similar to: Problem forwarding a call with an AGI script

Tzafrir Cohen wrote: > > The agent picks up the phone but neither the agent nor the caller > > > here anything. >So please provide a more complte trace and a the relevant partt of your >dialplan. > Here is the relevant part of the dialplan: [local] exten => 98,1,Dial(SIP/sguenther,20,tr) exten => 98,2,VoiceMail(98|u) exten => 98,3,hangup exten =>

Hi, I'm experiencing some strange problems with out Digium card. First the details abount hardware and software: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) Asterisk 1.6.0.20 dahdi-linux-complete-2.2.1-rc2+2.2.1-rc2 libpri-1.4.10.2.tar.gz The problem now is that there are a number of clients that I call and suddenly the connection drops. Just a few seconds later the clients calls

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

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

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hi, I recently upgraded our asterisk server from some 1.4 version to version 1.6.0.15. From this point on my AGI scripts aren't working anymore, here is a simple example: [isdin] exten => 83086921,1,AGI(test.php) exten => 83086921,2,NOOP("MARKE1") exten => 83086921,3,WAIT(2) exten => 83086921,4,Hangup() /var/lib/asterisk/agi-bin/test.php

Hello, How can I achieve to play a music file instead of typical ring ring (something like MusicOnHold)... I have the following dialplan, the time when the user calls the context is executed and the system calls both the user and I hear a Ringing sound. [inc-call] exten => s,1,Dial(DAHDI/i1/USER2&DAHDI/i1/USER1,20,A(sound-file)) Please suggest -- Regards, Ashish Agarwal

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hello, I'm new to this list. I'm trying to configure my Asterisk to have user access their email. SO far users can leave voicemail but they can't access voicemail. As you can see I had sip.conf and extensions.conf below. Please advice how to access configure extensions.conf to have users access their voicemail. Thanks in advance. -motty SIP.CONF [general] context=default

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

Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does "SET EXTENSION <new extension>" allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database

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 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

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Verdana">Hi all. I've written a AGI in C language.

Hello I'm pretty much a newbie when it comes to C, but I have to use this language to write a couple of AGI proggies because I need them to be statically compiled. Strangely enough, Google didn't return much when looking for the "Hello, world!" of AGI in C. The following doesn't work: The file never gets written: =========== //check_cid.c #include <stdio.h> #include

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

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