similar to: Using SetVar with System() is it possible?

I had lines 3 and 4 and added line 1 and 2 to extensions.conf exten => 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,2,Monitor(wav,${CALLFILENAME},m) exten => 106,3,hint,SIP/106 exten => 106,4,Macro(stdexten,106,${HINT}) I received this warning: WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension

The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid="John French" <103> mailbox="101" callwaiting=yes threewaycalling=yes

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I got this solved. Turns out the script WAS executing, but I forgot that apparently you need to follow "cron rules" in any BASH scripts executed via System() from an Asterisk dialplan. E. g. all paths must be fully and absolutely specified, there are no relative path references available. So I changed the the file wireless.sh which was: #!/bin/bash touch wireless-executed to

Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. -- -Shane Blog:

Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etc....but it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers

Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that we can use it without moral or legal restrictions? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

"Request For Comments on a Feature Suggestion" -- just wondering if others would find this useful. Frequently, when something really doesn't make sense, I like to bump up the console logging by editing logger.conf and changing console = error to console = debug,dtmf,error,event,info,notice,verbose,warning and entering logger reload When done, I

Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi... Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the asterisk version is 1.4.18. If in the machine is all ok, i can stop start the asterisk service no prob, my problem is when in another server (in my case, debian etch 4) using the ssh the stop service is ok, but the start service dosend finalise. Like this: ssh etx9 'sudo /etc/init.d/asterisk stop' operacao at

I'm trying to find a way to setup a SIP server that will mainly echo back a request from one agent only, my question is do i need to setup any of the other conf files besides extensions.conf and sip.conf? the book says "If you plan on a pure VoIP network, the only real requirement is the asterisk package," , but i'm doubtful about zaptel. can anyone help me out?

Hi there, Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? Thanks in

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the