similar to: AGI CDR Update (with set variable) problem.

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Thank You . But now i get solved with that error since I had some mistakes in installing googletts.agi Now when calling from my softphone i have written dialplan with an AGI script to convert from text to speech. It get executed without error but there is no sound getting played. My output, == Using SIP RTP CoS mark 5 -- Executing [1310 at Client-dial-Menu:1]

not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>

Hello! https://forums.asterisk.org/ is doing it again - "Content Encoding Error. An error occurred during a connection to forums.asterisk.org. Please contact the website owners to inform them of this problem". Which is odd, as the Qualys test seems to pass, only losing a point for supporting TLS 1.0. But I know it's not just me because Pingdom can't read the page, either.

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

i have a problem with AGI in Asterisk 1.0.5, the problem occurs either with PHP or C AGI scripts/programs. Well, its simple, either asterisk is not sending correctly the command responses to the standard output, or for some unknown reason to me the scripts/programs are not able to read it from standard input. I have the next C test program for AGI: #include <stdio.h> main() { char

I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2. I hve written a simple script that reads out the callerid using flite. My problem is that I seems the script is not getting the callerID. Bellow is the script _________________ #!/usr/bin/php -q <?php /** * @package phpAGI_examples * @version 2.0 */ set_time_limit(30);

Dear All, Just hope your guys out there can help me through..since I've been playing for serval hours....and still not able to resolve it... The workflow as I've created an .call file for Asterisk to call out and it's working fine with outdial, passing variable to asterisk..But the problem is when the calls reached Context and execute AGI script, the script didn't get any

Hi all, I have a very simple AGI program in Java, but it "sometimes" hangup before the whole process is completed. Looking at the /var/log/asterisk/full, I found the following: AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << STREAM FILE beep "" Jan 20 07:32:30 WARNING[-169026640]: Wait failed (No such file or directory) Jan

Hi list. I?m using ASTCC with callerid authentication and got things working fine, except for one single issue: Using this command -- DeadAGI(astcc.agi|${CALLERIDNUM}|${EXTEN:1}|5) -- and passing the 5 parameter for silent, it exits unexpectedly. I tested the 4, 3, 2 and 1 and they are working ok. But... when I use the five it returns zero and exit without executing the dial command! Take a

Apologies if this has been brought up before, but extensive googling and digging through my list archive didn't turn anything up. Basically, I'm working on an AGI web app and need to read some digit input. I'm having multiple issues with asterisk interpreting agi commands at the moment, but I figured I'd start with this one. when I call GET DATA or STREAM FILE I don't

-- jand. more than just a group Asterisk AGI Command GET DATA is usually of this form GET DATA timeout max_digits When I execute this command, I get only a single digit, regardless of what the value of max_digits is, Also the script quits Immediately after the press of the digit regardless of what the value of timeout is, This is really un-desirable as I will like to GET multiple DTMF digits

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Hi, I'm trying to achieve the following: Incoming call for user A (97), user A make a blind transfer to user B's phone (96). User B's phone rings and since there is no one to take the call, it returns the call to User A with an AGI script. The dialplan looks like this: [local] .... exten => 96,1,Dial(SIP/user4,10,tr) exten => 96,2,AGI(transfer.php) exten =>

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I