similar to: AGI Dial channel status

Hello I'm using an AGI script in Lua to make a callback through Zaptel. For this to work, I must wait until the channel is idle, or I get this kind of error, even after waiting over 10 seconds after the remote end rings once and hangs up: ============== channel.c:2863 __ast_request_and_dial: Unable to request channel Zap/1/123456 pbx_spool.c:341 attempt_thread: Call failed to go through,

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

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'

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 setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

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

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

I have an AGI program with an array containing a set of ${UNIQUEID} variables for channels that may be active on the system. I need a way for the program to tell if they are or not. It's certainly possible using the manager interface, or appropriate "asterisk -rx" commands, but I'd prefer to do it directly from AGI for performance, security, and ease of configuration. Does

I am looking for a way to quickly and easily test for on-hook channels from within a C-language AGI app. CHANNEL STATUS works but is a bit clumsy. I don't want to rely on strcmp'ing the returned '200 result=-1' as the meaning of this might change in the future. I am trying to create an ACD using MySQL and want to test each channel before I Dial it. Any ideas? Jim Friedeck

Hi folks -- I have a FastAGI Perl script running, handling calls. It works great. At one point I have a Dial() command. If the called party hangs up, Dial() returns 0, and when I call my own recordCdr() function using the channel variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine. However, if the called party picks up, and then the dialing party hangs up Dial() returns -1,

hi, I have an AGI running after an outgoing call file starts it up. Everything works fine except if my line has a problem. Trying to simulate this I unplug the line. So there is no dialtone. How do I detect this and let the AGI know so I can try line 2, 3, 4 etc... Detecting the the AGI or some other way is fine. I just need to know. I am using a TDM804B card at this time. Jerry

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =

I've isolated this problem the furthest that I can, and I'm now convinced this is a bug in asterisk. I have a context in extensions.conf like so: [my_context] exten => _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL}) exten => _X.,2,GOTO(my_other_context|${EXTEN}|1) exten => h,1,DeadAGI(my_agi_cleanup) For the purposes of this scenario, my_agi simply will try to HANGUP the channel to

I know that hangup handlers ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly. So it's no surprise that my speech to text agi which takes 8 seconds gets killed. However, can anyone think of a way round this? So, once the caller has hung up, I need to take one of the channel variables, and pass it to a python agi script which then does speech to text.

At the moment I'm prototyping an advanced ENUM application with PHP fetched from LDAP. When a user enters a full hostname as SIP adress I get loop problems from the AGI EXECUTE DIAL and from a Dial in the extension.conf. -- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php enum.php: 123 enum.php:

Hello List, This is my first message herein. I was playing around with System() and AGI() and found out something I cound not determine my configuration error. I added before.agi and after.agi to the agi-bin dir. Tried to make before.agi get run before the dial call and after.agi be run after. Only the first priority (step 1) gets executed. Here follows some relevant part of the tests: On

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

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

I have two possible senarios for making a call from an AGI.. Senario1 - Using GoTo In the extensions.conf I have.. [dial-out] exten => _9.,1,AGI(myagi) exten => _9.,2,Dial(SIP/blah/${EXTEN:1}) In the AGI I have.. EXEC GoTo dial-out|9555678|2 So using this method I don't have to really edit the AGI ever if I change the dialplan around as long at the context is correct.. At the end of

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to