[Asterisk-Dev] AGI script - Hang up

Steven Critchfield critch at basesys.com
Wed Sep 17 07:16:29 MST 2003


On Wed, 2003-09-17 at 06:48, Areski wrote:
> Hello,
> 
> 
> I tried to create an agi with the asterisk-perl but no success.
> 
> I used the most simple sample just to test and I cannot not hear
> anything, I don't know what's happend. Someone can help me ???
> 
> Perl prog:
> 
> #!/usr/bin/perl
> use Asterisk::AGI; 
> $AGI = new Asterisk::AGI;
> 
> # pull AGI variables into %input
> %input = $AGI->ReadParse();
> 
> # say the number 1984
> $AGI->say_number(1984)
> 
> 
> 
> that's the debug answer :
> 
> Sep 17 13:43:52 DEBUG[5126]: File chan_sip.c, Line 3786 (check_user):
> Setting NAT on RTP to 0
> Sep 17 13:43:52 DEBUG[5126]: File chan_sip.c, Line 4811
> (handle_request): Check for
> res
> Sep 17 13:43:52 DEBUG[5126]: File chan_sip.c, Line 951 (find_user): Call
> from user 'phone1' is 1 out of 0
> Sep 17 13:43:52 DEBUG[5126]: File chan_sip.c, Line 3252 (build_route):
> build_route:
> Contact hop: <sip:phone1 at 192.168.1.23:5061>
> Sep 17 13:44:02 DEBUG[14350]: File chan_sip.c, Line 979 (sip_hangup):
> find_user(phone1)
> Sep 17 13:44:02 DEBUG[5126]: File chan_sip.c, Line 538 (__sip_ack):
> Stopping retransmission on
> '0E3ED59B-8DC4-4C10-9DDB-14A7F46D232C at 192.168.1.23' of Response 20357:
> Found

It may be me, but it looks like you didn't even make it to your agi
script. Looks like you have a SIP phone problem to take care of first.
Possibly if it normally works with some other things, try answering the
line before you get to your agi app.
-- 
Steven Critchfield <critch at basesys.com>




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