similar to: AGI Diad number

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

I am running AGI script written in C. My script gets triggered but there is no action. I have taken the samplec file from http://home.cogeco.ca/~camstuff/agi.html. it says launched AGI script /var/lib/asterisk/agi-bin/sample.c exec format error . Did I miss anything? I am running asterisk1.0.7 can anyone please help me on these. thanks, Somesh -------------- next part -------------- An HTML

Hi Folks: I used that one example for AGI script on C web, only to fill the working with the Asterisk. I compiled and it worked great. I executed accidentally the ls -l command in directory where was the source and executable, I noted and was surprised that because the executable size was to further 20 times more than source. I executed the gcc -Os source.c -o executable.agi command several

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?

As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the callerid for internal phones and cdr? I think it can be done with AGI but don't know where to start

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com <http://www.griffin.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d983141f/attachment.htm

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

I was wondering if anyone knew how to do the following.... Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills

Also lookup AGI The WiKi and via google by using this: site:lists.digium.com <some words> W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven Critchfield Sent: Friday, March 04, 2005 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Placing a call

Hi, I am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again "Sorry". My situation is this: I havn't installed Asterisk yet but am curious the general way

I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks

Is it possible to select the audio stream that's played as a user enters a meetme conference? If you could, it would be very simple to record a users name, and then play that as the greeting to other attendees as they join the conference. If not, could someone tell me how hard it would be to modify the source? I presume at the moment the file to be played it stored in a var somewhere,

I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",