similar to: Can AGI be used in this way?

Folks, I want to build a custom IVR for my setup. I've got it working (well, the bells and whistles are not there yet, but the basic stuff works) using AGI, but I'm worried about how well this will scale. I've seen references to FastAGI, and presumably this will be more efficient. Question, though: how well do either of these (AGI or FastAGI) scale if my system is handling a large

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

I'm starting to write an AGI script. I want to pass more than one parameter to the script, but seem to be unable to. extensions.conf: exten => 85,1,AGI(/etc/asterisk/agi/args.agi,myarg1,myarg2) args.agi: #!/usr/bin/perl print STDERR "FNORD prog = $0\n"; print STDERR "FNORD arg 1 = $ARGV[0]\n"; print STDERR "FNORD arg 2 = $ARGV[1]\n"; print

Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

Hi. I am using #include to include a file in extensions.conf. I have an agi perl script which modifies the #included file and then forces an asterisk reload with 'system("asterisk -rx reload")'; After the reload I use set_context, set_extension and set_priority to tell asterisk where I want it to carry on from which is line 2 of the following. exten => *80,1,AGI(bla.pl,

I have set up an application which records incoming calls (customer problems), and then places outgoing calls (handing the problems to support people). I have written some AGI programs (in C) to implement parts of the functionality. So far I am very pleased with the overall application. But I have one very annoying, very nasty problem: The C AGI is handling a dialog with a user on an outgoing

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

I've read several of the Wiki sections on Voicemail and "Asterisk variables" but could not find an obvious answer to this question. I would like to run a script that post-processes the voicemail after the Voicemail application returns (with AGI or DeadAGI), but I cannot figure out how to easily determine the name of the file written by the Voicemail application. Does anyone

Does anyone how you might detect a period of x milliseconds of silence using AGI ? Rgds, Stuart

Hello. When the caller hangup the phone, asterisk kills my AGI python script without notification. I caught all signals, but none was trigered. How can i trap this event to resume some operations. Sorry for my poor english :) Thanks.

Hi there, Is there an accepted way of running an AGI script on answering of a channel? Is it even possible? I don't need to execute AGI commands, I just need to know a channel has been answered. Thanks, David. -- "One world, one web, one program" -- Microsoft promotional advert. "Ein Volk, ein Reich, ein Fuehrer" -- Adolf Hitler.

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, First off, can AGI scripts be created using PHP??.. This is where our skills are and since PHP can be run from a command line it would be easier to create and maintain.. I understand that AGI works using STDIN and STDOUT, so does that mean that I would simply "echo" the standard Asterisk commands that I wanted Asterisk to execute and it would process them? Are any call

Has anyone else had trouble with the AGI command GET DATA on the latest stable cvs? I can't get it to work with asterisk-perl, or by using print statements and reading stdin. I get "200 result= (timeout)". (this is from the print statements, and asterisk-perl reports nothing). But asterisk is getting DTMF because my menu in extensions.conf works. I will go through the code

Hi All, I've been playing about with the RECORD FILE agi function and am finding two distinct problems with the resulting wav file when using a non zero sample offset. Specifically, I call the function with a zero offset and a given filename (the "original" recording), and then later call it with the same filename and a non-zero offset (the "overdub"). When I do this, I

I say this note is important only because I (a AGI PHP newbie) was tormented by this problem for many an hour, even though I'm sure it's documented somewhere or obvious to more experienced users. So as I was experimenting with AGI in PHP scripting I was baffled by why Asterisk was properly receiving AGI commands written to stdout but always returning "510 invalid command" to the

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX

Hi All, When I call the stream_file function all goes well if the user doesn't clear the call. But if I do clear the call (on the handset for example), I get the following exception: -- Channel 0/31, span 1 got hangup RESULT_LINE: 200 result=-1 endpos=28000 == Spawn extension (default, 600006, 1) exited non-zero on 'Zap/31-1'