subnet: 255.255.0.0
	 - gateway: 10.10.10.1
	 - name server: 10.10.10.1
29.	Accept your network settings
30.	Startup services to run, change nothing and select OK to continue
31.	select NO for custom screen fonts
32.	Hardware clock to UTC, select NO
33.	Select your time zone and hit OK
34.	I usually select gnome as the window manager, even though you won't be 
using it on this machine
35.	Select Yes to enter a root password. type something that you will 
remember.
36.	Setup of Slackware Linux is complete, hit OK and EXIT then press 
CTRL-ALT-DELETE to reboot your computer





PHASE 2: COMPILING A CUSTOM LINUX KERNEL


From this step on you should be able to continue the installation remotely 
although it is wise to at least have quick access to the machine if something 
goes wrong.

To connect remotely through SSH on linux type "slogin serveripaddress" or to use 
Windows to connect you can use a piece of free software called putty available 
here: http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
Also, for windows you can use SSH file transferring(SFTP) with a program called
filezilla: http://filezilla.sourceforge.net/

This is an optional step if your linux system is running, but compiling your own 
custom kernel is always a way to optimize your system for the hardware you have 
installed or a way to remove the unnecesary modules that are in the default 
kernel. You will definately want to build your own kernel if you have a multi 
processor machine.

If you want to load a 2.6 kernel then start with step -5, otherwise to use the 
2.4 kernel that comes with Slackware(2.4.31) just start at step 1 below:

-5. cd /usr/src
-4. wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.15.6.tar.gz
-3. gunzip linux-2.6.15.6.tar.gz
-2. tar xvf linux-2.6.15.6.tar
-1. mv -f /usr/src/linux /usr/src/linux-old
-0. ln -s /usr/src/linux-2.6.15.6 /usr/src/linux
1.  cd /usr/src/linux			# move to your linux source directory
2.  cp .config config.save		# copy old config to a save file
3.  make mrproper			# prep for kernel assembly
4.  make menuconfig			# launch configuration menu app
    (this part is very dependant upon your own hardware)
	enable processor version	# select the processor that you have
	enable SMP			# if more than 1 processor or Intel HT
	enable high memory ()		# if more than 1GB of RAM
	enable SCSI Multiple		# if SCSI drives
	enable SCSI devices AMI Megaraid	# if SCSI Megaraid adapter
	enable 3com network devices	# if 3com network card
	enable ext3 file system		# for ext3 to work
	enable all ACPI options		# for SMP to work
	enable Enhanced Real Time Clock Support in Character devices section
					# for SMP to work
	enable any other hardware specific options
	exit and save configuration
5.  make dep				# build the kernel dependancies
6.  make clean				# clean up the kernel build areas
7.  make bzImage			# create a kernel bzImage
8.  make modules			# build the modules into the image
9.  make modules_install		# install kernel modules
10. # nothing# mkinitrd /boot/initrd-XXXXXX.img XXXXXX *not needed on Slackware*
11. cp arch/i386/boot/bzImage /boot/bzImage-XXXXXX	# copy image
	(put whatever you want in XXXXXX, that is your new kernel name)
12. cp System.map /boot/System.map-XXXXXXN	# copy system map
13. mv -f /boot/System.map /boot/System.map-orig
14. ln -s /boot/System.map-XXXXXX /boot/System.map	# symlink map
15. vi /etc/lilo.conf			# edit the lilo boot config file
	image=/boot/bzImage-XXXXXX	# add the new image in above-
	label=test-XXXXXX		#    the previous one
	root=/dev/hda1			# device of root partition
	read-only
16. /sbin/lilo				# run the lilo reload script
17. shutdown -r 0			# reboot machine and hope it worked





PHASE 3: INSTALLING SOFTWARE BEFORE ASTERISK

In this step we will be installing software that Asterisk and/or astGUIclient 
needs to be able to function to its fullest ability. Not all of these software 
packages are manditory to successfully install Asterisk and some of them can be 
installed on other machines on your network like MySQL or Apache. But, in this 
installation we are assuming that there are no other machines on our network to 
help the Asterisk server, so it must have everything it needs installed locally.


It's nice to have a non-root account to put applications and have things run on 
when you don't want to use root. We use cron user account for this purpose.



SUBPHASE 3.0: creating an underpriveledged user account

1. type "adduser cron" at the command prompt
2. follow the prompts and assign a password that you will remember to the new 
user. The home directory should be /home/cron/ for this new user.



SUBPHASE 3.1: Installing Perl Modules

NOTE - you can install ActiveState http://www.activestate.com perl for faster 
executing code that uses less processor time, but it is not required. Here's 
the source for ActiveState Perl 5.6: (it's free)
	http://downloads.activestate.com/ActivePerl/src/5.6/AP638_source.tgz
I hope to add the lengthy steps for installing it as your default perl on your
server but I need some time and a free machine to do that.

cpan is the "Comprehensive Perl Archive Network". It's a mirrored archive of 
most of the perl modules out there complete with a installation and management 
command-line interface. Here's what you do to start it:
*REQUIRED* (needed for perl AGIs)

1.  perl -MCPAN -e shell		# type in the command line 
2.  You will then go through CPAN setup, just hit ENTER for most prompts except 
for the mirrors list, you will want to select at least 4 mirrors
	- yes for manual configuration
	- enter for the next 18 prompts
	- for the "make install options" it's a good idea to add UNINST=1 
	- enter for the next 4 prompts
	- select your continent and country
	- select a few cpan mirrors
	- enter for the next 2 prompts
3.  Once you see the cpan> prompt you can begin installing modules
4.  If you've never installed cpan before you should probably install the 
following modules first:   (say YES if asked to install prerequisites)
   - install MD5
   - install Digest::MD5	
   - install Digest::SHA1	
   - install readline		(just hit Enter when it asks for operator)
   - install Bundle::CPAN
   - reload cpan
- then you can install other modules:
   - install DBI
   - install O/OY/OYAMA/Net-MySQL-0.08.tar.gz
   - install Net::Telnet
   - install Time::HiRes
   - install OLE::Storage_Lite		(needed for super list loader Excel)
   - install Spreadsheet::ParseExcel	(needed for super list loader Excel)
- then quit cpan, you are done
5. Go to http://asterisk.gnuinter.net/ and download the asterisk-perl module
(backup link: http://www.freedomphones.net/files/asterisk-perl-0.08.tar.gz)
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz
	- gunzip asterisk-perl-0.08.tar.gz
	- tar xvf asterisk-perl-0.08.tar
	- cd asterisk-perl-0.08
	- perl Makefile.PL
	- make all
	- make install
   - you are done



SUBPHASE 3.2: Installing other utilities

Sox is an audio utility that allows you to mix audio files together at their 
start point into one file. it is necessary for Asterisk recordings that record 
in and out as separate files
*REQUIRED* (needed for recording mixing)

1. Go to http://sourceforge.net/projects/sox/ and download the sox package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://easynews.dl.sourceforge.net/sourceforge/sox/sox-12.17.9.tar.gz
	- gunzip sox-12.17.9.tar.gz
	- tar xvf sox-12.17.9.tar
	- cd sox-12.17.9
	- ./configure
	- make			(if alsa.o errors add --disable-alsa-dsp to configure and redo)
	- make install
   - you are done


LAME is an MP3 encoder used to convert audio files from WAV to MP3. We prefer GSM
usually, but some users have standardized on MP3 so they would need this utility
to be loaded to use that option.
*OPTIONAL*

2. Go to http://lame.sourceforge.net/ and download the lame package
   - to install this directly on the command line type:
	- cd /usr/local
	-wget http://umn.dl.sourceforge.net/sourceforge/lame/lame-3.96.1.tar.gz
	-gunzip lame-3.96.1.tar.gz
	-tar xvf lame-3.96.1.tar
	-cd lame-3.96.1
	-./configure
	-make
	-make install
   - you are done


mpg123 is the mp3 player for asterisk and is used for music-on-hold 
applications. We do not usually use it on our system, but to be more 
comprehensive we will give the directions to install it.
*OPTIONAL* (only if you want to use Astrisk's music-on-hold)

3. Go to http://www.mpg123.de/ and download the mpg123 package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
	- gunzip mpg123-0.59r.tar.gz
	- tar xvf mpg123-0.59r.tar
	- cd mpg123-0.59r
	- ./configure
	- make linux
	- make install
   - you are done


Screen is a terminal emulator that allows you to run a process as command line 
and be able to detach from them('Ctrl+a' then 'd') and log all output of the 
terminal to a screenlog file if desired(add a '-L' to the launching command). 
In our installations this is how we launch Asterisk upon startup and still have
the ability to log output and still attach to the screen that executed asterisk
originally.
*REQUIRED*

4. Go to http://www.gnu.org/software/screen/ and download the screen package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://ftp.gnu.org/gnu/screen/screen-4.0.2.tar.gz
	 or for older version:
	- wget http://mirrors.kernel.org/gnu/screen/screen-3.9.15.tar.gz
	- gunzip screen-4.0.2.tar.gz
	- tar xvf screen-4.0.2.tar
	- cd screen-4.0.2
	- ./configure
	- make
	- make install
   - you are done


ttyload is a simple terminal application that shows the processor load in a 
graphical time-based scrolling graph. We use it to view how loaded the system is 
and it visualizes load spikes very well
*OPTIONAL* (only for obsessive admins like me)

5.  Go to http://www.daveltd.com/src/util/ttyload/ and download the ttyload 
package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.daveltd.com/src/util/ttyload/ttyload-0.4.4.tar.gz
	- gunzip ttyload-0.4.4.tar.gz
	- tar xvf ttyload-0.4.4.tar
	- cd ttyload-0.4.4
	- make
	- ln -s /usr/local/ttyload-0.4.4/ttyload /usr/bin/ttyload
   - you are done


ntpd is the network time protocol daemon that matches the time on your machine 
with the time of a master server somewhere in the world. We use it to make sure 
the time is the same on our client computers and our servers.
*MANDITORY FOR VICIDIAL USERS* (install on server and all clients)

6. Go to http://www.ntp.org/ and download the ntpd package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.eecis.udel.edu/~ntp/ntp_spool/ntp4/ntp-stable-4.2.0a-20060224.tar.gz
		If you get compilation errors here try 4.1.2:
	  - wget http://www.eecis.udel.edu/~ntp/ntp_spool/ntp4/ntp-4.1.2.tar.gz
	- gunzip ntp-stable-4.2.0a-20060224.tar.gz
	- tar xvf ntp-stable-4.2.0a-20060224.tar
	- cd ntp-4.2.0
	- ./configure
	- make
	- make install
	- vi /etc/ntp.conf	(change to just 1 line: "server 18.145.0.30")
	- cp /etc/ntp.conf /etc/ntpd.conf	# just to be sure
	- /usr/local/bin/ntpdate -u 18.145.0.30 # initial sync
	- /usr/sbin/ntpd			# run it
   - you are done


aftpd is a tftp server which is a file transfer method used by many VOIP phones 
to get their firmware and/or configuration files. This is optional.
*OPTIONAL* (only if you want it for a VOIP phone you are going to use)

7. Go to ftp://ftp.mamalinux.com/pub/atftp and download the aftpd package
   - to install this directly on the command line type:
	- cd /usr/local
	- ncftpget ftp://ftp.mamalinux.com/pub/atftp/atftp-0.7.tar.gz
	- gunzip atftp-0.7.tar.gz
	- tar xvf atftp-0.7.tar
	- cd atftp-0.7
	- ./configure
	- make
	- make install
	- mkdir /usr/local/tftp 	# create the directory to put files in
	- chmod 0777 /usr/local/tftp	# set permissions for new directory
	- "/usr/local/sbin/in.tftpd --verbose=6 --daemon --port=69 --user cron /usr/local/tftp"	# command to start tftpd server
   - you are done


iftop is a good console bandwidth visualization tool that shows you active 
connections, where they are going to/from and how much of your precious bandwidth 
they are using. *OPTIONAL*

8. Go to http://www.ex-parrot.com/~pdw/iftop/ and download the package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.tcpdump.org/release/libpcap-0.8.3.tar.gz
	- gunzip libpcap-0.8.3.tar.gz
	- tar xvf libpcap-0.8.3.tar
	- cd libpcap-0.8.3
	- ./configure
	- make
	- make install
	- cd /usr/local
	- wget http://www.ex-parrot.com/~pdw/iftop/download/iftop-0.17.tar.gz
	- gunzip iftop-0.17.tar.gz
	- tar xvf iftop-0.17.tar
	- cd iftop-0.17
	- ./configure
	- make
	- make install
	- iftop
   - you are done


ploticus is a free graph creation package that allows you to create line graphs
within PNG files simply by creating a config file and a data file. We use this 
package along with the included PHP script to generate server performance graphs
that can be displayed real-time on a web page. *OPTIONAL*

9. Go to http://ploticus.sourceforge.net/ and download the package
   - to install this directly on the command line type:
	NOTE: you may have to edit the Makefile to remove X11 if you don't have it
	- cd /usr/local	
	- wget http://ploticus.sourceforge.net/download/pl232src.tar.gz
	- gunzip pl232src.tar.gz
	- tar xvf pl232src.tar
	- cd pl232src/src/
	- make clean
	- make
	- make install
   - you are done
   NOTE: for the graphics to work on the AST_server_performance page you will 
   need the 'pl' script to be linked or copied into your htdocs/vicidial/ploticus directory
   NOTE: you may need to edit the Makefile for ploticus if you do not have X11

MySQL is a fast database system that is very easy to integrate with any 
application. You can either install it on the local Asterisk system or have one 
somewhere on your network. For our purposes, we are creating an Asterisk 
installation that is self contained and needs no other local servers to operate, 
so we will need to install mysql on this machine.
*OPTIONAL* (only install MySQL locally if you don't want to use an installation 
on another machine)
NOTE: a minimum of MySQL 4.0.X is required

10. Go to http://www.mysql.com/ and download the mysql package
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://mirror.trouble-free.net/mysql_mirror/Downloads/MySQL-4.0/mysql-4.0.27.tar.gz
	- gunzip mysql-4.0.27.tar.gz
	- tar xvf mysql-4.0.27.tar
	- cd mysql-4.0.27
	- groupadd mysql
	- useradd -g mysql mysql
	- "./configure --prefix=/usr/local/mysql --enable-large-files --enable-shared=yes"
		If only MySQL client is needed for DBD::mysql then use this:
		- "./configure --prefix=/usr/local/mysql --without-server --enable-shared=yes"
	- make
	- make install
	- scripts/mysql_install_db
	- chown -R root  /usr/local/mysql
	- chown -R mysql /usr/local/mysql/var
	- chgrp -R mysql /usr/local/mysql
	- cp support-files/my-medium.cnf /etc/my.cnf
	- /usr/local/mysql/bin/mysqld_safe --user=mysql &
   - you are done

   ***** NOTE: if you use MySQL version 4.0.X: *****
   (http://mirror.trouble-free.net/mysql_mirror/Downloads/MySQL-4.0/mysql-4.0.27.tar.gz)
   Then you need to make sure that you use the 0.08 version of the Net::MySQL
   perl module. This can be found in the astguiclient release zip file in the 
   libs/Net folder(MySQL.pm). Here are instructions on how to replace the 
   0.09 version of MySQL.pm with the 0.08 version which is MySQL 4.0.X compatible:
	- cpan
	- install O/OY/OYAMA/Net-MySQL-0.08.tar.gz
	    or you can use the version included with the astguiclient release:
	- locate -u
	- locate MySQL.pm (replace the path below with the path you find)
	- mv /usr/lib/perl5/site_perl/5.8.6/Net/MySQL.pm MySQL.pm-old
	- cp /home/cron/astguiclient_1.1.12-3/libs/Net/MySQL.pm /usr/lib/perl5/site_perl/5.8.6/Net/MySQL.pm
   If you do not do this, then you will see MySQL authentication errors on your 
   Asterisk server.

   ***** NOTE: if you use MySQL version 4.1.12 or greater: *****
   (http://mirror.services.wisc.edu/mysql/Downloads/MySQL-4.1/mysql-4.1.18.tar.gz)
   If you are using the 0.08 version of the Net::MySQL perl module with 4.1.X 
   Then you need to make sure that you use the SET PASSWORD ... OLD_PASSWORD to 
   allow the older client libraries to connect to the MySQL server, or you can 
   start your mysqld server with the "-old-passwords" option. If you do not do 
   this, then you will see MySQL authentication errors on your Asterisk server.

   ***** NOTE: if you will be using any of the DBI perl scripts: *****
   Every machine that you will be using the newer BDI perl scripts on will need
   to have the perl modules DBI and DBD::mysql installed on them. To do this
   you will also need to at least have the MySQL client installed on the server
   (see above) then you will need to go to 'cpan' and "install DBI" and 
   "install DBD::mysql". You may need to "force install DBD::mysql" if the DBD
   tests fail on your first try, but that is OK since the tests are not needed


openssh is a remote login protocol server that is always a good idea to have 
updated on your system, so we're going to install a new version now.
*OPTIONAL* (only updated as a precaution, not manditory)
	[NOTE: newer zlib is needed before installing]

11. Go to http://www.openssh.org/ and download the linux source for openssh
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.zlib.net/zlib-1.2.3.tar.gz
	- gunzip zlib-1.2.3.tar.gz
	- tar xvf zlib-1.2.3.tar
	- cd zlib-1.2.3
	- ./configure
	- make
	- make install
	- cd /usr/local
        - wget http://gulus.usherbrooke.ca/pub/appl/ssh/OpenSSH/portable/openssh-4.3p2.tar.gz
	- gunzip openssh-4.3p2.tar.gz
	- tar xvf openssh-4.3p2.tar
	- cd openssh-4.3p2
	- ./configure
	- make
	- make install
   - you are done


openssl is the open-source SSL libraries package, and to install a fake SSL cert 
locally and have it work with apache, you need it installed on your machine
*OPTIONAL* (only install openssl if you want to use SSL secured web pages on 
your locally installed copy of Apache web server)

12. Go to http://www.openssl.org/ and download the linux source for openssl
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://www.openssl.org/source/openssl-0.9.8b.tar.gz
	- gunzip openssl-0.9.8b.tar.gz
	- tar xvf openssl-0.9.8b.tar
	- cd openssl-0.9.8b
	- ./config
	- make
	- make install
   - you are done


apache is a web server that allows you to use many different modules with it to 
extend it's functionality. In order to use some of the astguiclient 
functionalities we need to have Apache and PHP installed on this machine.
*OPTIONAL* (only install Apache and PHP locally if you don't want to use an 
installation on another machine)

13. Go to http://www.apache.org/ and download the apache unix source
    Go to http://www.php.net/ and download the php unix source code
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://apache.secsup.org/dist/httpd/httpd-2.0.58.tar.gz
	- gunzip httpd-2.0.58.tar.gz
	- tar xvf httpd-2.0.58.tar
	- wget http://us2.php.net/distributions/php-4.4.2.tar.gz
		NOTE: PHP 5.1.1 has also been tested with this release
	- gunzip php-4.4.2.tar.gz
	- tar xvf php-4.4.2.tar
	- cd httpd-2.0.58
	- ./configure --enable-so --with-apxs2
	- make
	- make install
	- cd ../php-4.4.2
	- ./configure --with-apxs2=/usr/local/apache2/bin/apxs --with-mysql
	- make
	- make install
	- cp php.ini-dist /usr/local/lib/php.ini
	   NOTE: you will want to make sure notice logging is turned off:
	   error_reporting  =  E_ALL & ~E_NOTICE         ; (this is default)
	- vi /usr/local/apache2/conf/httpd.conf
		add the following lines:
			"AddType application/x-httpd-php .php .phtml"
			"LoadModule php4_module libexec/libphp4.so"
		modify the index.html line and add index.php to the list
	- /usr/local/apache2/bin/apachectl start
   - go to http://your-new-asterisk-server-ipaddress/ to see if it worked
   - you are done


balance is a load-balancing application for Linux that will allow you to spread
the load of your web traffic across many servers. If you are running more than 
70 agents on a single server you may want to install this application and build
another cheap web server to handle the extra load.
*OPTIONAL*

14. Go to http://balance.sourceforge.net to download the most recent source version
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://voxel.dl.sourceforge.net/sourceforge/balance/balance-3.34.tgz
	- gunzip balance-3.34.tgz
	- tar xvf balance-3.34.tar
	- cd balance-3.34
	- make
	- make install
	- /usr/sbin/balance -f 81 localhost:80 10.10.10.16:80
	That command will take port 81 traffic and send it evenly to the local 
	server and the 10.10.10.16 server reducing the load and speeding up 
	the applications. More info on balance: http://www.inlab.de/balance.pdf


subversion is the new code control framework use by the Asterisk community. If 
you want to use the latest development code of Asterisk you will need to have
this loaded on your system.
*OPTIONAL*

15. Go to http://subversion.tigris.org to download the most recent source version
   - to install this directly on the command line type:
	- cd /usr/local
	- wget http://subversion.tigris.org/downloads/subversion-1.3.0.tar.gz
	- gunzip subversion-1.3.0.tar.gz
	- tar xvf subversion-1.3.0.tar
	- cd subversion-1.3.0
	- ./configure --enable-ssl --with-neon
	- make
	- make install



PHASE 4: INSTALLING ASTERISK

OK, all the prep work is done, now it's time to start having fun with Asterisk. 
There are four basic ways to install Asterisk, an official release(at the time of 
this writing the official release is 1.2.9.1), the CVS_v1-0 branch(1.0
with bug fixes not yet packaged and released) and the SVN_DEV version(development
branch). Because there are bugs that affect astGUIclient in releases 1.0.8 and 
1.0.9, we recommend using Asterisk 1.2.9.1 or 1.0.11.1. These instructions are how we 
get our Asterisk system with it's T1 line installed with our 2 SIP VOIP devices
and one IAX2 softphone. 

NOTE: If you want to use release 1.0.8 or 9 we would recommend either using the 
CVS_v1-0_Stable branch where the issues are fixed, or patching your 1.0.8/1.0.9 code
with the following patch: 
(http://astguiclient.sourceforge.net/experimental_code/localmasq.patch)
- If you do patch your system make sure you put the asterisk version 
  field for the server on the admin pages as '1.0.11.1'

1. follow these command line steps:
	- mkdir /usr/src/asterisk
	- cd /usr/src/asterisk
   A. if you want 1.2 release (reliable with new features):
	- wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.9.1.tar.gz
	- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.6.tar.gz
	- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.3.tar.gz
	- gunzip asterisk-1.2.9.1.tar.gz
	- tar xvf asterisk-1.2.9.1.tar
	- gunzip zaptel-1.2.6.tar.gz
	- tar xvf zaptel-1.2.6.tar
	- gunzip libpri-1.2.3.tar.gz
	- tar xvf libpri-1.2.3.tar
   B. If you want latest 1.0 tree stable release:
	- wget http://ftp.digium.com/pub/asterisk/old-releases/asterisk-1.0.11.1.tar.gz
	- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.0.10.tar.gz
	- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.0.9.tar.gz
	- gunzip asterisk-1.0.11.1.tar.gz
	- tar xvf asterisk-1.0.11.1.tar
	- gunzip zaptel-1.0.10.tar.gz
	- tar xvf zaptel-1.0.10.tar
	- gunzip libpri-1.0.9.tar.gz
	- tar xvf libpri-1.0.9.tar
   C. if you want latest SVN_DEV version (sometimes unstable) [1.3 tree]
	- svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
	- svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
	- svn checkout http://svn.digium.com/svn/libpri/trunk libpri
   D. if you want latest STABLE 1.2 tree version (stable)
	- svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
	- svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
	- svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
  - ALL ->
	- (1.0 tree)if you want to allow for more than 100 voicemail messages in a 
          mailbox(warning this will slightly increase memory usage when a call
	  is in voicemail) edit the voicemail source code file:
	   -  vi /usr/src/asterisk/asterisk/apps/app_voicemail.c
		edit this line and change 100 to 999:
		#define MAXMSG 100
	- (1.0 tree)if you have no X server installed on your Asterisk machine, then you 
	  will need to comment out the gtk console lib(only affects stable tree 
	  and 1.0 releases) edit the voicemail source code file:
	   -  vi /usr/src/asterisk/asterisk/pbx/Makefile
		edit this line at the top and just add a hash # in front of it as shown:
		#PBX_LIBS+=$(shell $(CROSS_COMPILE_BIN)gtk-config --cflags >/dev/null 2>/dev/null && echo "pbx_gtkconsole.so")

	- cd ./zaptel-1.2.6
	- make clean
	- make
	- make install
	- cd ../libpri-1.2.3
	- make clean
	- make
	- make install
	- cd ../asterisk-1.2.9.1
		- (1.2 tree) If you want to include Answering Machine Detection ability
		  you will need to download app_amd.c and amd.conf and alter the 
		  apps/Makefile to compile it properly
		- cd apps
		- wget http://www.eflo.net/files/app_amd2.c
		- mv app_amd2.c app_amd.c
		- vi Makefile
		  replace this line(line 32):
		       app_mixmonitor.so app_stack.so
		  with this line:
		       app_mixmonitor.so app_stack.so app_amd.so
		- wget http://www.eflo.net/files/amd2.conf
		- mkdir /etc/asterisk
		- mv amd2.conf /etc/asterisk/amd.conf
		- cd ../

		-(1.2 tree) apply the cli delimiter patch
		- wget http://www.eflo.net/files/cli_chan_concise_delimiter.patch
		- patch -p1 < ./cli_chan_concise_delimiter.patch
			- File to patch: cli.c
	- make clean
	- make
	- make install
	- make samples		# this makes sample conf files
		- (1.2 tree) EXPERIMENTAL!!! If you want to use app_conference
		instead of meetme for VICIDIAL then follow these instructions
		- cd /usr/src/asterisk
		- wget http://www.eflo.net/files/VD_app_conference_0.6.zip
		- unzip VD_app_conference_0.6.zip
		- cd app_conference
		- make clean
		- make
		- make install

	- modprobe zaptel	# this loads the zaptel module
	- install the module for the digium device that you are using, we are 
using the T100P single span T1 card so we use:
	- modprobe wct1xxp
	   Here's the list of all digium cards and the modules you use with 
them:
		Card      Module
		-----------------
		TDM400P   wctdm
		X100P     wcfxo
		TDM*	  wcfxs
		S100U     wcusb
		T100P     wct1xxp
		E100P     wct1xxp
		T400P     tor2
		E400P     tor2
		TE110P    wcte11xp
		TE410P    wct4xxp
		TE405P    wct4xxp
		TE411P    wct4xxp
		TE406P    wct4xxp
		TE210P    wct2xxp
		TE205P    wct2xxp
		TDM2400P  wctdm24xxp
	- If you have chosen a Sangoma T1/E1 or analog card, you will need to 
	  follow their instructions for installation of their driver software
	  LATEST Sangoma Wanpipe drivers: 
	  ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-beta4-2.3.4.tgz
   - now your asterisk installation is built and loaded and it's time to 
configure it.

NOTES: If you want to install zttool diagnostics you may need the newt package installed:
http://www.freedomphones.net/files/newt-0.51.6.tar.gz
ln -s /usr/lib/libnewt.so.0.51.6 /usr/lib/libnewt.so.0.51
then go to your zaptel folder and do 'make zttool' 

Digium/Clone X100P EXAMPLE:
Here is an example of a configuration where an X100P single FXO card is used for
zaptel timing and not used for calling:

NOTE: you can get an X100P through ebay for $10-$30 that will work with Asterisk

	/etc/zaptel.conf:
	loadzone=us
	defaultzone=us
	fxsks=1

	/etc/asterisk/zapata.conf:
	[trunkgroups]
	[channels]
	context=unused
	signalling=fxs_ks
	channel => 1

	Added this to the rc.local file:
	# Load zaptel drivers for x100p
	modprobe zaptel
	modprobe wcfxo



PHASE 5: CONFIGURING ASTERISK AND YOUR SIP PHONES

In this phase we will configure the telco lines, the SIP phones, the extensions, 
meetme(conference calling) rooms, dialplan extensions and the voicemail boxes. 
After this phase your Asterisk system should be able to place and receive calls 
to and from the SIP phones you have installed over the telco lines you've hooked 
up. There are several things that we will not be showing how to do because 
Asterisk is extremely flexible and has so many different ways of being 
configured, that if we were to try to explain them all in this document it would 
be 99% asterisk configuration and be 20,000 lines long, and that would 
just be a barrier for those who just want to get it set up. The "Wiki" and the 
mailing list are two very good resources for finding answers if you run into 
problems configuring your system, here are links to them:
The Wiki:  http://www.voip-info.org/tiki-index.php
The Lists: http://www.asterisk.org/index.php?menu=support

I need to note that it is possible to install Asterisk and use astGUIclient 
applications with no Zaptel(Digium) cards installed, but it is not recommended 
even if you are not going to use Zap trunks for your inbound-outbound calls
with no real Zap devices, you would need to use a dummy timer(zt_dummy) based
on you USB ports to get meetme conference rooms working properly and you may 
have other issues along the way. We would at least recommend getting a X100 or 
X101 board from Digium or a clone manufacturer so there is a dedicated hardware
timer in place on your system.



SUBPHASE 5.0: setting up your Asterisk configuration files

1. edit zaptel.conf
	- vi /etc/zaptel.conf
	   There are many examples inside of the zaptel.conf file that is 
generated with the "make samples" command that we issued at the end 
of the last phase. There are many different parameters for the 
different telco line possibilities, because we are installing a T1 
that is NON-PRI-isdn B8ZS ExtendedSuperframe(ESF) E&M Wink start and 24 
channels, we will use the following settings for zaptel.conf:
	span=1,1,0,esf,b8zs
	e&m=1-24
	loadzone = us
	defaultzone=us
   FOR A PRI YOU WOULD USE SOMETHING LIKE THIS:
	span=2,2,0,esf,b8zs
	bchan=25-47
	dchan=48

2. edit zapata.conf
	- vi /etc/asterisk/zapata.conf
	   There are also many examples of how to configure zapata.conf online. 
we decided to separate our T1 into two line groups to keep some 
incoming calls from being busy if we filled up all of our lines. 
Here's what we used(you can set echocancel=no if you are using PRIs):
	[channels]
	group=1
	language=en
	signalling=em_w
	usecallerid=yes
	callerid=asreceived
	context=default
	echocancel=64
	echocancelwhenbridged=yes
	rxgain=1.0
	txgain=1.0
	channel => 1-2
	group=2
	language=en
	signalling=em_w
	usecallerid=yes
	callerid=asreceived
	context=default
	echocancel=64
	echocancelwhenbridged=yes
	rxgain=1.0
	txgain=1.0
	channel => 3-24
   FOR A PRI YOU WOULD USE SOMETHING LIKE THIS:
	group=3
	language=en
	signalling=pri_net
	usecallerid=yes
	callerid=asreceived
	callprogress=no
	busydetect=no
	context=default
	echocancel=64
	echocancelwhenbridged=yes
	rxgain=1.0
	txgain=1.0
	channel => 25-47

3. edit sip.conf
	- vi /etc/asterisk/sip.conf
	   here is where we will edit the configuration of our SIP compatible 
phone devices. As stated at the beginning, we will be setting up a 
Grandstream Budgetone 102 phone and a Sipura SPA-2000 adapter with 
two analog phones connected(each with it's own extension). Here are 
the settings we used to set each of them up:
	[general]
	port = 5060
	bindaddr = 0.0.0.0
	context = default
	   ; register SIP account on remote machine if using SIP trunks
	   ; register => testSIPtrunk:test@10.10.10.16:5060
	   ;
	   ; setup account for SIP trunking:
	   ; [SIPtrunk]
	   ; disallow=all
	   ; allow=ulaw
	   ; allow=alaw
	   ; type=friend
	   ; username=testSIPtrunk
	   ; secret=test
	   ; host=10.10.10.16
	   ; dtmfmode=inband
	   ; qualify=1000

	[gs102]
	disallow=all
	allow=ulaw
	allow=alaw
	type=friend
	username=gs102
	secret=test
	host=dynamic
	dtmfmode=inband
	defaultip=10.10.10.16
	qualify=1000
	mailbox=102
	[spa2000]
	disallow=all
	allow=ulaw
	allow=alaw
	type=friend
	username=spa2000
	secret=test
	host=dynamic
	dtmfmode=inband
	defaultip=10.10.10.17
	qualify=1000
	mailbox=2000
	[spa2001]
	disallow=all
	allow=ulaw
	allow=alaw
	type=friend
	username=spa2001
	secret=test
	host=dynamic
	dtmfmode=inband
	defaultip=10.10.10.17
	qualify=1000
	mailbox=2001
4.  edit meetme.conf
	- vi /etc/asterisk/meetme.conf
	   This is known as the conference calling configuration file. We are 
just going to add two conferences(one without a pin number and one 
with a pin number required for entry):
	[rooms]
	conf => 8600
	conf => 8601,1234
5.  edit iax.conf (only to do IAX trunks, not required)
	- vi /etc/asterisk/iax.conf
	   This is the IAX configuration file, below is a very simple config for
	   having two Asterisk servers connect natively to each other, if you 
	   will be using this, make sure to add the optional lines included 
	   after the extensions.conf section. Also, there is an account setup 
	   here for a firefly IAX softphone to use.(details on that later)
	   * IMPORTANT NOTE * if you plan to use IAX2 trunks for VICIDIAL 
	   outbound dialing you must register with the remote IAX2 server 
	   through the iax.conf file, not just in the Dial or TRUNK line 
	   of the extensions.conf dialplan.
	[general]
	bindport=4569
	iaxcompat=yes
	bandwidth=high
	allow=all
	allow=gsm                      ; Always allow GSM, it's cool :)
	jitterbuffer=no
	tos=lowdelay
	register => ASTtest1:test@10.10.10.16:4569

	[ASTtest2]
	type=friend
	accountcode=IAXtrunk2
	context=default
	auth=plaintext
	host=dynamic
	permit=0.0.0.0/0.0.0.0
	secret=test
	disallow=all
	allow=ulaw
	qualify=yes

	[firefly01]
	type=friend
	accountcode=firefly01
	context=default
	auth=plaintext
	host=dynamic
	permit=0.0.0.0/0.0.0.0
	secret=test
	disallow=all
	allow=gsm
	qualify=yes

##### EXAMPLE - This is a config example for setting up Binfone service(http://www.binfone.com)

NOTE: The "1112223333" is your iax username. When you signup, Binfone creates
a default IAX username for you, (a 5 digit number, usually, starting with a 1).
This works for most customers.  Folks that want inbound then also sign up for
DIDs, each of which has its own IAX username.  (Which is the 10 digit DID).
Each username has its own password which is managed through their web interface.

      iax.conf:
	[general]
	register => 1112223333:PASSWORD@iax.binfone.com

	[1112223333]
	auth=md5
	type=friend
	username=1112223333
	secret=PASSWORD
	host=iax.binfone.com
	context=incoming-IAX-context-in-extensions.conf


      extensions.conf:
	[global]
	TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com	; IAX trunk interface

	[default]
	exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
	exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN},55,o)
	exten => _91NXXNXXXXXX,3,Hangup

	[incoming]
	exten => 1112223333,1,AGI(call_log.agi,${EXTEN})
	exten => 1112223333,2,Dial(sip/gs102,55,o)
	exten => 1112223333,3,Hangup

##### END EXAMPLE


6.  edit voicemail.conf
	- vi /etc/asterisk/voicemail.conf
	   This is where we set up the voicemail boxes for the extensions that 
we have set up:
	[general]
	format=wav49|gsm|wav
	serveremail=asterisk
	attach=yes
	skipms=3000
	maxsilence=10
	silencethreshold=128
	maxlogins=3
	[zonemessages]
	eastern=America/New_York|'vm-received' Q 'digits/at' IMp
	central=America/Chicago|'vm-received' Q 'digits/at' IMp
	central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' 
M 'hours'
	[default]
	102 => 102,Grandstream Mailbox,root@localhost
	2000 => 2000,Sipura Mailbox 1
	2001 => 2001,Sipura Mailbox 2
	3001 => 3001,Firefly Mailbox 1
7.  edit manager.conf
	- vi /etc/asterisk/manager.conf
	   This is where we set up remote logins to the asterisk manager 
interface, to allow sending of Action commands from remote 
connections to the Asterisk server, this will be important for the 
astguiclient applications so let's set that up now:
	[general]
	enabled = yes
	port = 5038
	bindaddr = 0.0.0.0
	[cron]
	secret = 1234
	read = system,call,log,verbose,command,agent,user
	write = system,call,log,verbose,command,agent,user
	[updatecron]
	secret = 1234
	read = command
	write = command
	[listencron]
	secret = 1234
	read = system,call,log,verbose,command,agent,user
	write = command
	[sendcron]
	secret = 1234
	read = command
	write = system,call,log,verbose,command,agent,user

8.  edit logger.conf
	- vi /etc/asterisk/logger.conf
	   This file determines the messages that are logged to the console and 
the /var/log/asterisk/messages file. We usually turn on full logging 
to the messages file to more easily diagnose any problems that we may 
run into, the problem with this is that is does produce very large 
files, so be warned:
	[logfiles]
	console => notice,warning,error
	messages => notice,warning,error,debug,verbose
9.  edit extensions.conf
	- vi /etc/asterisk/extensions.conf
	   This is known as the dialplan. Since we are installing a 
Long-Distance T1 with one 800 number on it, we will need to put that 
800 number in the plan, as well as how to dial out through the T1 
lines and we will need to add an entry for each of the phones that we 
have just set up in the sip.conf file. There are many examples both 
in the sample file and online for what to put in your dialplan, here 
is the simplified dialplan that we are using:
######------ START extensions.conf example ------######
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
TRUNK=Zap/g1                                    ; Trunk interface
TRUNKX=Zap/g2					; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569	; IAX trunk interface

[default]
; Extension 8600 + 8601 conference rooms
exten => 8600,1,Meetme,8600
exten => 8601,1,Meetme,8601

; Extension 102 - Grandstream hardphone
exten => 102,1,Playback,transfer|skip     ; "Please hold while..."
exten => 102,2,Dial,sip/gs102|20|to       ; Ring, 20 secs max
exten => 102,3,Voicemail,u102             ; Send to voicemail...
; Extension 2000 Sipura line 1
exten => 2000,1,Dial,sip/spa2000|30|to    ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000           ; Send to voicemail...
; Extension 2001 Sipura line 2
exten => 2001,1,Dial,sip/spa2001|30|to    ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001           ; Send to voicemail...
; Extension 2020 rings both sipura lines
exten => 2001,1,Dial,sip/spa2000&sip/spa2001|30|to    ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2000           ; Send to voicemail...
; Extension 301 rings the firefly softphone
exten => 301,1,Dial,(IAX2/firefly01@firefly01/s)
exten => 301,2,Hangup

; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,Answer
exten => _**3429,3,Dial,sip/spa2000&sip/spa2001|30|to    ; Ring, 30 secs max
exten => _**3429,4,Voicemail,u2000           ; Send to voicemail...
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,Answer
exten => _*NXXNXXXXXX*3429,3,Dial,sip/spa2000&sip/spa2001|30|to    ; Ring, 30 
secs max
exten => _*NXXNXXXXXX*3429,4,Voicemail,u2000           ; Send to voicemail...

; dial a long distance outbound number to the UK
exten => _901144XXXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},55,tTo)
exten => _901144XXXXXXXXXX,2,Hangup

; dial a long distance outbound number to Australia
exten => _901161XXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _901161XXXXXXXXX,2,Hangup

; dial an 800 outbound number
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91800NXXXXXX,2,Hangup
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91888NXXXXXX,2,Hangup
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91877NXXXXXX,2,Hangup
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91866NXXXXXX,2,Hangup

; dial a local 727 outbound number with area code
exten => _9727NXXXXXX,1,Dial(${TRUNK}/1${EXTEN:1},,tTo)
exten => _9727NXXXXXX,2,Hangup

; dial a local 813 outbound number with area code
exten => _9813NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _9813NXXXXXX,2,Hangup

; dial a long distance outbound number
exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,2,Hangup

; dial a local outbound number (modified because of only LD T1)
exten => _9NXXXXXX,1,Dial(${TRUNK}/1727${EXTEN:1},,tTo)
exten => _9NXXXXXX,2,Hangup

; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; # timeout invalid rules
exten => #,1,Playback(invalid)              ; "Thanks for trying the demo"
exten => #,2,Hangup                     ; Hang them up.
exten => t,1,Goto(#,1)                  ; If they take too long, give up
exten => i,1,Playback(invalid)          ; "That's not valid, try again"

; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)

; ASTERISK AGENTS LOGINS FOR QUEUES (NOT part of VICIDIAL)
; the following assumes phone agent login and exten are 3 digits and the same
; also assumes that 3-digit login is present in agents.conf and queueus.conf
;Agent Logout then stay onhook, DIAL 54 + 3-digit ID
exten => _54XXX,1,AgentCallbackLogin(||)
; the following are used to login and logout of Asterisk Queues from phone
;Agent Login then stay offhook on the phone, DIAL 55 + 3-digit ID
exten => _55XXX,1,AgentLogin(${EXTEN:1})
;Agent Login then stay onhook, phones will ring, DIAL 56 + 3-digit ID
exten => _56XXX,1,AgentCallbackLogin(||${EXTEN:1}@default)

######------ END extensions.conf example ------######

### OPTIONAL IAX trunk extensions entries for long distance dialing over IAX
	exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
	exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN},55,o)
	exten => _91NXXNXXXXXX,3,Hangup

### OPTIONAL SIP trunk extensions entries for long distance dialing over SIP
	exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
	exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,55,o)
	exten => _91NXXNXXXXXX,3,Hangup

### OPTIONAL IAX Load Balance extens to allow for Overflow and Balanced VDAD
### In this setup, the serverIP is the prefix followed by agent conf_exten
### server 1 extens:
	exten => _010*010*010*016*8600XXX,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)
	exten => _010*010*010*016*8600XXX,2,Hangup
### server 2 extens:
	exten => _010*010*010*015*8600XXX,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
	exten => _010*010*010*015*8600XXX,2,Hangup


10.  Now that you have configured Asterisk, it is time to try to start it for the 
first time.
   - First make sure that your T1 line(or other telco line) is connected to the 
digium card. 
   - next type the following at the command prompt: "ztcfg -vvvvvv"
	- you should see a confirmation that the Zaptel device has loaded
   - now you can launch asteris with the following command:
	"asterisk -vvvvvvvvvvvvgc"
	- you should see a lot of messages scroll by and at the end you should 
be given a CLI> prompt if everything loaded OK.



SUBPHASE 5.1: setting up your SIP phones

You will need to follow the instructions for the phones that you are using with 
your system, but here's the way to set up a Grandstream 102 and a Sipura 
SPA-2000

1.  Here are basic instructions for setting up a Grandstream BT 102:
   - On the phone plug it in to power only at first and follow these 
instructions:
	- wait for the phone to boot up and press the MENU button
	- go through the menu screens with the menu key and the up/down arrow 
keys to move from setting to setting. We are setting the following 
values:
	   - DHCP OFF
	   - IP Addr: 010.010.010.016
	   - Subnet: 255.255.000.000
	   - router: 010.010.010.001
	   - dnS: 010.010.010.001
	   - tftp: we leave this blank for now
	   - menu 7 we change the codec to G-711u
	- now wait 10 seconds and unplug the power and plug it back in
	- you can also plug the network cable into the LAN port on the phone
	- at this time you can go to your workstation and open a new web browser
	- go to http://10.10.10.16/ the password is "admin"
	- here is where you will enter in the configuration details for the 
phone to register with the Asterisk server
	   - SIP server: 10.10.10.15
	   - SIP user ID: gs102
	   - Authenticate ID: gs102
	   - Password: test
	   - Name: gs102
	   - Voice Mail UserID: 102
	   - Send DTMF: in-audio
	   - NTP Server: tick.mit.edu
	- then click update, click review changes, and click reboot
   - your phone should now be able to register with the Asterisk server. If you 
still have your console screen up you should see a registration message 
appear telling you that gs102 has registered.

2. Here are the basic instructions for setting up a Sipura SPA-2000 analog 
adapter with 2 lines.
   - Plug power and two analog phones into the adapter.
   - pick up the phone plugged into line1 and press **** to enter admin menu
   - press 101# then 0# to disable DHCP
   - press 111# then 10*10*10*17# to change the IP address
   - press 121# then 255*255*0*0# to change the subnet mask
   - press 131# then 10*10*10*1# to change the default gateway
   - hang up the phone, unplug the power, plug in the network cable and plug in 
the power cable
   - now you can go the the admin website: http://10.10.10.17/admin/advanced
   - you will need to make these setting changes:
   - click on the "Line 1" tab at the top and change the following values:
	- Proxy: 10.10.10.15
	- Display Name: spa2000
	- userID: spa2000
	- password: test
	- authID: spa2000
	- change the dialplan to the following:
(*xx|xxx|xxxx|xxxxx|xxxxxx|xxxxxxx|xxxxxxxx|xxxxxxxxxxx|xxxxxxxxxxxx|xxxxxxxxxxxxxxx|xxxxxxxxxxxxxxxx.)
	- then click the "submit all changes" button at the bottom of the page 
and your first phone line should work now
   - to register the second line, simply click on the "Line 2" tab and go 
through the above steps except use spa2001 instead of spa2000 for the use IDs

3. Now both of your phone devices are set up and you can try making phone calls 
between the three phones



SUBPHASE 5.2: setting up an IAX2 phone

This is optinal and we won't go into too much detail about this, but currently
there are several IAX hard and softphones on the market and more are coming 
every month. Follow the instructions with the IAX phone you have chosen and 
follow the steps below:

1. Add an entry into your iax.conf file like below if you have not already
	[firefly01]
	type=friend
	accountcode=firefly01
	context=default
	auth=plaintext
	host=dynamic
	permit=0.0.0.0/0.0.0.0
	secret=test
	qualify=yes

2. Add an entry into your extensions.conf file like below if it is not in there
	; Extension 3001 rings IAX phone
	exten => 301,1,Dial(IAX2/firefly01@firefly01/s)
	exten => 301,2,Voicemail,u301           ; Send to voicemail...

3. Download Firefly 3rd party application for windows:
  - http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
  MIRRORS:
  - http://mirror.isp.net.au/ftp/pub/firefly/firefly-thirdparty.exe
  - http://www.freedomphones.net/files/firefly-thirdparty.exe
	- Install the application
	- Launch Firefly Softphone
	- click the "I wish to connect to a 3rd party network" button
	- Enter in network name: Asterisk
	- Select IAX2 as the protocol
	- enter in your server address: "10.10.10.15" in our case
	- enter login and pass: "firefly01" and "test" for in our case
	- click OK and you should be logged in and can place calls


SUBPHASE 5.3: setting up a Zap phone

This is optinal and we won't go into too much detail about this either, there
are a few ways to use Zap devices as phones on your Asterisk system: Zaptel 
phone cards, Channel Banks going through Zaptel T1 card, outside line call 
coming in going through Zaptel line card. There isn't much to do but set your
Zaptel config files up and put entries into your extensions.conf file:

1. Add an entry into your extensions.conf file like below
	; Extension 4001 rings IAX phone
	exten => 4001,1,Dial,Zap/1|30|		; ring Zap device 1
	exten => 4001,2,Voicemail,u4001         ; Send to voicemail...




PHASE 6: INSTALLING ASTGUICLIENT AND VICIDIAL

Now that Asterisk is installed and running we can add the astGUIclient and 
VICIDIAL components to the system. (NOTE: make sure you have added the user "cron")



SUBPHASE 6.0: putting the files in place

1. Go to http://astguiclient.sf.net/ and download the latest astguiclient 
package(as of this writing it is 1.1.12)
   - to install this directly on the command line type:
	- cd /home/cron
	- mkdir astguiclient
	- cd astguiclient
	- wget http://internap.dl.sourceforge.net/sourceforge/astguiclient/astguiclient_1.1.12-3.zip
	- unzip astguiclient_1.1.12-3.zip
	- chmod 0755 install_server_files.pl
	- perl install_server_files.pl
	- cd ../
	- vi AST_SERVER_conf.pl
	   - edit the values of this file(here's what we used):
		$server_ip = '10.10.10.15';
		$DB_server = 'localhost';
		$DB_database = 'asterisk';
		$DB_user = 'cron';
		$DB_pass = '1234';
   - there is one more file you need that's not included with the download 
package, it's the conf.gsm file(this is the half-hour music file that we use 
to put people on hold). I have a free classical music file that is available 
free for download at the following two sites:
	http://www.freedomphones.net/conf.gsm
	http://astguiclient.sf.net/conf.gsm
   Once you have downloaded it, you will need to copy it to this folder:
		/var/lib/asterisk/sounds/
   Then you will need to execute this command to copy it as the park file
   'cp /var/lib/asterisk/sounds/conf.gsm /var/lib/asterisk/sounds/park.gsm'
   - you are done



SUBPHASE 6.1: creating the MySQL "asterisk" database

we will create the database and add a few initial records so that we can 
use the administrative web interface. Since this is a new install it is easier 
to use our new mysql script file to add the tables to the database:

1.  at the command prompt type go to the mysql client: 
/usr/local/mysql/bin/mysql
2.  type the following into the mysql client prompt:
   (make sure you put your IP address in place of "10.10.10.15" in the queries below)
######------ BEGIN Mysql data entry(you can copy and paste this into terminal) #
create database asterisk;

GRANT SELECT,INSERT,UPDATE,DELETE on asterisk.* TO cron@'%' IDENTIFIED BY '1234';
GRANT SELECT,INSERT,UPDATE,DELETE on asterisk.* TO cron@localhost IDENTIFIED BY '1234';

# NOTE: if using MySQL 4.1.12 or higher you may need to run this query too:
UPDATE mysql.user set password=OLD_PASSWORD('1234') where user='cron';

use asterisk;

\. /home/cron/astguiclient/MySQL_AST_CREATE_tables.sql

GRANT SELECT on asterisk.phones TO idcheck@'%' IDENTIFIED BY '1234';
GRANT SELECT on asterisk.phones TO idcheck@localhost IDENTIFIED BY '1234';

insert into servers (server_id,server_description,server_ip,active,asterisk_version)values('TESTasterisk','Test install of Asterisk server', '10.10.10.15','Y','1.2.6');

insert into server_updater values('10.10.10.15','');

insert into phones (extension, dialplan_number, voicemail_id, phone_ip, computer_ip, server_ip, login, pass, status, active, phone_type, fullname, company, picture, messages, old_messages, protocol) values('gs102','102','102','10.10.10.16','10.10.9.16','10.10.10.15','gs102','test', 'ADMIN','Y','Grandstream BT 102','Test Admin Phone','TEST','','0','0','SIP');

quit

   run this command from command line:
	- /home/cron/ADMIN_area_code_populate.pl

######------ END Mysql data entry ------######

NOTE: if you are not installing using default user/pass or have MySQL on another 
server, you will need to edit the dbconnect.php files in the astguiclient, vicidial 
and agc directories of your webroot.

3. Enter the astguiclient administration page:
http://10.10.10.15/astguiclient/admin.php
NOTE: if you click on the Logout button you must leave the user/pass empty and click OK
   - Here you will enter the login and password that you inserted into the mysql 
database in the phones table and gave a status of ADMIN (subphase 6.1 [gs102/test])
   - Now that you are logged into the astGUIclient administration system we can 
add a new phone entry for each of the sipura lines we created.
	- click on the "ADD A NEW PHONE" link at the top and enter in the proper 
information for each of the new phone lines. Here's what we entered for 
spa2000:
	   - Phone extension: spa2000
	   - Dialplan Number: 2000
	   - Voicemail Box: 2000
	   - Phone IP address: 10.10.10.17
	   - Computer IP address: 10.10.9.17
	   - Server IP: 10.10.10.15
	   - Login: spa2000
	   - Password: test
	   - Status: ACTIVE
	   - Active Account: Y	
	   - Phone Type: Sipura SPA-2000 line 1
	   - Full Name: Sipura line 1 test
	   - Company: TEST
	   - Picture:
	- for the next phone simply replace 2000 with 2001 in the above example
   - now your phones are all all set up in the astguiclient system and you can 
use this website to add new phones to be used with astguiclient and monitor 
the number of calls people are making.
	- click on the "ADD A NEW CONFERENCE" link at the top and enter in the 
proper information for each of the new conference lines we will be 
adding in a future step. Here's what we entered for out server:
	   - Phone extension: 
	   - Conference Number: 8600011
	   - Server IP: 	10.10.10.15
	- for the rest of the conference entries simply increment the conference 
number up to 8600029 in the above example
   - now your database is set up for the astguiclient conferences which will 
allow you to have up to 6 remote parties that you called from your GUI client 
application in one conference.
	- click on the "LIST ALL SERVERS" link at the top then click on the
	server to modify. Verify that the GMT time zone and all other fields 
	are what you want them to be. There is a setting(Max VICIDIAL Trunks)
	that can be modified to limit the number of VICIDIAL outbound trunks 
	that will be allowed to use on this server.

4. **OPTIONAL** For IAX clients you will need to use full phone name as the 
extension on the admin page entry: "firefly01@firefly01" for our IAX phone example 
previously. And do not forget to set the protocol on this page to IAX2

5. **OPTIONAL** For Zap clients you will need to use full Zap Channel name as the
extension on the admin page entry: "1-1" for our Zap phone example 
previously. And do not forget to set the protocol on this page to Zap




SUBPHASE 6.2: making additions to your Asterisk conf files

Now that the database is set up and our phones have entries in the system we can 
make the additions to the running Asterisk system that will allow astguiclient 
to work with it.

1. Add the call_log entries to all incoming/outgoing extensions entries:
   - here is how our sample dialplan changes for adding call_log entries(only 
effected extension groups are show):
######------ START extensions.conf changes for call_log ------######

##### This 'h' exten is VERY important for VICIDIAL usage, 
##### you will have problems if it is not in your dialplan!
;exten => h,1,AGI(call_log.agi,${EXTEN}) ; use regular AGI for older versions
exten => h,1,DeadAGI(call_log.agi,${EXTEN}) ; DeadAGI is new
exten => h,2,DeadAGI(VD_hangup.agi,PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(call_log.agi,${EXTEN})
exten => _**3429,3,Answer
exten => _**3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _**3429,5,Voicemail,u2000           
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(call_log.agi,${EXTEN})
exten => _*NXXNXXXXXX*3429,3,Answer
exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000           

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
exten => 7275551212,1,Ringing
exten => 7275551212,2,Wait(1)
exten => 7275551212,3,AGI(call_logCID.agi,${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 7275551212,4,Answer
exten => 7275551212,5,Dial,sip/spa2000&sip/spa2001|30|to
exten => 7275551212,6,Voicemail,u2000

; dial a long distance outbound number to the UK
exten => _901144XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,tTo)
exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
exten => _901161XXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _901161XXXXXXXXX,3,Hangup

; dial an 800 outbound number
exten => _91800NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91800NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91800NXXXXXX,3,Hangup
exten => _91888NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91888NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91888NXXXXXX,3,Hangup
exten => _91877NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91877NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91877NXXXXXX,3,Hangup
exten => _91866NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91866NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _91866NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code
exten => _9727NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,tTo)
exten => _9727NXXXXXX,3,Hangup

; dial a local 813 outbound number with area code
exten => _9813NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _9813NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _9813NXXXXXX,3,Hangup

; dial a long distance outbound number
exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
exten => _9NXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,tTo)
exten => _9NXXXXXX,3,Hangup

######------ END extensions.conf changes ------######


2. Add the call_inbound entries to all incoming extensions entries that you want 
CallerID popups on:
   - here is how our sample dialplan changes for adding call_inbound 
entries(only effected extension groups are show):
######------ START extensions.conf changes for call_inbound ------######
; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):
; 1. the extension of the phone to ring as defined in the asterisk.phones table
; 2. the phone number that was called, for the live_inbound/_log entry
; 3. a text description of the number that was called in
; 4-7. optional fields, they are also passed as fields in the GUI to web browser

; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(call_log.agi,${EXTEN})
exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _**3429,4,Answer
exten => _**3429,5,Dial,sip/spa2000&sip/spa2001|30|to
exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(call_log.agi,${EXTEN})
exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _*NXXNXXXXXX*3429,3,Answer
exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000


; parameters for agi-VDADcloser.agi (2 fields separated by five dashes "-----"):
; 1. the full extension formatted by VICIDIAL for internal transfers * separated
; 2. the word START to denote the beginning of the acceptance of the transfer
; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover or IAX channel]
exten => _90009.,1,Answer                  ; Answer the line
exten => _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-----START)
exten => _90009.,3,Hangup

; parameters for agi-VDADcloser_inbound....agi (7 fields separated by five dashes "-----"):
; 1. the full name of the IN GROUP to be used in vicidial for the inbound call
; 2. the phone number that was called, for the log entry
; 3. the callerID or lead_id of the person that called(usually overridden)
; 4. the park extension audio file name if used
; 5. the status of the call initially(usually not used)
; 6. the list_id to insert the new lead under if it is new (and callerID available)
; 7. the phone dialing code to insert with the new lead if new (and callerID available)

; inbound VICIDIAL call with CID delivery through T1 PRI
exten => 1234,1,Ringing                 ; call ringing
exten => 1234,2,Wait(1)                 ; Wait 1 second for CID delivery from PRI
exten => 1234,3,Answer                  ; Answer the line
exten => 1234,4,AGI(agi-VDADcloser_inboundCID.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,5,Hangup

; inbound VICIDIAL call with ANI delivery through robbed-bit T1 (*NXXNXXXXXX*DNIS)
exten => 1234,1,Answer                  ; Answer the line
exten => 1234,2,AGI(agi-VDADcloser_inboundANI.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

; inbound VICIDIAL call with prompt for 4-digit fronter code
exten => 1234,1,Answer                  ; Answer the line
exten => 1234,2,AGI(agi-VDADcloser_inbound.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

### follow these instructions if you plan to have VICIDIAL agents take inbound or closer calls:
	1. create a new campaign in VICIDIAL  called "CLOSER"
	2. in VICIDIAL web admin "add a new in-group" (the above examples would be "CL_GALLERIA")
		- group IDs cannot contain spaces ' ' or dashes '-' or plusses '+'
		- if you are using a HEX color value make sure to include the hash '#' at the beginning
	3. have agents log in to the CLOSER campaign and select the CL_GALLERIA button
	4. they will now start receiving inbound calls
	5. as calls come in, each call is inserted into the vicidial_list table under the 
	   list specified int the AGI string, In the above example that would be list 999
	6. if you want to take closer calls from the campaign "TEST" you will need to create
	   an in-group called "CL_TEST_" for internal closing(on the same system) or "CL_TEST_L"
	   for local closing(closer on different system from fronter) and then the fronter will
	   click on the "internal closer" button to send the call to a closer

* NOTE, you need to set the dial_level of the CLOSER campaign to 1 or higher for inbound/closers to work

######------ END extensions.conf changes for call_inbound ------######


3. Add the ZapBarge entries for all zap lines:
   - here is how our sample dialplan changes for adding zapbarge line-specific 
entries(this is a pure addition, nothing is being modified):
######------ START extensions.conf additions for direct zapbarges ------######
; ZapBarge direct channel extensions
exten => 8612001,1,ZapBarge(1)
exten => 8612002,1,ZapBarge(2)
exten => 8612003,1,ZapBarge(3)
exten => 8612004,1,ZapBarge(4)
exten => 8612005,1,ZapBarge(5)
exten => 8612006,1,ZapBarge(6)
exten => 8612007,1,ZapBarge(7)
exten => 8612008,1,ZapBarge(8)
exten => 8612009,1,ZapBarge(9)
exten => 8612010,1,ZapBarge(10)
exten => 8612011,1,ZapBarge(11)
exten => 8612012,1,ZapBarge(12)
exten => 8612013,1,ZapBarge(13)
exten => 8612014,1,ZapBarge(14)
exten => 8612015,1,ZapBarge(15)
exten => 8612016,1,ZapBarge(16)
exten => 8612017,1,ZapBarge(17)
exten => 8612018,1,ZapBarge(18)
exten => 8612019,1,ZapBarge(19)
exten => 8612020,1,ZapBarge(20)
exten => 8612021,1,ZapBarge(21)
exten => 8612022,1,ZapBarge(22)
exten => 8612023,1,ZapBarge(23)
exten => 8612024,1,ZapBarge(24)
######------ END extensions.conf additions for direct zapbarges ------######


4. Add the meetme entries for astguiclient and VICIDIAL conferences to 
meetme.conf:
   - here is how our sample meetme.conf file changes for adding conference 
entries (this is a pure addition, nothing is being modified):
######------ START meetme.conf additions for conferences ------######
conf => 8600011
conf => 8600012
conf => 8600013
conf => 8600014
conf => 8600015
conf => 8600016
conf => 8600017
conf => 8600018
conf => 8600019
conf => 8600020
conf => 8600021
conf => 8600022
conf => 8600023
conf => 8600024
conf => 8600025
conf => 8600026
conf => 8600027
conf => 8600028
conf => 8600029
conf => 8600051
conf => 8600052
conf => 8600053
conf => 8600054
conf => 8600055
conf => 8600056
conf => 8600057
conf => 8600058
conf => 8600059
conf => 8600060
conf => 8600061
conf => 8600062
conf => 8600063
conf => 8600064
conf => 8600065
conf => 8600066
conf => 8600067
conf => 8600068
conf => 8600069
conf => 8600070
conf => 8600071
conf => 8600072
conf => 8600073
conf => 8600074
conf => 8600075
conf => 8600076
conf => 8600077
conf => 8600078
conf => 8600079
conf => 8600080
conf => 8600081
conf => 8600082
conf => 8600083
conf => 8600084
conf => 8600085
conf => 8600086
conf => 8600087
conf => 8600088
conf => 8600089
conf => 8600090
conf => 8600091
conf => 8600092
conf => 8600093
conf => 8600094
conf => 8600095
conf => 8600096
conf => 8600097
conf => 8600098
conf => 8600099
conf => 8600100
######------ END meetme.conf additions for conferences ------######


5. Add the conference entries for astguiclient conferences:
   - here is how our sample dialplan changes for adding conference entries (this 
is a pure addition, nothing is being modified):
######------ START extensions.conf additions for agc conferences ------######
exten => 8600011,1,Meetme,8600011|q
exten => 8600012,1,Meetme,8600012|q
exten => 8600013,1,Meetme,8600013|q
exten => 8600014,1,Meetme,8600014|q
exten => 8600015,1,Meetme,8600015|q
exten => 8600016,1,Meetme,8600016|q
exten => 8600017,1,Meetme,8600017|q
exten => 8600018,1,Meetme,8600018|q
exten => 8600019,1,Meetme,8600019|q
exten => 8600020,1,Meetme,8600020|q
exten => 8600021,1,Meetme,8600021|q
exten => 8600022,1,Meetme,8600022|q
exten => 8600023,1,Meetme,8600023|q
exten => 8600024,1,Meetme,8600024|q
exten => 8600025,1,Meetme,8600025|q
exten => 8600026,1,Meetme,8600026|q
exten => 8600027,1,Meetme,8600027|q
exten => 8600028,1,Meetme,8600028|q
exten => 8600029,1,Meetme,8600029|q
######------ END extensions.conf additions for agc conferences ------######


6. Add the conference entries for VICIDIAL conferences:
   - here is how our sample dialplan changes for adding VICIDIAL conference 
entries(this is a pure addition, nothing is being modified):
NOTE: see below these entries for app_conference instructions is used
######------ START extensions.conf changes for VD conf ------######
exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
exten => 8600053,1,Meetme,8600053
exten => 8600054,1,Meetme,8600054
exten => 8600055,1,Meetme,8600055
exten => 8600056,1,Meetme,8600056
exten => 8600057,1,Meetme,8600057
exten => 8600058,1,Meetme,8600058
exten => 8600059,1,Meetme,8600059
exten => 8600060,1,Meetme,8600060
exten => 8600061,1,Meetme,8600061
exten => 8600062,1,Meetme,8600062
exten => 8600063,1,Meetme,8600063
exten => 8600064,1,Meetme,8600064
exten => 8600065,1,Meetme,8600065
exten => 8600066,1,Meetme,8600066
exten => 8600067,1,Meetme,8600067
exten => 8600068,1,Meetme,8600068
exten => 8600069,1,Meetme,8600069
exten => 8600070,1,Meetme,8600070
exten => 8600071,1,Meetme,8600071
exten => 8600072,1,Meetme,8600072
exten => 8600073,1,Meetme,8600073
exten => 8600074,1,Meetme,8600074
exten => 8600075,1,Meetme,8600075
exten => 8600076,1,Meetme,8600076
exten => 8600077,1,Meetme,8600077
exten => 8600078,1,Meetme,8600078
exten => 8600079,1,Meetme,8600079
exten => 8600080,1,Meetme,8600080
exten => 8600081,1,Meetme,8600081
exten => 8600082,1,Meetme,8600082
exten => 8600083,1,Meetme,8600083
exten => 8600084,1,Meetme,8600084
exten => 8600085,1,Meetme,8600085
exten => 8600086,1,Meetme,8600086
exten => 8600087,1,Meetme,8600087
exten => 8600088,1,Meetme,8600088
exten => 8600089,1,Meetme,8600089
exten => 8600090,1,Meetme,8600090
exten => 8600091,1,Meetme,8600091
exten => 8600092,1,Meetme,8600092
exten => 8600093,1,Meetme,8600093
exten => 8600094,1,Meetme,8600094
exten => 8600095,1,Meetme,8600095
exten => 8600096,1,Meetme,8600096
exten => 8600097,1,Meetme,8600097
exten => 8600098,1,Meetme,8600098
exten => 8600099,1,Meetme,8600099
exten => 8600100,1,Meetme,8600100
; quiet entry and leaving conferences for VICIDIAL
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
exten => 78600053,1,Meetme,8600053|q
exten => 78600054,1,Meetme,8600054|q
exten => 78600055,1,Meetme,8600055|q
exten => 78600056,1,Meetme,8600056|q
exten => 78600057,1,Meetme,8600057|q
exten => 78600058,1,Meetme,8600058|q
exten => 78600059,1,Meetme,8600059|q
exten => 78600060,1,Meetme,8600060|q
exten => 78600061,1,Meetme,8600061|q
exten => 78600062,1,Meetme,8600062|q
exten => 78600063,1,Meetme,8600063|q
exten => 78600064,1,Meetme,8600064|q
exten => 78600065,1,Meetme,8600065|q
exten => 78600066,1,Meetme,8600066|q
exten => 78600067,1,Meetme,8600067|q
exten => 78600068,1,Meetme,8600068|q
exten => 78600069,1,Meetme,8600069|q
exten => 78600070,1,Meetme,8600070|q
exten => 78600071,1,Meetme,8600071|q
exten => 78600072,1,Meetme,8600072|q
exten => 78600073,1,Meetme,8600073|q
exten => 78600074,1,Meetme,8600074|q
exten => 78600075,1,Meetme,8600075|q
exten => 78600076,1,Meetme,8600076|q
exten => 78600077,1,Meetme,8600077|q
exten => 78600078,1,Meetme,8600078|q
exten => 78600079,1,Meetme,8600079|q
exten => 78600080,1,Meetme,8600080|q
exten => 78600081,1,Meetme,8600081|q
exten => 78600082,1,Meetme,8600082|q
exten => 78600083,1,Meetme,8600083|q
exten => 78600084,1,Meetme,8600084|q
exten => 78600085,1,Meetme,8600085|q
exten => 78600086,1,Meetme,8600086|q
exten => 78600087,1,Meetme,8600087|q
exten => 78600088,1,Meetme,8600088|q
exten => 78600089,1,Meetme,8600089|q
exten => 78600090,1,Meetme,8600090|q
exten => 78600091,1,Meetme,8600091|q
exten => 78600092,1,Meetme,8600092|q
exten => 78600093,1,Meetme,8600093|q
exten => 78600094,1,Meetme,8600094|q
exten => 78600095,1,Meetme,8600095|q
exten => 78600096,1,Meetme,8600096|q
exten => 78600097,1,Meetme,8600097|q
exten => 78600098,1,Meetme,8600098|q
exten => 78600099,1,Meetme,8600099|q
exten => 78600100,1,Meetme,8600100|q
; quiet monitor extensions for meetme rooms (for room managers)
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq
exten => 68600053,1,Meetme,8600053|mq
exten => 68600054,1,Meetme,8600054|mq
exten => 68600055,1,Meetme,8600055|mq
exten => 68600056,1,Meetme,8600056|mq
exten => 68600057,1,Meetme,8600057|mq
exten => 68600058,1,Meetme,8600058|mq
exten => 68600059,1,Meetme,8600059|mq
exten => 68600060,1,Meetme,8600060|mq
exten => 68600061,1,Meetme,8600061|mq
exten => 68600062,1,Meetme,8600062|mq
exten => 68600063,1,Meetme,8600063|mq
exten => 68600064,1,Meetme,8600064|mq
exten => 68600065,1,Meetme,8600065|mq
exten => 68600066,1,Meetme,8600066|mq
exten => 68600067,1,Meetme,8600067|mq
exten => 68600068,1,Meetme,8600068|mq
exten => 68600069,1,Meetme,8600069|mq
exten => 68600070,1,Meetme,8600070|mq
exten => 68600071,1,Meetme,8600071|mq
exten => 68600072,1,Meetme,8600072|mq
exten => 68600073,1,Meetme,8600073|mq
exten => 68600074,1,Meetme,8600074|mq
exten => 68600075,1,Meetme,8600075|mq
exten => 68600076,1,Meetme,8600076|mq
exten => 68600077,1,Meetme,8600077|mq
exten => 68600078,1,Meetme,8600078|mq
exten => 68600079,1,Meetme,8600079|mq
exten => 68600080,1,Meetme,8600080|mq
exten => 68600081,1,Meetme,8600081|mq
exten => 68600082,1,Meetme,8600082|mq
exten => 68600083,1,Meetme,8600083|mq
exten => 68600084,1,Meetme,8600084|mq
exten => 68600085,1,Meetme,8600085|mq
exten => 68600086,1,Meetme,8600086|mq
exten => 68600087,1,Meetme,8600087|mq
exten => 68600088,1,Meetme,8600088|mq
exten => 68600089,1,Meetme,8600089|mq
exten => 68600090,1,Meetme,8600090|mq
exten => 68600091,1,Meetme,8600091|mq
exten => 68600092,1,Meetme,8600092|mq
exten => 68600093,1,Meetme,8600093|mq
exten => 68600094,1,Meetme,8600094|mq
exten => 68600095,1,Meetme,8600095|mq
exten => 68600096,1,Meetme,8600096|mq
exten => 68600097,1,Meetme,8600097|mq
exten => 68600098,1,Meetme,8600098|mq
exten => 68600099,1,Meetme,8600099|mq
exten => 68600100,1,Meetme,8600100|mq
######------ END extensions.conf changes for VD conf ------######

NOTE: If you want to use app_conference instead of meetme and you already have 
it installed, you will need to do the following:

In your extensions.conf file you would replace these lines:
exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
...continue through 8600100...
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
...continue through 78600100...
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq
...continue through 68600100...

with these lines:
exten => 8600051,1,Conference(8600051)
exten => 8600052,1,Conference(8600052)
...continue through 8600100...
exten => 78600051,1,Conference(8600051|q)
exten => 78600052,1,Conference(8600052|q)
...continue through 78600100...
exten => 68600051,1,Conference(8600051|Lq)
exten => 68600052,1,Conference(8600052|Lq)
...continue through 68600100...

NOTE: If you want to do DTMF passthru with app_conference bee sure to add the
"i" and "t" flags to the 8600XX lines: Conference(8600051|it)


7. Add the more entries for astGUIclient specific uses:
   - here are some more dialplan additions needed to use astGUIclient(this is a 
pure addition, nothing is being modified):
######------ START extensions.conf other additions ------######
; park channel for client GUI parking, hangup after 30 minutes
;    create a GSM formatted audio file named "park.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup 
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup 

; park channel for client GUI conferencing, hangup after 30 minutes
;    create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
;    create a GSM formatted audio file complies with safe harbor rules
;    and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERIDNAME})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
;    replace conf with the message file you want to leave
exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

; this is used to allow the GUI to send you directly into voicemail
;     don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail
;     don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(2)
exten => _85026666666666.,2,Voicemail(${EXTEN:14})
exten => _85026666666666.,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
;    sends the digits to be played in the callerID field
;    sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; prompt recording AGI script, ID is 4321
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi)
exten => 8168,3,Hangup

; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,AGI(call_log.agi,${EXTEN})
exten => 8365,2,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,AGI(call_log.agi,${EXTEN})
exten => 8366,2,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,AGI(call_log.agi,${EXTEN})
exten => 8367,2,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,3,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,4,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,AGI(call_log.agi,${EXTEN})
exten => 8368,2,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,3,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,AGI(call_log.agi,${EXTEN})
exten => 8369,2,AMD(3500|1500|300|5000|120|50|5|256)
exten => 8369,3,AGI(VD_amd.agi,${EXTEN})
exten => 8369,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,6,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,AGI(call_log.agi,${EXTEN})
exten => 8372,2,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

### OPTIONAL IAX Load Balance extens to allow for Overflow and Balanced VDAD
### In this setup, the serverIP is the prefix followed by agent conf_exten
### If you have more than 1 server doing VICIDIAL outbound, these lines are
### REQUIRED for Load Balancing to work properly
### server 1 extens:
	exten => _010*010*010*016*8600XXX,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)
	exten => _010*010*010*016*8600XXX,2,Hangup
### server 2 extens:
	exten => _010*010*010*015*8600XXX,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
	exten => _010*010*010*015*8600XXX,2,Hangup


SUBPHASE 6.3: adding entries to your MySQL "asterisk" database for vicidial 
applications

We need to add a few initial values to the vicidial tables in the "asterisk" 
database in order to start setting up the vicidial dialer system for use.

1.  at the command prompt type this to go to the mysql client: 
/usr/local/mysql/bin/mysql
2.  type the following into the mysql client prompt:
   (make sure you put your IP address in place of "10.10.10.15" in the queries below)
######------ BEGIN Mysql data entry(you can copy and paste this into terminal) #

insert into vicidial_users (user,pass,full_name,user_level,user_group,load_leads,campaign_detail,ast_admin_access) values('6666','1234','Admin','9','ADMIN','1','1','1');

insert into vicidial_conferences values('8600051','10.10.10.15','');
insert into vicidial_conferences values('8600052','10.10.10.15','');
insert into vicidial_conferences values('8600053','10.10.10.15','');
insert into vicidial_conferences values('8600054','10.10.10.15','');
insert into vicidial_conferences values('8600055','10.10.10.15','');
insert into vicidial_conferences values('8600056','10.10.10.15','');
insert into vicidial_conferences values('8600057','10.10.10.15','');
insert into vicidial_conferences values('8600058','10.10.10.15','');
insert into vicidial_conferences values('8600059','10.10.10.15','');
insert into vicidial_conferences values('8600060','10.10.10.15','');
insert into vicidial_conferences values('8600061','10.10.10.15','');
insert into vicidial_conferences values('8600062','10.10.10.15','');
insert into vicidial_conferences values('8600063','10.10.10.15','');
insert into vicidial_conferences values('8600064','10.10.10.15','');
insert into vicidial_conferences values('8600065','10.10.10.15','');
insert into vicidial_conferences values('8600066','10.10.10.15','');
insert into vicidial_conferences values('8600067','10.10.10.15','');
insert into vicidial_conferences values('8600068','10.10.10.15','');
insert into vicidial_conferences values('8600069','10.10.10.15','');
insert into vicidial_conferences values('8600070','10.10.10.15','');
insert into vicidial_conferences values('8600071','10.10.10.15','');
insert into vicidial_conferences values('8600072','10.10.10.15','');
insert into vicidial_conferences values('8600073','10.10.10.15','');
insert into vicidial_conferences values('8600074','10.10.10.15','');
insert into vicidial_conferences values('8600075','10.10.10.15','');
insert into vicidial_conferences values('8600076','10.10.10.15','');
insert into vicidial_conferences values('8600077','10.10.10.15','');
insert into vicidial_conferences values('8600078','10.10.10.15','');
insert into vicidial_conferences values('8600079','10.10.10.15','');
insert into vicidial_conferences values('8600080','10.10.10.15','');
insert into vicidial_conferences values('8600081','10.10.10.15','');
insert into vicidial_conferences values('8600082','10.10.10.15','');
insert into vicidial_conferences values('8600083','10.10.10.15','');
insert into vicidial_conferences values('8600084','10.10.10.15','');
insert into vicidial_conferences values('8600085','10.10.10.15','');
insert into vicidial_conferences values('8600086','10.10.10.15','');
insert into vicidial_conferences values('8600087','10.10.10.15','');
insert into vicidial_conferences values('8600088','10.10.10.15','');
insert into vicidial_conferences values('8600089','10.10.10.15','');
insert into vicidial_conferences values('8600090','10.10.10.15','');
insert into vicidial_conferences values('8600091','10.10.10.15','');
insert into vicidial_conferences values('8600092','10.10.10.15','');
insert into vicidial_conferences values('8600093','10.10.10.15','');
insert into vicidial_conferences values('8600094','10.10.10.15','');
insert into vicidial_conferences values('8600095','10.10.10.15','');
insert into vicidial_conferences values('8600096','10.10.10.15','');
insert into vicidial_conferences values('8600097','10.10.10.15','');
insert into vicidial_conferences values('8600098','10.10.10.15','');
insert into vicidial_conferences values('8600099','10.10.10.15','');
insert into vicidial_conferences values('8600100','10.10.10.15','');

######------ END Mysql data entry ------######



SUBPHASE 6.4: setting up asterisk and helper applications for startup

1. Make several entries in the rc.local of your system:
   - on the command line type:
	- vi /etc/rc.d/rc.local
	   - add the following entries(here's what we used):
	      
### startup tftp server (if you have them installed)
/usr/local/sbin/in.tftpd --verbose=6 --daemon --port=69 --user cron /usr/local/tftp

### add the CVSROOT variable definition
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot

### start time server
/usr/local/bin/ntpdate -u 18.145.0.30
/usr/sbin/ntpd

### start up the MySQL server
/usr/local/mysql/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache &

### start up the MySQL 4.1.X server (with old passwords)
/usr/local/mysql/bin/safe_mysqld --old-passwords --skip-name-resolve --skip-host-cache &

### start up the apache web server
/usr/local/apache2/bin/apachectl start

### load digium tormenta 4xT1 drivers into system
modprobe zaptel
modprobe wct1xxp
/sbin/ztcfg -vvvvvvvvvvvv

### sybsys local login
touch /var/lock/subsys/local

### sleep for 20 seconds before launching Asterisk
sleep 20

### start up asterisk
/home/cron/start_asterisk_boot.pl

   - you are done




SUBPHASE 6.5: setting up astguiclient scripts for continuous running

1. Make several entries in the cron of your system:
   - on the command line type:
	- cd /home/cron
	- crontab -e
	   - add the following entries(here's what we used):
	      (Note if you have not edited the AST_CRON_mix_recordings.pl script 
to FTP to a different server, then you should use the 
AST_CRON_mix_recordings_BASIC.pl file in it's place)

### recording mixing for Asterisk run every 5 minutes
1,6,11,16,21,26,31,36,41,46,51,56 * * * 1,2,3,4,5,6 /home/cron/AST_CRON_mix_recordings_BASIC.pl

### keepalive script for Asterisk updaters
* * * * * /home/cron/ADMIN_keepalive_AST_update.pl

### keepalive script for Asterisk manager queue system
* * * * * /home/cron/ADMIN_keepalive_AST_send_listen.pl

### keepalive script for Asterisk VICIDIAL autodial system
* * * * * /home/cron/ADMIN_keepalive_AST_VDautodial.pl

### keepalive script for VICIDIAL remote agents
* * * * * /home/cron/ADMIN_keepalive_AST_VDremote_agents.pl

### kill Hangup script for Asterisk updaters
* * * * * /home/cron/AST_manager_kill_hung_congested.pl

### updater for voicemail
* * * * * /home/cron/AST_vm_update.pl

### updater for conference validator
* * * * * /home/cron/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour
11 * * * * /home/cron/AST_flush_DBqueue.pl -q

### fix the vicidial_agent_log once every hour
33 * * * * /home/cron/AST_cleanup_agent_log.pl

### updater for VICIDIAL hopper
* * * * * /home/cron/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table
1 1 * * * /home/cron/ADMIN_adjust_GMTnow_on_leads.pl --debug

### reset several temporary-info tables in the database
2 1 * * * /home/cron/AST_reset_mysql_vars.pl

### optimize the database tables within the asterisk database
3 1 * * * /home/cron/AST_DB_optimize.pl

## adjust time on the server with ntp
30 * * * * /usr/local/bin/ntpdate -u 18.145.0.30 2>/dev/null 1>&2

### VICIDIAL agent time log weekly summary report generation
2 0 * * 0 /home/cron/AST_agent_week.pl

   - once your system starts up you can attach to the screen running asterisk by 
typing "screen -r <screen>" find which screen by typing "screen -r" and 
looking for the lowest screen number. Then to detach again from the screen 
while you are in it type 'Ctrl+a' then 'd'
   - you are done




SUBPHASE 6.6: adding test leads to the VICIDIAL database and configuring a 
VICIDIAL campaign and users

First we will add a few test leads to the vicidial_list table so that we can 
test our system. There is also an application included with the distribution 
that will accept a delimited file of leads placed in the /home/cron/VICIDIAL/LEADS_IN/
directory and load it into the database automatically(VICIDIAL_IN_new_leads_file.pl
[a sample lead file in the proper format is included with this release:
 - test_VICIDIAL_lead_file.txt])
If you want to use the lead import script I suggest looking at the code to make
sure it is entering what you want it to. We are not going to go over that 
particular script in this document.

Also, there is a web-based lead loader that was made available as of the 1.1.1
release and is accessible from the VICIDIAL admin.php web page(click on the 
"LOAD NEW LEADS" link at the top of the admin page). To get to this page you 
must have permissions in the vicidial_user table(Load Leads set to 1) . 
Instructions on it's use are included on the page through the help question 
mark link.
NOTE: in PHP you must have "fileuploads" enabled for this page to work.

Second we need to add the disposition statuses into the system, all of these 
queries are below:
(Note: you may want to replace 7275551212 with a real number to test in these 
records)

1.  at the command prompt type go to the mysql client: 
/usr/local/mysql/bin/mysql
2.  type the following into the mysql client prompt:
   (make sure you put your IP address in place of "10.10.10.15" in the queries 
below)
######------ BEGIN Mysql data entry(you can copy and paste this into terminal) #

insert into vicidial_list values('','2004-01-06','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead01','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-06','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead02','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead03','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead04','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead05','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead06','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead07','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');

### these first 5 must be in all VICIDIAL systems for it to work properly #
insert into vicidial_statuses values('NEW','New Lead','N');
insert into vicidial_statuses values('QUEUE','Lead To Be Called','N');
insert into vicidial_statuses values('INCALL','Lead Being Called','N');
insert into vicidial_statuses values('DROP','Agent Not Available','N');
insert into vicidial_statuses values('NA','No Answer AutoDial','N');

insert into vicidial_statuses values('CALLBK','Call Back','Y');
insert into vicidial_statuses values('A','Answering Machine','Y');
insert into vicidial_statuses values('B','Busy','Y');
insert into vicidial_statuses values('DC','Disconnected Number','Y');
insert into vicidial_statuses values('DEC','Declined Sale','Y');
insert into vicidial_statuses values('DNC','DO NOT CALL','Y');
insert into vicidial_statuses values('SALE','Sale Made','Y');
insert into vicidial_statuses values('N','No Answer','Y');
insert into vicidial_statuses values('NI','Not Interested','Y');
insert into vicidial_statuses values('NP','No Pitch No Price','Y');
insert into vicidial_statuses values('XFER','Call Transferred','Y');

quit
######------ END Mysql data entry ------######

Now that the sample leads and disposition codes have been entered, we can go 
into the VICIDIAL administration website and set up our campaigns, lists and users.
But first let's make sure that they have the right GMT offset:
	run this on the command line:
	 - /home/cron/ADMIN_adjust_GMTnow_on_leads.pl

3. Enter the astguiclient administration page:
http://10.10.10.15/vicidial/admin.php
(use the username and password created when we entered a record into the 
vicidial_users table in SUBPHASE 6.3, In our case this is 6666 and 1234)
NOTE: if you click on the Logout button you must leave the user/pass empty and click OK
   - Now that you are logged into the vicidial administration system we can add 
new user entries for each of the new users and enter new campaigns and new 
lists.
	- The first step is to enter your new users, Click on the ADD A NEW USER 
and fill in the appropriate information for each now user you want to 
add.
	- Next, you need to create a new campaign, click on the ADD A NEW 
CAMPAIGN link and fill in what you want the campaign to be called as 
well as a description
	- Next, you need to define a new list, click on the ADD A NEW LIST link 
and fill in what you want the list to be called as well as a using the 
list ID of the leads that we loaded in the previous step "101" and 
select the new campaign from the pull-down menu that we just created.
	- Now that you have created your list, make it active by changing active to Y
	- now modify your campaign ang change the first status to be called to 
NEW and submit. Now your system is ready to dial.
   - you are done



SUBPHASE 6.7: VICIDIAL remote agents:

With v1.0 of VICIDIAL we have the ability to use a simple web form to give
remote agents a way to receive calls to whatever number they happen to be at, 
and they can view/edit call details and see a call log all through a web page
(vdremote.php) or http://10.10.10.15/vicidial/vdremote.php on this installation.
Remote Agents is only recommended for inbound calls because of the extra time 
needed to dial a number out to transfer the call to. To set up remote agents,
just go to the vicidial admin.php page and ADD NEW REMOTE AGENTS(Make sure the 
userID start also has a user login so they can get to the vdremote page). You 
will see that you can set up a remote agent entry to take multiple lines if you
wish meaning that, for example, if you need to send all of your calls to another
location because of a massive snowstorm(and none of your agents showed up at work)
you just log in that remote agent record with say 10 lines and then all of those
calls will be directed to the same number you set up for the remote agent. Then 
again you could just get your agents to log in from home if they have a phone 
and computer



SUBPHASE 6.8: astGUIclient web-only client:

With 1.1.1 release of astguiclient we have completely rewritten the astGUIclient
client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time 
GUI interface using only a web browser. The browser requirements for this are:
- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
- Mozilla 1.7 or greater
- Netscape 8 or greater
- Opera 8.5 or greater
- Microsoft Internet Explorer 5.0 or greater (we have noticed some serious
  memory leaks in IE5 and IE6 when using this program, they do not occur in
  Firefox which is why we recommend it overall.)

This new version also has more flexibility and functionality than the perl/Tk
version as well as being prettier. We have successfully tested this on many 
platforms and in remote locations. It functioned wonderfully off-site with one
of our IAX hardphones and offers a lot of promise for road warriors who need
a lot of options on their phone usage like conferencing and a detailed call log.

To log into this app you will need a login setup in the vicidial_users table 
with a user_level of 4 or greater as well as an entry for the phone you are 
using in the phones table. You will first get a login prompt for the vicidial
login then you will have the phone login where you enter the Login and Password
for that phone entry. From there the app should display and you will see the 
MAIN screen with your phone information, voicemail display and your inbound/
outbound phone call log.

The example web page you would go to on this installation would be:
http://10.10.10.15/agc/astguiclient.php

The inbound log and callerID popup is dependant on having a call_inbound.agi 
entry in your dialplan before you phone is dialed(see subphase 6.2 step 2)

Another thing to note is that you can have the agc folder(with the .php files 
in it) copied to multiple web servers, you just need to make sure that the 
MySQL database connection works (check the settings in the dbconnect.php file
that is in the agc directory). We have had astguiclient.php running on 3 
separate web servers for the same DB server and Asterisk server. This is an 
easy way to allow for auto failover and/or redundancy. Also, this client will 
work over SSL connections(https) for encrypted communications with the server.

New in astGUIclient release 1.1.7 is multi-language support. multi-language 
versions of web-clients and admin pages are available in the LANG_agc.zip and 
LANG_admin.zip files and can be unzipped into your webroot directory. Make sure
you check the dbconnect.php file in each directory.

NOTE: make sure you have your machine time-syncing with an ntp server
 Windows instructions: http://www.tech-recipes.com/windows_tips466.html
                       http://support.microsoft.com/kb/q216734/


SUBPHASE 6.9: VICIDIAL web-only client:

NOTE: There is a VICIDIAL Agent manual available from http://www.eflo.net

With 1.1.6 release of astguiclient we have finished the rewrite of the VICIDIAL
client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time
GUI interface using only a web browser like we have done with astGUIclient.
The browser requirements for this are:
- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
- Mozilla 1.7 or greater
- Netscape 8 or greater
- Opera 8.5 or greater
- Microsoft Internet Explorer 5.0 or greater (we have noticed some serious
  memory leaks in IE5 and IE6 when using this program, they do not occur in
  Firefox which is why we recommend it overall.)

This version is fully functional and has been tested in our production 
call center with no problems. On RedHat 9.0 Linux workstations we have noticed 
a decrease in system load running the web-client as compared to the Perl/Tk 
client. Also, the memory usage stayed at about 22MB for a full shift of 
activity, so no memory leaks with Firefox. 

To log into this app you will need a login setup in the vicidial_users table 
with a user_level of 1 or greater as well as an entry for the phone you are 
using in the phones table. You will first get a login prompt for the vicidial
login then you will have the phone login where you enter the Login and Password
for that phone entry. From there the app should display and you will see the 
VICIDIAL screen with your phone information.

The example web page you would go to on this installation would be:
http://10.10.10.15/agc/vicidial.php

One more feature that the VICIDIAL web-client offers is the ability to set up 
an EXTERNAL phone extension in the astguiclient admin section so that you can 
have agents log in to vicidial.php wherever they have access to a phone with 
an external phone number and a web browser. To do this follow these steps:
- "ADD A NEW PHONE" in the admin.php web page and enter whatever name you want
- For the dialplan number field put in the full digits that you would dial from
  the Asterisk server to get to that agent's external phone(with 91 if used)
- For the Protocol select EXTERNAL
- make sure the agent knows the login and password set for this phone entry.
Then the agent will go to the vicidial.php page and enter in their phone 
login/pass, their vicidial user/pass/campaign and their phone should ring in a 
few seconds, and they are logged in and ready to take calls.

Another thing to note is that you can have the agc folder(with the .php files 
in it) copied to multiple web servers, you just need to make sure that the 
MySQL database connection works (check the settings in the dbconnect.php file
that is in the agc directory). We have had astguiclient.php running on 3 
separate web servers for the same DB server and Asterisk server. This is an 
easy way to allow for auto failover and/or redundancy. Also, this client will 
work over SSL connections(https) for encrypted communications with the server.

New in astGUIclient release 1.1.7 is multi-language support. multi-language 
versions of web-clients and admin pages are available in the LANG_agc.zip and 
LANG_admin.zip files and can be unzipped into your webroot directory. Make sure
you check the dbconnect.php file in each directory.

Admin Note: If you want to enable your agents to login with only their user/pass
you can hardcode the phone_login and phone_pass into the query string(URL) and
use a bookmark on their desktop, taking one more step out of their login process
example: http://10.10.10.15/agc/vicidial.php?pl=gs102&pp=test

It is recommended if you are in a call center environment that you would disable
the "Saved Form Information" option in Firefox settings. This is a checkbox in 
the Privacy settings under the Options menu.

NOTE: make sure you have your machine time-syncing with an ntp server
 Windows instructions: http://www.tech-recipes.com/windows_tips466.html
                       http://support.microsoft.com/kb/q216734/



PHASE 7.0: You are done with installation

Now you can celebrate, you have installed a complete phone system with dialer 
and GUI applications on all of your clients. the astGUIclient has a help screen 
that pops up and explains most of it's features. 

If you have problems and it is not working right(and are NOT celebrating right
now), feel free to take a look at the FAQ for solutions to common installation 
errors or send an email to the mailing list:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users

Also, check out our weblog: http://astguiclient.blogspot.com/


**** IMPORTANT - In order for vicidial/astguiclient to function correctly please
read the REQUIREMENTS.txt for a minimum requirements list. ***
