similar to: AGI scripts

Displaying 20 results from an estimated 7000 matches similar to: "AGI scripts"

2010 Dec 06
1
Execute DialPlan Context without Answer App
Hi, i have context in a dialplan, I want to "execute" this context without insert the Answer Application (s? ..without call any ext). Example : [sistema-allarmi-principale] exten => s,1,Set(GRUPPO=${DIAL:-2:1}) exten => s,2,Set(ALLARME=${DIAL:1:1}) exten => s,3,AGI(checkgroup.php|${GRUPPO}) ;rest of... I tried with a Call Data File.. i create a CallDataFile like this :
2006 Feb 14
9
Solution for 1 time blast of 200, 000 recorded calls
Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on
2006 Dec 26
1
agi+cepstral driving me nuts
I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =
2011 Apr 09
1
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even
2009 Mar 09
3
problem with an agi in PHP
Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725000 at mnupprx1:1] Answer("SIP/33179977999-b6c18478", "") in new stack -- Executing [0170725000 at mnupprx1:2] GotoIf("SIP/33179977999-b6c18478", "0?6:3)") in new stack -- Goto (mnupprx1,0170725000,3) -- Executing
2007 May 14
3
Proper AGI use with MySQL
Hi, We have a "simple" AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect
2006 Nov 02
1
AGI Problems
Hi, I've got a setup whereby calls come into the asterisk server (1.2.7.1) over a IAX2 trunk and into a dialplan that launches a php AGI script: [live-full] exten => _X.,1,Set(TIMEOUT(absolute)=0) exten => _X.,2,NoOp(${EXTEN}) exten => _X.,3,DEADAGI(live-full.php) exten => _X.,4,Wait,2 exten => _X.,5,Hangup The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
2005 May 11
1
HELP: ASTCC (AGI) meets call forward ERROR
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
2007 Sep 15
2
AGI/PHP: missing arguments
hi folks, I've built a simple PHP-script utilizing the AGI-interface. in extensions.conf I trigger the script and pass a single value as first argument: exten => h,1,DeadAGI(process.php|${Enter}) On the Asterisk-console, I can actually see that the script is called correctly (something like "DeadAGI(process.php|1234)"). However, when I read stdin in the PHP script, I receive
2007 May 09
0
Trixbox drops call after running AGI script
Hey, I'm hoping somebody knows the answer to this. The script works fine on the old Trixbox 1.0 but have recently upgraded (just testing in VMWare) to Trixbox 2.2 What happens is Trixbox will drop the call after I call the AGI command in my dial plan. I first of generate a call file to call the user, then connect them to an extension in the dial plan [voice-report] exten =>
2006 Feb 20
1
Dial from AGI = no ring back ??
Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back
2007 Dec 07
2
PHP AGI script
I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use
2007 Jun 15
2
combining AGI with dialplans
On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have
2007 Aug 13
1
AGI answering the channel even though I never asked it to
I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a
2005 Mar 15
2
How to determine the voicemail file name for an AGI script
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
2006 Jan 13
1
Re: <Ben Higley> Can you send us your AGI CDR logging application?
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 > > >
2006 May 30
1
Asterisk::AGI and DIALEDTIME
Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the
2006 Mar 20
4
simple perl-agi - where's the error?
Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200
2008 Feb 06
0
Problem forwarding a call with an AGI script
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 =>
2007 Aug 28
1
calls being forwarded to neighbor?? please help, thx :)
hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box