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webrtc problem

Added by Mohamed Lariche over 4 years ago

Hi,

i have problem on goautodial v4 outbound calls, when agent start calls there is warnings on asterisk cli here is it:

[Oct 7 23:14:48] WARNING9175[C-00000166]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI") in new stack
-- <Local/99997525170237@default-000000df;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
-- Called 55558600052@default
-- Executing [55558600052@default:1] Hangup("Local/55558600052@default-000000e0;2", "") in new stack
Spawn extension (default, 55558600052, 1) exited non-zero on 'Local/55558600052@default-000000e0;2'
[Oct 7 23:14:48] WARNING[9396][C-00000167]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("Local/55558600052@default-000000e0;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
-- &lt;Local/55558600052@default-000000e0;2&gt;AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
-- Called 99997525170237@default
-- Executing [99997525170237@default:1] Dial("Local/99997525170237@default-000000e1;2", "SIP/7525170237@kamailio,,tTo") in new stack
Using SIP RTP CoS mark 5
-- Called SIP/7525170237@kamailio
-- SIP/kamailio-000000d1 is ringing
-- Local/99997525170237@default-000000e1;1 is ringing
> 0x7f57e80386d0 -- Strict RTP learning after remote address set to: 192.168.1.198:30258
-- SIP/kamailio-000000d1 answered Local/99997525170237@default-000000e1;2
-- Local/99997525170237@default-000000e1;1 answered
-- Executing [8600052@default:1] Konference("Local/99997525170237@default-000000e1;1", "8600052,R") in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- Channel SIP/kamailio-000000d1 joined 'simple_bridge' basic-bridge &lt;23404737-9515-48d4-b49e-6dd27d274ad8&gt;
-- Channel Local/99997525170237@default-000000e1;2 joined 'simple_bridge' basic-bridge &lt;23404737-9515-48d4-b49e-6dd27d274ad8&gt;
> 0x7f57e80386d0 -- Strict RTP switching to RTP target address 192.168.1.198:30258 as source
> 0x7f57e80386d0 -- Strict RTP learning complete - Locking on source address 192.168.1.198:30258
Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 7 23:15:07] WARNING9442[C-00000169]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format


Replies (3)

RE: webrtc problem - Added by Demian Biscocho over 4 years ago

You can safely ignore those warnings.

RE: webrtc problem - Added by Mohamed Lariche over 4 years ago

Demian Lizandro Biscocho wrote:

You can safely ignore those warnings.

Hi Demian, thanks for the repily,

But there is no call that passes when agent is waiting however the manual calls work very well

RE: webrtc problem - Added by Levy Ryan Nolasco over 4 years ago

Hi,

Please post your asterisk CLI logs while attempting a call.

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