similar to: AGI Channel Status

Ok, here's a weird one. I have an AGI script where one user calls another. The call is answered. Everything is peachy. If the call is terminated by the CALLEE hanging up the call, then Asterisk returns control back to where the Dial() command left off, and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great. HOWEVER, if the CALLER hangs up the call, it seems as if

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->answer(); my $i; $i = $AGI->channel_status(); $AGI->say_digits($i); $i =

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Well, I'm about ready to throw Asterisk across the room. Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI, if the callee hangs up the call, Asterisk sends a return code, but if the caller hangs up, it does not??? This means if an agi script services a call, and after the two parties have finished speaking, the person who initiated the call hangs up, the agi

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

I'm getting an error back from an AGI Dial command. Weird thing is that it's STILL performing the Dial. Here's what I am sending (without the paranthesis): (EXEC DIAL "SIP/10000|5|tr") and here's what I am getting (without the paranthesis): (510 Invalid or unknown command) Why would I get this response to a seemingly good command? And why would Asterisk go ahead and

I've got several issues with AGI/FastAGI 1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a Dial command to Asterisk, I don't get a return result until AFTER the call is HUNG UP. Not when it's ringing, not when the call is connected, but when it's

I have the following python AGI script. I know it's been abstracted, but it's still pretty easy to see what's happening. self.agi.channelAnswer() self.agi.wait(1) self.agi.execCmd("background","enter-conf-call-number","") self.agi.execCmd("Read","confNum|||","") confNum =

Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

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

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Some questions regarding calling Fast AGI from the dial plan. Considering that the server side of the Fast AGI has to be able to a) use threading and b) connect to MySQL, this causes some serious limitations. I'm not a C programmer, so development options are either perl or python. It appears that the perl DBI isn't thread safe, so that rules out Perl. Python's database interfaces

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. -- regards Vikram (http://www.vicramresearch.com)