After setting up opus and callback to a softphone (connected to an asterisk). I was able to get the softphone to call. However when answering the call falls and gives this error:
[19:55:30.454][t:23209][p:23201][pjsip][debug] RX 883 bytes Response msg 200/INVITE/cseq=14581 (rdata0x7f655c002848) from UDP 192.168.0.200:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bKPjZLjBibP3Bj3BaiuH2cwTFjRSigclfn89;received=192.168.0.21;rport=5060
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14581 INVITE
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5070
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 1356515537 1356515537 IN IP4 192.168.0.200
s=Asterisk PBX 11.25.3
c=IN IP4 192.168.0.200
t=0 0
m=audio 10732 RTP/AVP 120 96
a=rtpmap:120 opus/48000/2
a=maxptime:60
a=fmtp:120 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=sendrecv
--end msg--
[19:55:30.454][t:23209][p:23201][pjsip][debug] Call 0: updating media..
[19:55:30.454][t:23209][p:23201][pjsip][debug] Media stream call00:0 is destroyed
[19:55:30.454][t:23209][p:23201][pjsip][debug] Audio channel update..
[19:55:30.457][t:23209][p:23201][pjsip][debug] Encoder stream started
[19:55:30.457][t:23209][p:23201][pjsip][debug] Decoder stream started
[19:55:30.457][t:23209][p:23201][pjsip][debug] Audio updated, stream #0: opus (sendrecv)
[19:55:30.457][t:23209][p:23201][pjsip][debug] TX 341 bytes Request msg ACK/cseq=14581 (tdta0x7f655c0285b8) to UDP 192.168.0.200:5070:
ACK sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPj1drWwM.U3TQ-6Jfe-za93kd8wc96ruXQ
Max-Forwards: 70
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14581 ACK
Content-Length: 0
--end msg--
[19:55:30.856][t:23201][p:23201][core][debug] [23201-1] creating voip for TG #1
[19:55:30.868][t:23201][p:23201][tgvoip][info] update data saving mode, config 0, enabled 0, reqd by peer 0
[19:55:30.868][t:23201][p:23201][tgvoip][warning] Set remote endpoints, allowP2P=0, connectionMaxLayer=92
[19:55:30.868][t:23201][p:23201][tgvoip][warning] Starting voip controller
[19:55:30.868][t:23201][p:23201][tgvoip][debug] Bound to local UDP port 25380
[19:55:30.869][t:23201][p:23201][core][debug] [23201-1] bridging tgvoip audio with SIP#0
[19:55:30.869][t:23201][p:23201][pjsip][debug] Switch connect: 1 --> 2
[19:55:30.869][t:23201][p:23201][pjsip][debug] Set sound device: capture=-99, playback=-99
[19:55:30.869][t:23201][p:23201][pjsip][debug] Setting null sound device..
[19:55:30.869][t:23201][p:23201][pjsip][debug] Opening null sound device..
[19:55:30.870][t:23201][p:23201][pjsip][error] pjsua_conf_connect(id, sink.id) error: Media ports are not compatible (PJMEDIA_ENOTCOMPATIBLE) (status=220160) [../src/pjsua2/media.cpp:203]
[19:55:30.870][t:23221][p:23201][tgvoip][info] before create audio io
[19:55:30.870][t:23221][p:23201][tgvoip][info] AEC: 0 NS: 0 AGC: 0
[19:55:30.872][t:23219][p:23201][tgvoip][info] Receive thread starting
[19:55:30.874][t:23220][p:23201][tgvoip][warning] Send udp pings
[19:55:30.948][t:23201][p:23201][tgvoip][debug] Entered VoIPController::Stop
[19:55:30.948][t:23201][p:23201][tgvoip][debug] before shutdown socket
[19:55:30.948][t:23201][p:23201][tgvoip][debug] before join sendThread
[19:55:30.949][t:23219][p:23201][tgvoip][info] === recv thread exiting ===
[19:55:30.956][t:23221][p:23201][tgvoip][info] Audio initialization took 0.085953 seconds
[19:55:30.957][t:23221][p:23201][tgvoip][info] === send thread exiting ===
[19:55:30.957][t:23201][p:23201][tgvoip][debug] before join recvThread
[19:55:30.957][t:23201][p:23201][tgvoip][debug] before stop messageThread
[19:55:30.957][t:23201][p:23201][tgvoip][debug] Before stop audio I/O
[19:55:30.957][t:23201][p:23201][tgvoip][debug] Left VoIPController::Stop [need rate = 0]
[19:55:30.957][t:23201][p:23201][core][debug] [23201-1] hangup TG #1
[19:55:30.959][t:23201][p:23201][pjsip][debug] Call 0 hanging up: code=500..
[19:55:30.959][t:23201][p:23201][pjsip][debug] TX 341 bytes Request msg BYE/cseq=14582 (tdta0x25c8718) to UDP 192.168.0.200:5070:
BYE sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPjXKPQ092VLQZRSVN2BKusJyecHBPwm-NF
Max-Forwards: 70
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14582 BYE
Content-Length: 0
--end msg--
[19:55:30.960][t:23201][p:23201][pjsip][debug] Call 0 hanging up: code=0..
[19:55:30.960][t:23201][p:23201][pjsip][debug] TX 341 bytes Request msg BYE/cseq=14583 (tdta0x25cc068) to UDP 192.168.0.200:5070:
BYE sip:[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPjwifKzVYJJ8dLUYtRg1nLqnES-gM6HdIe
Max-Forwards: 70
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14583 BYE
Content-Length: 0
--end msg--
[19:55:30.961][t:23201][p:23201][tgvoip][debug] Entered VoIPController::~VoIPController
[19:55:30.961][t:23201][p:23201][tgvoip][debug] before close socket
[19:55:30.961][t:23201][p:23201][tgvoip][debug] before delete audioIO
[19:55:30.961][t:23201][p:23201][tgvoip][debug] before delete encoder
[19:55:30.961][t:23201][p:23201][tgvoip][debug] before delete echo canceller
[19:55:30.962][t:23201][p:23201][tgvoip][debug] Left VoIPController::~VoIPController
[19:55:30.969][t:23209][p:23201][pjsip][debug] RX 474 bytes Response msg 200/BYE/cseq=14582 (rdata0x7f655c002848) from UDP 192.168.0.200:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bKPjXKPQ092VLQZRSVN2BKusJyecHBPwm-NF;received=192.168.0.21;rport=5060
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14582 BYE
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
--end msg--
[19:55:30.970][t:23209][p:23201][pjsip][debug] Call 0: deinitializing media..
[19:55:30.971][t:23209][p:23201][pjsip][debug] Media stream call00:0 is destroyed
[19:55:30.971][t:23209][p:23201][pjsip][debug] RX 474 bytes Response msg 200/BYE/cseq=14583 (rdata0x7f655c002848) from UDP 192.168.0.200:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bKPjwifKzVYJJ8dLUYtRg1nLqnES-gM6HdIe;received=192.168.0.21;rport=5060
From: sip:[email protected];tag=Tm535bOi8HAF.Rn9STvvswSmw0HMayxs
To: sip:[email protected];tag=as530bb5ad
Call-ID: wgYgBgtxzcFlqG3UFkUohl1DMOeOy9qA
CSeq: 14583 BYE
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
--end msg--
[19:55:32.062][t:23209][p:23201][pjsip][debug] Closing sound device after idle for 1 second(s)
[19:55:32.062][t:23209][p:23201][pjsip][debug] Closing null sound device..