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Asterisk/astguiclient install from scratch. v.2.2.0 2009-04-03
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Asterisk/astguiclient install from scratch. v.2.2.0 2009-06-28
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By the VICIDIAL group info@vicidial.com
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**** IMPORTANT - In order for vicidial/astguiclient to function correctly please
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@@ -6,6 +6,11 @@ read the REQUIREMENTS.txt for a minimum requirements list. ***
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End-user Manuals for Agents and Managers are available from http://www.eflo.net
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There is also an easier Ubuntu install doc: Ubuntu_Install.txt
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If you want to skip the experience of installing step-by-step completely, take
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a look at our ISO installer: http://www.vicibox.com
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This document is meant to be a very in-depth step-by-step explanation of
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installing the Asterisk open-source PBX on a Linux system and also installing
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@@ -15,46 +20,40 @@ in choices of hardware and software.
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The actual installation that I am doing as I write these instructions will be on
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the following hardware:
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(Changed to more realistic modern hardware)
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- Intel Core 2 Quad CPU
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- SuperMicro Motherboard
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- 4GB RAM
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- 160GB Western Digital SATA ES drive
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- Sangoma VoiceTime USB hardware timer
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- 2U rackmount case with 450W power supply
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- VOIP Phone will be a Zoiper softphone because it's free and does both IAX/SIP
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- Pentium 3 500MHz
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- Intel motherboard D815BN
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- 256MB PC133 RAM
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- 80GB IBM deskstar 7200RPM Hard Drive
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- Digium Wildcard Single Span T1 Card T100P
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- 2U rackmount case with 250W power supply
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- Phone hardware will be a Grandstream BT102 and a Sipura SPA-2000 because they
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are so cheap and readily available
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All of these parts cost less than $1500 together.
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All of these parts, aside from the Digium card and the two SIP VOIP devices,
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were purchased from ebay and the entire package(with the two VOIP devices and
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all server hardware included) cost me about $1100 to put together including the
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phones and Digium adapter.
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We have many other Asterisk servers at our main office, but this one can be
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experimented with easily because it was so cheap to make and has a relatively
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small capacity when compared with a multi-processor server with a quad span T1
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card.
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This is our test Asterisk server and functions well for a dozen or so extensions
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This is our test Asterisk server and functions well for 100 or so extensions
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in use if it were to be used in production. A size that is optimal for many
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small offices operating with a fractional data/voice T1 for instance.
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small offices operating with a fractional data/voice T1 for instance. This
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hardware would also be good for a stand-alone ViciDial system with about 50
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agents handling inbound calls with full recording, or 25 agents doing heavy
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predictive outbound dialing.
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For hardware you can use almost any Pentium-class processor(PII, PIII, Athlon,
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Xeon, etc), and you can use any digium telco interface card. Both of these
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choices will determine what the capacity of your Asterisk server will be. If you
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want to do simple IVR or conference calling and a few extensions, then a PIII
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with a single Digium T1 card will work just fine for you. If you want to use the
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VICIDIAL application, you will want to get as high-powered of a machine as you
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can afford and get a digium quad-span T1 card.
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For hardware you can use almost any modern CPU processor(Intel core2, Xeon, etc)
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, and you can use any Digium or Sangoma telco interface card or USB timer. Both
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of these choices will determine what the capacity of your Asterisk/ViciDial
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server will be. If you want to do simple IVR or conference calling and a few
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extensions, then an Intel Celeron 2GHz with a single Sangoma USB VoiceTime
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dongle will work just fine for you. If you want to use the ViciDial applications
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, you will usually want to get as high-powered of a machine as you can afford.
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The following is assumed for these installation procedures:
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- You have access to a CD burner and 3 blank CDs
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- You have access to a CD burner and 3 blank CDs(or 1 CD and 1 DVD)
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- You have some sort of broadband internet connection
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- You understand basic linux commands and can use a file editor like vi
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- You have all of the necessary hardware:
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- a pentium-class computer
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- a digium telco interface card with appropriate telco lines
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- at least 1 SIP VOIP device
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- at least 1 SIP/IAX VOIP device(this can be a softphone too)
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- a Local Area Network(LAN) with extra ports enough for the new server
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and the number of phones you want
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@@ -65,7 +64,7 @@ and the number of phones you want
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PHASE 1: INSTALLING AN OPERATING SYSTEM
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This installation will be using Slackware 11 for the linux distribution,
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This installation will be using Slackware 12 for the linux distribution,
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Slackware 12.X will also work with these instructions. There are several easier
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linux distributions and there are others that are more popular, but Slackware is
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a nice non-commercial distro that has been around for a long time and proven
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@@ -83,7 +82,7 @@ to it. You will select "filesystem utilities" and then "XFDISK"
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3. Select any old partitions and delete them and then create 2 new
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partitions:
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- 70000 MB, select yes to validate, change partition type to "Linux Native"
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- 150000 MB, select yes to validate, change partition type to "Linux Native"
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- 3332 MB, select yes to validate, change partition type to "Linux Swap"
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- press F3 to exit and let it do it's thing, this will take an hour or so.
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@@ -172,10 +171,9 @@ If you are using a newer Digium Octasic-based echo-cancellation quad T1/E1 card
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then you need to use a 2.6 Linux kernel in order to use the echo-cancellation
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functionality of the card.
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If you want to compile a 2.6 kernel then start with OPTION 1, otherwise to
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compile the 2.4 kernel that comes with Slackware(2.4.33) start with OPTION 2:
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If you want to compile a 2.6 kernel then start with OPTION 1:
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OPTION 1: compile Linux kernel 2.6.17 *RECOMMENDED*
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OPTION 1: compile Linux kernel 2.6.17 *NOT REQUIRED*
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1. cd /usr/src
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2. wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.17.11.tar.gz
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@@ -273,44 +271,6 @@ OPTION 1: compile Linux kernel 2.6.17 *RECOMMENDED*
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tmpfs /var/spool/asterisk/monitor tmpfs rw 0 0
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21. shutdown -r 0 # reboot machine and hope it worked
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OPTION 2: compile Linux kernel 2.4.33.3 (not recommended, very old)
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1. cd /usr/src/linux # move to your linux source directory
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2. cp .config config.save # copy old config to a save file
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3. make mrproper # prep for kernel assembly
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4. make menuconfig # launch configuration menu app
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(this part is very dependant upon your own hardware)
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enable processor version # select the processor that you have
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enable SMP # if more than 1 processor or Intel HT
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enable high memory () # if more than 1GB of RAM
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enable SCSI Multiple # if SCSI drives
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enable SCSI devices AMI Megaraid # if SCSI Megaraid adapter
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enable 3com network devices # if 3com network card
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enable ext3 file system # for ext3 to work
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enable all ACPI options # for SMP to work
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enable Enhanced Real Time Clock Support in Character devices section
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# for SMP to work
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enable any other hardware specific options
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exit and save configuration
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5. make dep # build the kernel dependancies
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6. make clean # clean up the kernel build areas
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7. make bzImage # create a kernel bzImage
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8. make modules # build the modules into the image
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9. make modules_install # install kernel modules
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10. # nothing# mkinitrd /boot/initrd-XXXXXX.img XXXXXX *not needed on Slackware*
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11. cp arch/i386/boot/bzImage /boot/bzImage-XXXXXX # copy image
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(put whatever you want in XXXXXX, that is your new kernel name)
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12. cp System.map /boot/System.map-XXXXXXN # copy system map
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13. mv -f /boot/System.map /boot/System.map-orig
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14. ln -s /boot/System.map-XXXXXX /boot/System.map # symlink map
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15. vi /etc/lilo.conf # edit the lilo boot config file
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image=/boot/bzImage-XXXXXX # add the new image in above-
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label=test-XXXXXX # the previous one
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root=/dev/hda1 # device of root partition
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read-only
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16. /sbin/lilo # run the lilo reload script
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17. shutdown -r 0 # reboot machine and hope it worked
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After compiling your kernel you can run a few commands to verify that you are
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running your new kernel and that devices are running as they are supposed to:
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@@ -397,6 +357,7 @@ a more loaded system.
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skip-name-resolve
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# comment out the line 'log-bin=mysql-bin'
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max_connections = 200
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connect_timeout=60
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**** For some systems you may need to add the mysql/bin directory to your PATH:
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- PATH=$PATH:$HOME/bin:/usr/local/mysql/bin/
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@@ -423,12 +384,10 @@ a more loaded system.
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SUBPHASE 3.2: Installing Perl Modules
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NOTE - you can install ActiveState http://www.activestate.com perl which may
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improve performance, but it is not required. Here's the source for ActiveState
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NOTE - you can install ActiveState http://www.activestate.com perl, but it is
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not required. Here's the source for ActiveState
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Perl 5.8: (it's free)
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http://downloads.activestate.com/ActivePerl/src/5.8/AP817_source.tgz
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I hope to add the lengthy steps for installing it as your default perl on your
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server but I need some time and a free machine to do that.
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cpan is the "Comprehensive Perl Archive Network". It's a mirrored archive of
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most of the perl modules out there complete with a installation and management
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@@ -475,7 +434,7 @@ following modules first: (say YES if asked to install prerequisites)
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- then quit cpan, you are done
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5. Go to http://asterisk.gnuinter.net/ and download the asterisk-perl module
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(backup link: http://download.vicidial.com/packages/asterisk-perl-0.08.tar.gz)
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NOTE: Do NOT use the 0.09 version, it does not work with VICIDIAL
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NOTE: Do NOT use the 0.09 or any newer version, they do not work with VICIDIAL
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- to install this directly on the command line type:
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- cd /usr/local
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- wget http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz
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@@ -886,121 +845,54 @@ PHASE 4: INSTALLING ASTERISK
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OK, all the prep work is done, now it's time to start having fun with Asterisk.
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There are two basic ways to install Asterisk, an official release(at the time of
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this writing the official release is 1.2.30.2) and the SVN_DEV version(development
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branch). We recommend using Asterisk 1.2.30.2. These instructions are how we get
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our Asterisk system with it's T1 line installed with our 2 SIP VOIP devices and
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one IAX2 softphone.
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this writing the recommended release is 1.4.21.2). These instructions are how we
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get our Asterisk system with it's T1 line installed with our Zoiper softphone.
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NOTE: If you want to use Asterisk 1.4, you will need to make sure that you set
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NOTE: If you want to use Asterisk 1.2, you will need to make sure that you set
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the servers table "asterisk_version" field to the proper version number and you
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can use the docs/conf_examples/extensions.conf.sample-1.4 file for your default
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can use the docs/conf_examples/extensions.conf.sample file for your default
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dialplan
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NOTE: If you want to use release 1.0.8 or 9 we would recommend either using the
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CVS_v1-0 branch where the issues are fixed, or patching your 1.0.8/1.0.9 code
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with the following patch:
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(http://astguiclient.sourceforge.net/experimental_code/localmasq.patch)
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- If you do patch your system make sure you put the asterisk version
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field for the server on the admin pages as '1.0.11.1'
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1. follow these command line steps:
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- mkdir /usr/src/asterisk
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- cd /usr/src/asterisk
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A. if you want 1.2 release (reliable with new features):
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- wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.2.30.2.tar.gz
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- wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.2.27.tar.gz
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- wget http://downloads.digium.com/pub/libpri/releases/libpri-1.2.5.tar.gz
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- gunzip asterisk-1.2.30.2.tar.gz
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- tar xvf asterisk-1.2.30.2.tar
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- gunzip zaptel-1.2.27.tar.gz
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- tar xvf zaptel-1.2.27.tar
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- gunzip libpri-1.2.5.tar.gz
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- tar xvf libpri-1.2.5.tar
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B. if you want latest SVN_1.2 version (release tree with new patches)
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- svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
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- svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
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- svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
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C. if you want latest SVN_DEV version (not recommended) [1.6 tree]
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- svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
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- svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
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- svn checkout http://svn.digium.com/svn/libpri/trunk libpri
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- ALL ->
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- (1.0 tree)if you want to allow for more than 100 voicemail messages in a
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mailbox(warning this will slightly increase memory usage when a call
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is in voicemail) edit the voicemail source code file:
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- vi /usr/src/asterisk/asterisk/apps/app_voicemail.c
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edit this line and change 100 to 999:
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#define MAXMSG 100
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- (1.0 tree)if you have no X server installed on your Asterisk machine, then you
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will need to comment out the gtk console lib(only affects 1.0 releases)
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edit the voicemail source code file:
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- vi /usr/src/asterisk/asterisk/pbx/Makefile
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edit this line at the top and just add a hash # in front of it as shown:
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#PBX_LIBS+=$(shell $(CROSS_COMPILE_BIN)gtk-config --cflags >/dev/null 2>/dev/null && echo "pbx_gtkconsole.so")
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- cd ./zaptel-1.2.27
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- make clean
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- make
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- make install
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- cd ../libpri-1.2.5
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- make clean
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- make
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- make install
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- cd ../asterisk-1.2.30.2
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- (1.2 tree) If you want to include Answering Machine Detection ability
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you will need to download app_amd.c and amd.conf and alter the
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apps/Makefile to compile it properly
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- cd apps
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- wget http://www.eflo.net/files/app_amd2.c
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|
|
- 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
|
|
|
|
|
*OPTIONAL*(1.2.23 thru 1.2.30.2) apply the meetme DTMF passthru patch
|
|
|
|
|
- wget http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch
|
|
|
|
|
- patch -p1 < ./meetme_DTMF_passthru-1.2.23.patch
|
|
|
|
|
- File to patch: app_meetme.c
|
|
|
|
|
*OPTIONAL*(1.2.12.1 thru 1.2.30.2) apply the meetme volume control patch
|
|
|
|
|
*Different patches available for 1.2.7.1 through 1.2.14
|
|
|
|
|
- wget http://www.eflo.net/files/meetme_volume_control_1.2.16.patch
|
|
|
|
|
- patch -p1 < ./meetme_volume_control_1.2.16.patch
|
|
|
|
|
- File to patch: app_meetme.c
|
|
|
|
|
- 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
|
|
|
|
|
|
|
|
|
|
-(gcc version 4.2) apply the gsm audio codec patch to fix gsm
|
|
|
|
|
- wget http://download.vicidial.com/asterisk-patches/1.2-gsm-gcc4.2.patch
|
|
|
|
|
- patch -p1 ./codecs/gsm/Makefile 1.2-gsm-gcc4.2.patch
|
|
|
|
|
|
|
|
|
|
*OPTIONAL*(1.2.14 thru 1.2.30.2) rewrite of waitforsilence
|
|
|
|
|
- wget http://download.vicidial.com/asterisk-patches/app_waitforsilence.c
|
|
|
|
|
- mv -f app_waitforsilence.c apps/app_waitforsilence.c
|
|
|
|
|
|
|
|
|
|
*OPTIONAL* shorter enter and leave sounds for meetme
|
|
|
|
|
- wget http://www.eflo.net/files/enter.h
|
|
|
|
|
- wget http://www.eflo.net/files/leave.h
|
|
|
|
|
- mv -f enter.h apps/enter.h
|
|
|
|
|
- mv -f leave.h apps/leave.h
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
- make clean
|
|
|
|
|
- make
|
|
|
|
|
- make install
|
|
|
|
|
- make samples # this makes sample conf files (only use for new installs)
|
|
|
|
|
mkdir /usr/src/asterisk-1.4
|
|
|
|
|
cd /usr/src/asterisk-1.4
|
|
|
|
|
wget http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.4.21.2.tar.gz
|
|
|
|
|
wget http://downloads.digium.com/pub/zaptel/zaptel-1.4.12.1.tar.gz
|
|
|
|
|
wget http://downloads.digium.com/pub/libpri/libpri-1.4.9.tar.gz
|
|
|
|
|
tar xzf asterisk-1.4.21.2.tar.gz
|
|
|
|
|
tar xzf zaptel-1.4.12.1.tar.gz
|
|
|
|
|
tar xzf libpri-1.4.9.tar.gz
|
|
|
|
|
cd libpri-1.4.9
|
|
|
|
|
make clean; make; make install
|
|
|
|
|
cd ../zaptel-1.4.12.1
|
|
|
|
|
./configure; make clean; make; make install
|
|
|
|
|
cd ../asterisk-1.4.21.2
|
|
|
|
|
wget http://www.eflo.net/files/enter.h
|
|
|
|
|
wget http://www.eflo.net/files/leave.h
|
|
|
|
|
mv -f enter.h apps/enter.h
|
|
|
|
|
mv -f leave.h apps/leave.h
|
|
|
|
|
wget http://download.vicidial.com/asterisk-patches/1.4-gsm-gcc4.2.patch
|
|
|
|
|
patch -p1 ./codecs/gsm/Makefile 1.4-gsm-gcc4.2.patch
|
|
|
|
|
wget http://download.vicidial.com/conf/res_agi_defunct.patch
|
|
|
|
|
patch -p1 < res_agi_defunct.patch
|
|
|
|
|
File to patch: res/res_agi.c
|
|
|
|
|
cd apps/
|
|
|
|
|
rm -f app_waitforsilence.c
|
|
|
|
|
wget http://download.vicidial.com/conf/app_waitforsilence.c
|
|
|
|
|
cd ../channels/
|
|
|
|
|
rm chan_sip.c
|
|
|
|
|
wget http://download.vicidial.com/conf/chan_sip.c
|
|
|
|
|
cd ../
|
|
|
|
|
./configure; make clean; make; make install
|
|
|
|
|
# this makes sample conf files (only use for new installs)
|
|
|
|
|
make samples
|
|
|
|
|
modprobe zaptel
|
|
|
|
|
|
|
|
|
|
- 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
|
|
|
|
|
or if you have no timer, just "modprobe ztdummy"
|
|
|
|
|
Here's the list of all digium cards and the modules you use with
|
|
|
|
|
them:
|
|
|
|
|
Card Module
|
|
|
|
@@ -1429,141 +1321,8 @@ files, so be warned:
|
|
|
|
|
- vi /etc/asterisk/extensions.conf
|
|
|
|
|
|
|
|
|
|
You should be using the sample extensions.conf that is included with the
|
|
|
|
|
release of VICIDIAL that you installed, below is just an explanation of what
|
|
|
|
|
most of those entries do and why they are there.
|
|
|
|
|
|
|
|
|
|
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
|
|
|
|
|
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
|
|
|
|
|
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing
|
|
|
|
|
|
|
|
|
|
[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(||${EXTEN:1}@default)
|
|
|
|
|
|
|
|
|
|
######------ END extensions.conf example ------######
|
|
|
|
|
|
|
|
|
|
### OPTIONAL IAX trunk extensions entries for long distance dialing over IAX
|
|
|
|
|
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
|
|
|
|
|
exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,o)
|
|
|
|
|
exten => _91NXXNXXXXXX,3,Hangup
|
|
|
|
|
|
|
|
|
|
### OPTIONAL SIP trunk extensions entries for long distance dialing over SIP
|
|
|
|
|
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
|
|
|
|
|
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@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
|
|
|
|
|
### FOR MORE INFORMATION, READ THE LOAD_BALANCING.txt DOCUMENT
|
|
|
|
|
### server 1 extens:
|
|
|
|
|
exten => _010*010*010*016*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)
|
|
|
|
|
### server 2 extens:
|
|
|
|
|
exten => _010*010*010*015*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
|
|
|
|
|
release of VICIDIAL that you installed. There is also an auto-generated
|
|
|
|
|
extensions-vicidial.conf file that contains dynamic dialplan entries.
|
|
|
|
|
|
|
|
|
|
10. edit dnsmgr.conf: # It is very helpful to enable dnsmgr
|
|
|
|
|
[general]
|
|
|
|
@@ -1781,7 +1540,7 @@ create database asterisk;
|
|
|
|
|
|
|
|
|
|
NOTE: if you will be using lead files with a language that does not use the
|
|
|
|
|
standard latin character set then you will want to use UTF8 for your default
|
|
|
|
|
characterset in the MySQL database. This requires at least MySQL 4.1.11 and you
|
|
|
|
|
characterset in the MySQL database. This requires at least MySQL 5.0.X and you
|
|
|
|
|
can use the following query to create the database:
|
|
|
|
|
CREATE DATABASE `asterisk` DEFAULT CHARACTER SET utf8 COLLATE utf8_unicode_ci;
|
|
|
|
|
|
|
|
|
@@ -1794,9 +1553,6 @@ GRANT RELOAD ON *.* TO cron@localhost;
|
|
|
|
|
|
|
|
|
|
flush privileges;
|
|
|
|
|
|
|
|
|
|
# NOTE: if using MySQL 4.1.X or higher(not 5.X) you may need to run this query too:
|
|
|
|
|
UPDATE mysql.user set password=OLD_PASSWORD('1234') where user='cron';
|
|
|
|
|
|
|
|
|
|
# To make sure that new processes can connect to the Database under load we should
|
|
|
|
|
# increase the global connect_timeout
|
|
|
|
|
|
|
|
|
@@ -1909,7 +1665,9 @@ to work with it.
|
|
|
|
|
|
|
|
|
|
Again, if you have selected to use the sample conf files during installation
|
|
|
|
|
then you do not have to add any of these lines to your conf files, they should
|
|
|
|
|
already be included.
|
|
|
|
|
already be included. There is also an auto-generated
|
|
|
|
|
extensions-vicidial.conf file that contains dynamic dialplan entries.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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
|
|
|
|
@@ -2009,7 +1767,9 @@ exten => _9NXXXXXX,3,Hangup
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
2. Add the call_inbound entries to all incoming extensions entries that you want
|
|
|
|
|
CallerID popups on:
|
|
|
|
|
CallerID popups on(ONLY WORKS IN astguiclient.php APPLICATION):
|
|
|
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(Using the DID configuration is the preferred method of call routing in ViciDial)
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- here is how our sample dialplan changes for adding call_inbound
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entries(only effected extension groups are show):
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######------ START extensions.conf changes for call_inbound ------######
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@@ -2034,61 +1794,6 @@ exten => _*NXXNXXXXXX*3429,3,Answer
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exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to
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exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000
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; parameters for agi-VDAD_ALL_inbound.agi (9 fields separated by five dashes "-----"):
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; 1. the method of call handling for the script:
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; - CID - CID received, add record with phone number
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; - CIDLOOKUP - Lookup CID to find record in whole system
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; - CIDLOOKUPRL - Restrict lookup to one list
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; - CIDLOOKUPRC - Restrict lookup to one campaign's lists
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; - CLOSER - Closer calls from VICIDIAL fronters
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; - ANI - ANI received, add record with phone number
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; - ANILOOKUP - Lookup ANI to find record in whole system
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; - ANILOOKUPRL - Restrict lookup to one list
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; - 3DIGITID - Enter 3 digit code to go to agent
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; - 4DIGITID - Enter 4 digit code to go to agent
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; - 5DIGITID - Enter 5 digit code to go to agent
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; - 10DIGITID - Enter 10 digit code to go to agent
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; 2. the method of searching for an available agent:
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; - LO - Load Balance Overflow only (priority to home server)
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; - LB - <default> Load Balance total system
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; - SO - Home server only
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; 3. the full name of the IN GROUP to be used in vicidial for the inbound call
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; 4. the phone number that was called, for the log entry
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; 5. the callerID or lead_id of the person that called(usually overridden)
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; 6. the park extension audio file name if used
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; 7. the status of the call initially(usually not used)
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; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)
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; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)
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; 10. the campaign_id to search within lists if CIDLOOKUPRC
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; inbound VICIDIAL call with CID delivery through T1 PRI
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exten => 1234,1,Answer ; Answer the line
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exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)
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exten => 1234,3,Hangup
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; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
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exten => _90009.,1,Answer ; Answer the line
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exten => _90009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
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exten => _90009.,3,Hangup
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exten => _990009.,1,Answer ; Answer the line
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exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
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exten => _990009.,3,Hangup
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### follow these instructions if you plan to have VICIDIAL agents take inbound or closer calls:
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1. in VICIDIAL web admin "add a new in-group" (the above examples would be "CL_GALLERIA")
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- group IDs cannot contain spaces ' ' or dashes '-' or plusses '+'
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- if you are using a HEX color value make sure to include the hash '#' at the beginning
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2. create a new campaign in VICIDIAL called "CLOSER" and set "allow inbound blended" to Y
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3. check the CL_GALLERIA checknox in the "Allowed In-Groups" section
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4. have agents log in to the CLOSER campaign and select the CL_GALLERIA in-group
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5. they will now start receiving inbound calls
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6. as calls come in, each call is inserted into the vicidial_list table under the
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list specified int the AGI string, In the above example that would be list 999
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7. if you want to take closer calls from the campaign "TEST" you will need to create
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an in-group called "CL_TEST_" for internal closing(on the same system) or "CL_TEST_L"
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for local closing(closer on different system from fronter) and then the fronter will
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click on the "internal closer" button to send the call to a closer
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* NOTE, you need to set the dial_level of the CLOSER campaign to 1 or higher for inbound/closers to work
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######------ END extensions.conf changes for call_inbound ------######
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@@ -2320,7 +2025,7 @@ exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
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6. Add the conference entries for VICIDIAL conferences:
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- here is how our sample dialplan changes for adding VICIDIAL conference
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entries(this is a pure addition, nothing is being modified):
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entries(this is already in the sample extensions.conf file):
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NOTE: see below these entries for app_conference instructions is used
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######------ START extensions.conf changes for VD conf ------######
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exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
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@@ -2364,6 +2069,7 @@ NOTE: If you want to do DTMF passthru with app_conference bee sure to add the
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7. Add the more entries for astGUIclient specific uses:
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- here are some more dialplan additions needed to use astGUIclient(this is a
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pure addition, nothing is being modified):
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(this is already in the sample extensions.conf file)
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######------ START extensions.conf other additions ------######
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; park channel for client GUI parking, hangup after 30 minutes
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; create a GSM formatted audio file named "park.gsm" that is 30 minutes long
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@@ -2808,7 +2514,7 @@ SUBPHASE 6.8: astGUIclient web-only client:
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With 1.1.1 release of astguiclient we have completely rewritten the astGUIclient
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client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time
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GUI interface using only a web browser. The browser requirements for this are:
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- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
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- Firefox 1.0 or greater *** RECOMMENDED ***
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- Mozilla 1.7 or greater
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- Netscape 8 or greater
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- Opera 8.5 or greater
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@@ -2856,7 +2562,7 @@ With 1.1.6 release of astguiclient we have finished the rewrite of the VICIDIAL
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client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time
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GUI interface using only a web browser like we have done with astGUIclient.
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The browser requirements for this are:
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- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)
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- Firefox 1.0 or greater *** RECOMMENDED ***
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- Mozilla 1.7 or greater
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- Netscape 8 or greater
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- Opera 8.5 or greater
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mattf