similar to: phpagi - Event "On Hangup"

I have a script that I want to leave running in the background to handle specific manager events. I'm running into a problem where it gets stuck in the wait_response function in phpagi-asmanager.php and the PHP maximum execute timeout kills the script. The script doesn't interact with the dialplan, so I cannot launch it from within Asterisk. Any pointers would be appreciated. I did

Hello everyone, Where can I find instructions on how to install PHPAGI? BTW, what's the difference between PHPAGI and PHPAGI2? Are they different products? It's hard to tell from voip-info.org... Best, Leo

hi i'd like to write a simply application in php with phpAgi that: - connect to Asterisk - call an external number using a Zap channel - play a message here is some code: <?php $asm = new AGI_AsteriskManager(); if ($asm->connect()) { $asm->Originate("Zap/g1/1","number","default","1"); /* play message... */ } else {

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi all, we use OpenSER together with Asterisk. All SIP users registers with OpenSER and asterisk is doing the voicemail thing. We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table. The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI

Hello, I am getting problem for delay call hang-up with the below scenario: PSTN User (calling Party)------------->PSTN Line ----------> FXO with Asterisk Box----------->SIP IP Phone (called party) I am using X100P card with my Asterisk-1.0.7 box. I am also using Zaptel-1.0.7 version. When PSTN user makes call to my PSTN line and after getting IVR, PSTN user dial my SIP

Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload

Hey! I'm going to implement same actions (answer,wait,dial,playback,queue,hangup) for 3 phonenumbers when call is coming. Do I have to write three separate lines for same actions for each phonenumber or is there any way to write one command lines for all three lines? Any examples? Thank you for your answers! ---------------------------------------------------------------- This mail sent

Hello, all! I'm having a problem with the following snippet that executes upon hangup: exten => h,n,Wait(5) exten => h,n,System(mv /some/file /some/other/dir/) Wait() doesn't want to seem to wait! So instead I tried: exten => h,n,System(sleep 5; mv /tmp/${CALLFILENAME} /var/spool/asterisk/outgoing/) This only executes sleep, not mv. How can I make it wait before moving the

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The

---------- Forwarded message ---------- From: Arun Kumar Chaudhary <uniquearun04 at gmail.com> Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: asterisk-users at lists.digium.com. Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi script. I am unable to detect whether it is connected to the dialed number, if

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hello, How do I install phpagi? http://phpagi.sourceforge.net/ I couldn't find any info about setup in that site, and I couldn't email the developers.so I'm lost. I know it isn't a real question for this list, but I suppose many people here already have installed it. So, how can I install it? Thanks Carlos -------------- next part -------------- An HTML

Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/

Hi, I've recently reinstalled a Diva in my asterisk server (alongside a QuadBRI :-) to test the nice features Armin has been adding in chan_capi. The capi.conf format has changed, so my question is how do I define a deflect= statement for different incoming MSN's? I've tried to define a section for each (group of) MSN with a different "deflect". Is that correct? [DIVA1]