search for: asmanager

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 look through the wiki and gave google a chance, but the results found didn't really suggest a solu...

Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

...00,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten =>121,4,MeetMe(900,MDbF) exten =>121,5,MeetMe(900,MD) exten =>121,100,Playback(demo-moreinfo) #!/usr/bin/php -q <?php //include("phpagi.php"); //include("phpagi-asmanager.php"); //$agi = new AGI; set_time_limit(60); ob_implicit_flush(false); if(!defined('STDIN')) { define('STDIN',fopen('php://stdin','r')); } if(!defined('STDOUT')) { define('STDOUT',fopen('php://stdout','w')); } if(!defined(...

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, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

Hi, I would like to know how to see which channels are used in my PRI E1 link from Asterisk to another locally-connected commercial PBX. If I run "dahdi show channels", I can see the used channels in the second column "extension" but only if it's an "incoming" call (ie. legacy PBX to Asterisk). If I dial from an Asterisk extension to an extension in the other

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so