similar to: Passing SIP headers to AGI applications

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

Hi List, Can any one please let me know how to pass arguments to the agi script from the dialplan? I read that it is possible to pass arguments to an AGI script here, http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed by a vertical bar but it doesn't seem to work for me. I'm using a basic AGI script to query a database and then returns to specific contexts

Hellos, I know this might be an easy one but either way I am stuck...I need to execute asterisk cli commands using php agi and get the output via the same script. How to I execute let's say "show hints" and get the output back to the script? I have tried $agi->exec("show hints"); but I am getting the output below on the cli debug AGI Rx << EXEC show hints AGI

Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/115b0aee/attachment.htm

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

exten => auth_dial,1,DigitTimeout,5 exten => auth_dial,2,ResponseTimeout,15 exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0) exten => auth_dial,4,agi(call_start.php|${dialed}) exten => auth_dial,5,dial(SIP/${dialed}@146.82.15.241) I'm trying to get What they dialed put into the PHP script. How do I get the contents of this variable in the php script?

Hi all, Sorry for this elementary question (I'm a newbie). I'm trying to write an agi script (test.agi) and run it when I call in. However, I'm getting an error that says application agi isn't being found. I've put test.agi into agi-bin with permissions 755. Do I have to compile agi support into Asterisk, or is it built in? My test.agi script is php, but not using anything

Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has

I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser

At the moment I'm prototyping an advanced ENUM application with PHP fetched from LDAP. When a user enters a full hostname as SIP adress I get loop problems from the AGI EXECUTE DIAL and from a Dial in the extension.conf. -- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php enum.php: 123 enum.php:

Requirements: 1. http://sourceforge.net/projects/odbcsock 2. The SQL server must be in the same subnet as your * server Howto: 1. Install ODBCSocketServer on your SQL server and verify connection with the included VB COM app from a Windows box. 2. Decide how you want to transform the Caller ID. In my case, I want to do a lookup of the CallerID number in SQL server and prepend the Caller ID

So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and

Before coming in here , I had a deep dig into Google and couldn't find an answer, Simply spoken, using agi->getdat in an AGI application , the call disconnects if digits are entered fast by user. I'm certain that others have been though this problem, please pour your experience here :-) . Ali Mughrabi

I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks > Dear Ben, > I've also the problems as Chris Mason, Can you send us your own AGI CDR > logging application? > Best regards, > Jian Hong Guan > France > www.directcentrex.com > > >

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween

Hi, I've noticed that the UNIQUEID for a call is not the same in the Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is when passed via STDIN to an AGI script. Is this normal, and is this supposed to behave this way? The UNIQUEID received in the AGI is usually .001 higher than the one in the dial plan -- but sometimes it is also a second behind. Here's an example

I'm wondering if anyone has a solution to this before I begin looking at making some changes to the SIP channel. Basically when calling SIPAddHeader() twice from the Dialplan or an AGI script with the same header name it adds duplicate headers instead of overwriting the existing one. Here's a practical situation where this applies. A call is to be terminated via SIP and we have two

Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. The

Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find