SVN Version: 2812
DB Schema Version: 1523
Asterisk 1.8
Ubuntu Server 12.04.5 LTS 64bit
Hi guys i have setup an INbound DID# on my Dialer and when i call this Number outside of the dialer - i mean from other regular phones and VoIP device calls are coming in perfectly and able to received calls in the Ingroup. BUT the Problem is when i register a local extension to my Dialer sample extension 100 and dial directly the Inbound number from there calls are not pushing through only see the call progress but its dropping no audio at all no sign of calls connecting
- Code: Select all
[2018-06-20 22:49:07] -- Executing [21461383759034@default:1] AGI("SIP/cc150-0000001c", "agi://127.0.0.1:4577/call_log") in new stack
[2018-06-20 22:49:07] -- <SIP/cc150-0000001c>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2018-06-20 22:49:07] -- Executing [21461383759034@default:2] Dial("SIP/cc150-0000001c", "SIP/RUBIX/61383759034,50,o") in new stack
[2018-06-20 22:49:07] == Using SIP RTP CoS mark 5
[2018-06-20 22:49:07] -- Called SIP/RUBIX/61383759034
[2018-06-20 22:49:09] -- SIP/RUBIX-0000001d answered SIP/cc150-0000001c
[2018-06-20 22:49:09] -- Locally bridging SIP/cc150-0000001c and SIP/RUBIX-0000001d
[2018-06-20 22:49:09] NOTICE[1374]: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 95 received from 'xx.xxx.xxx.xxx:8000'
[2018-06-20 22:49:30] -- Executing [h@default:1] AGI("SIP/cc150-0000001c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----23-----21") in new stack
[2018-06-20 22:49:30] -- <SIP/cc150-0000001c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----23-----21 completed, returning 0
[2018-06-20 22:49:30] == Spawn extension (default, 21461383759034, 2) exited non-zero on 'SIP/cc150-0000001c'
Any idea and somehow can help on this please. Thanks a lot in Advance