[SIP-Account] Can't answer a SIP-call
using Jami 20230616-01 on Android 13 (Samsung Galaxy A53 5G)
might be related: #1302 (closed)
Step to reproduce:
- In the past you've created a SIP-account, used it successfully and disabled it for a few weeks (never used again; however other clients were used for the same SIP-account; but the SIP-Account in Jami has been always disabled; so the phone has been probably rebooted many times)
- (Now you disable all other clients for the same SIP-account) Now enable the SIP-account in Jami again
- Make a call (Phone 1 (SIM) -> Phone 2(SIP-Account; using Jami) )
- You'll get a notification "Decline" or "Answer in audio"
- If you tap "Answer in audio" nothing will happen but another noficiation "Answer in audio" will appear
- You can tap like 4 next notification "Answer in audio" but nothing will happen
- After that the call will fail
- You'll get a "missed call" notification
NOTE:
- Enabeling/Disabeling the STUN-option didn't change anything
- Enabeling/Disabeling the SIP-account didn't change anything
- Switching between wifi and mobile networks didn't change anything
Activity
Here are the logs (I just replaced the account and used "acc123" instead; and username "123456" and SIP "IP 12.345.67", etc.)
[1688635828.904|26037|manager.cpp :935 ] ############## START MONITORING ############## [1688635828.904|26037|manager.cpp :936 ] Using PJSIP version 2.12.1 for aarch64-unknown-linux-android [1688635828.905|26037|manager.cpp :937 ] Using GnuTLS version 3.7.6 [1688635828.905|26037|manager.cpp :938 ] Using OpenDHT version 2.5.5 [1688635828.905|26037|manager.cpp :953 ] ############## END MONITORING ############## [1688635835.180|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635835.232|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635835.232|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635835.233|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z1hG1bK64c1.1aabd418d6f4648454c9e4db69f9e373.0 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK17ced233 Max-Forwards: 69 From: "Anonymous" ;tag=as1dca9806 To: Contact: Call-ID: 37b3f3d9020ee6da329f8ea357336648@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 360 [1688635835.233|26082|sipcall.cpp :149 ] [call:4700878282458380] Create a new [INCOMING] SIP call with 1 media [1688635835.234|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3f049560 - call Id 4700878282458380 [1688635835.234|26082|ringbuffer.cpp :55 ] Create new RingBuffer 4700878282458380 [1688635835.234|26082|sipcall.cpp :2007] [call:4700878282458380] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635835.234|26082|sipcall.cpp :2010] [call:4700878282458380] Created 1 Media streams [1688635835.235|26082|sipcall.cpp :456 ] [call:4700878282458380] Setting transport to [0xb400006eced067c8] [1688635835.235|26082|sipcall.cpp :473 ] [call:4700878282458380] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635835.235|26082|sipcall.cpp :1706] [call:4700878282458380] Set peer's User-Agent to [Asterisk] [1688635835.235|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635835.235|26082|call.cpp :247 ] [call:4700878282458380] state change 0/0, cnx 0/2, code 0 [1688635835.235|26082|call.cpp :278 ] [call:4700878282458380] emit client call state change CONNECTING, code 0 [1688635835.244|26082|sipcall.cpp :744 ] [call:4700878282458380] Set new invite session [0xb400006f2eda9518] [1688635835.245|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635835.245|26082|sipvoiplink.cpp :901 ] [call:4700878282458380] INVITE@0xb400006f2eda9518 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed64d98 status 100 (Trying) [1688635835.245|26082|call.cpp :247 ] [call:4700878282458380] state change 0/0, cnx 2/1, code 0 [1688635835.246|26082|call.cpp :278 ] [call:4700878282458380] emit client call state change INACTIVE, code 0 [1688635835.247|26082|sipvoiplink.cpp :901 ] [call:4700878282458380] INVITE@0xb400006f2eda9518 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed64d98 status 180 (Ringing) [1688635835.247|26082|sipcall.cpp :1811] [call:4700878282458380] Peer ringing [1688635835.247|26082|call.cpp :247 ] [call:4700878282458380] state change 0/0, cnx 1/3, code 0 [1688635835.247|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635835.247|26082|call.cpp :278 ] [call:4700878282458380] emit client call state change INCOMING, code 0 [1688635835.248|26082|call.cpp :247 ] [call:4700878282458380] state change 0/0, cnx 3/3, code 0 [1688635835.248|26082|manager.cpp :2483] Incoming call 4700878282458380 on account 1234 with 1 media [1688635835.271|26082|call.cpp :247 ] [call:4700878282458380] state change 0/0, cnx 3/3, code 0 [1688635835.272|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635835.272|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635835.279|26082|audio_player.cpp :200 ] OpenSL playback start [1688635835.336|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635835.346|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635835.346|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635835.347|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635835.348|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635835.452|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635835.458|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635835.458|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635835.459|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z1hG1bK64c1.1aabd418d6f4648454c9e4db69f9e373.1 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK17ced233 Max-Forwards: 69 From: "Anonymous" ;tag=as1dca9806 To: Contact: Call-ID: 37b3f3d9020ee6da329f8ea357336648@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 360 [1688635835.459|26082|sipcall.cpp :149 ] [call:2721051908266330] Create a new [INCOMING] SIP call with 1 media [1688635835.459|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3eede6c0 - call Id 2721051908266330 [1688635835.459|26082|ringbuffer.cpp :55 ] Create new RingBuffer 2721051908266330 [1688635835.459|26082|sipcall.cpp :2007] [call:2721051908266330] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635835.459|26082|sipcall.cpp :2010] [call:2721051908266330] Created 1 Media streams [1688635835.459|26082|sipcall.cpp :456 ] [call:2721051908266330] Setting transport to [0xb400006eced067c8] [1688635835.459|26082|sipcall.cpp :473 ] [call:2721051908266330] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635835.459|26082|sipcall.cpp :1706] [call:2721051908266330] Set peer's User-Agent to [Asterisk] [1688635835.459|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635835.460|26082|call.cpp :247 ] [call:2721051908266330] state change 0/0, cnx 0/2, code 0 [1688635835.460|26082|call.cpp :278 ] [call:2721051908266330] emit client call state change CONNECTING, code 0 [1688635835.461|26082|sipcall.cpp :744 ] [call:2721051908266330] Set new invite session [0xb400006f2ee40418] [1688635835.461|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635835.461|26082|sipvoiplink.cpp :901 ] [call:2721051908266330] INVITE@0xb400006f2ee40418 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed6d098 status 100 (Trying) [1688635835.462|26082|call.cpp :247 ] [call:2721051908266330] state change 0/0, cnx 2/1, code 0 [1688635835.462|26082|call.cpp :278 ] [call:2721051908266330] emit client call state change INACTIVE, code 0 [1688635835.462|26082|sipvoiplink.cpp :901 ] [call:2721051908266330] INVITE@0xb400006f2ee40418 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed6d098 status 180 (Ringing) [1688635835.463|26082|sipcall.cpp :1811] [call:2721051908266330] Peer ringing [1688635835.463|26082|call.cpp :247 ] [call:2721051908266330] state change 0/0, cnx 1/3, code 0 [1688635835.463|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635835.463|26082|call.cpp :278 ] [call:2721051908266330] emit client call state change INCOMING, code 0 [1688635835.464|26082|call.cpp :247 ] [call:2721051908266330] state change 0/0, cnx 3/3, code 0 [1688635835.464|26082|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635835.464|26082|audio_player.cpp :239 ] OpenSL playback stop [1688635835.464|26082|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635835.464|29127|audio_player.cpp :56 ] AudioPlayer buffer lost [1688635835.465|26082|manager.cpp :2483] Incoming call 2721051908266330 on account 1234 with 1 media [1688635835.475|26069|sipcall.cpp :2224] [call:2721051908266330] Stopping all media [1688635835.475|26069|audio_rtp_session.cpp :229 ] [0xb400006f3eede6c0] Stopping receiver [1688635835.475|26069|sipcall.cpp :756 ] [call:2721051908266330] Terminate SIP session [1688635835.475|26069|sipvoiplink.cpp :901 ] [call:2721051908266330] INVITE@0xb400006f2ee40418 state changed to 6 (DISCONNCTD): cause=486, tsx@0xb400006f4ed6d098 status 486 (Busy Here) [1688635835.475|26069|call.cpp :247 ] [call:2721051908266330] state change 0/3, cnx 3/0, code 0 [1688635835.475|26069|call.cpp :278 ] [call:2721051908266330] emit client call state change BUSY, code 0 [1688635835.697|26069|sipcall.cpp :730 ] [call:2721051908266330] Delete current invite session [1688635835.697|26069|call.cpp :247 ] [call:2721051908266330] state change 3/3, cnx 0/0, code 103 [1688635835.697|26069|sipcall.cpp :1589] [call:2721051908266330] removeCall() [1688635835.697|26069|call_factory.cpp :72 ] Removing call 2721051908266330 [1688635835.697|26069|call_factory.cpp :75 ] Remaining 1 call [1688635835.697|26069|call.cpp :247 ] [call:2721051908266330] state change 3/6, cnx 0/0, code 0 [1688635835.697|26069|call.cpp :278 ] [call:2721051908266330] emit client call state change OVER, code 0 [1688635835.697|26069|sipcall.cpp :456 ] [call:2721051908266330] Setting transport to [0x0] [1688635835.698|26069|manager.cpp :1648] [call:2721051908266330] Remove local audio [1688635835.698|26082|call.cpp :247 ] [call:2721051908266330] state change 6/6, cnx 0/3, code 0 [1688635835.698|26069|ringbufferpool.cpp :262 ] Unbind call 2721051908266330 from all bound calls [1688635835.698|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635835.698|26069|manager.cpp :1098] Could not hang up non-existant call 2721051908266330 [1688635835.698|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635835.698|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635835.698|26079|manager.cpp :1973] [call:2721051908266330] Busy [1688635835.698|26079|sipcall.cpp :1589] [call:2721051908266330] removeCall() [1688635835.699|26079|call_factory.cpp :72 ] Removing call 2721051908266330 [1688635835.699|26079|call_factory.cpp :75 ] Remaining 1 call [1688635835.699|26079|call.cpp :247 ] [call:2721051908266330] state change 6/6, cnx 3/3, code 0 [1688635835.706|26082|audio_player.cpp :200 ] OpenSL playback start [1688635835.708|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635835.716|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635835.717|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635835.717|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635835.718|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635835.815|26082|audio_rtp_session.cpp :229 ] [0xb400006f3eede6c0] Stopping receiver [1688635835.816|26082|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3eede6c0 - call Id 2721051908266330 [1688635835.816|26082|ringbuffer.cpp :60 ] Destroy RingBuffer 2721051908266330 [1688635836.876|26069|manager.cpp :1049] Answer call 4700878282458380 [1688635836.879|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635836.887|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635836.888|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635836.891|26069|sipcall.cpp :889 ] [call:4700878282458380] Answering incoming call with following media: [1688635836.891|26069|sipcall.cpp :892 ] [call:4700878282458380] Media @0 - type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635836.891|26069|sipcall.cpp :2336] [call:4700878282458380] [audio_0] already un-muted [1688635836.891|26069|sdp.cpp :606 ] Processing received offer for [Call ID 4700878282458380] with 1 media [1688635836.892|26069|sdp.cpp :503 ] [SDP OFFER] Remote session: v=0 o=root 58394789 58394789 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 49718 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635836.892|26069|sdp.cpp :263 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES]] [1688635836.892|26069|sdp.cpp :503 ] [SDP ANSWER] Local session: v=0 o=localhost 3897624636 0 IN IP4 77.119.82.20 s=Call ID 4700878282458380 c=IN IP4 77.119.82.20 t=0 0 m=audio 28738 RTP/SAVP 104 9 2 112 111 110 8 0 101 a=rtpmap:104 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:112 speex/32000 a=rtpmap:111 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:28739 IN IP4 77.119.82.20 a=sendrecv [1688635836.892|26069|sipcall.cpp :3463] [call:4700878282458380] No ICE username fragment attribute in remote SDP [1688635836.893|26069|sipvoiplink.cpp :1121] [call:4700878282458380] INVITE@0xb400006f2eda9518 media update: status 0 [1688635836.893|26069|sdp.cpp :139 ] Set active local session to [0xb400006f2ee74018]. Was [0x0] [1688635836.893|26069|sdp.cpp :503 ] [SDP ANSWER] Local active session: v=0 o=localhost 3897624636 1 IN IP4 77.119.82.20 s=Call ID 4700878282458380 c=IN IP4 77.119.82.20 t=0 0 m=audio 0 RTP/AVP 8 111 0 3 110 101 [1688635836.893|26069|sdp.cpp :147 ] Set active remote session to [0xb400006f2ee178e8]. Was [0x0] [1688635836.893|26069|sdp.cpp :503 ] [SDP ANSWER] Remote active session: v=0 o=root 58394789 58394789 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 49718 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635836.893|26069|sipcall.cpp :970 ] [call:4700878282458380] Answering with contact header: [1688635836.894|26069|sipvoiplink.cpp :901 ] [call:4700878282458380] INVITE@0xb400006f2eda9518 state changed to 4 (CONNECTING): cause=0, tsx@0xb400006f4ed64d98 status 200 (OK) [1688635836.894|26069|call.cpp :247 ] [call:4700878282458380] state change 0/1, cnx 3/4, code 0 [1688635836.894|26069|call.cpp :278 ] [call:4700878282458380] emit client call state change CURRENT, code 0 [1688635836.901|26069|manager.cpp :605 ] ----- Switch current call id to '4700878282458380' ----- [1688635836.901|26069|manager.cpp :1616] Add audio to call 4700878282458380 [1688635836.901|26069|manager.cpp :1627] [call:4700878282458380] Attach audio [1688635836.901|26069|ringbufferpool.cpp :174 ] Bind call 4700878282458380 to call audiolayer_id [1688635836.901|26069|ringbufferpool.cpp :155 ] Bind rbuf '4700878282458380' to callid 'audiolayer_id' [1688635836.901|26069|ringbufferpool.cpp :155 ] Bind rbuf 'audiolayer_id' to callid '4700878282458380' [1688635836.901|26079|sipcall.cpp :2607] [call:4700878282458380] Media negotiation complete [1688635836.901|26069|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635836.902|26069|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635836.902|26079|sipcall.cpp :3463] [call:4700878282458380] No ICE username fragment attribute in remote SDP [1688635836.902|26079|sipcall.cpp :2050] [call:4700878282458380] updating negotiated media [1688635836.903|26079|sipcall.cpp :2070] [call:4700878282458380] [SDP:slot#0] The media is disabled, skipping [1688635836.903|26079|sipcall.cpp :2633] [call:4700878282458380] ICE media disabled, using default media ports [1688635836.903|26079|sipcall.cpp :2224] [call:4700878282458380] Stopping all media [1688635836.904|26079|audio_rtp_session.cpp :229 ] [0xb400006f3f049560] Stopping receiver [1688635836.904|26079|sipcall.cpp :2146] [call:4700878282458380] Starting all media [1688635836.904|26079|sipcall.cpp :2155] [call:4700878282458380] Crypto (SRTP) is negotiated over an insecure signaling transport [1688635836.904|26079|audio_rtp_session.cpp :229 ] [0xb400006f3f049560] Stopping receiver [1688635836.904|26079|sipcall.cpp :2259] [call:4700878282458380] Updating remote media [1688635836.946|26082|sipcall.cpp :1652] [call:4700878282458380] onAnswered() [1688635836.947|26082|sipvoiplink.cpp :901 ] [call:4700878282458380] INVITE@0xb400006f2eda9518 state changed to 6 (DISCONNCTD): cause=200, tsx@0xb400006f4ed64d98 status 200 (OK) [1688635836.947|26079|manager.cpp :1951] [call:4700878282458380] Peer hung up [1688635836.948|26079|sipcall.cpp :2224] [call:4700878282458380] Stopping all media [1688635836.948|26079|audio_rtp_session.cpp :229 ] [0xb400006f3f049560] Stopping receiver [1688635836.948|26079|sipcall.cpp :756 ] [call:4700878282458380] Terminate SIP session [1688635836.948|26079|sipcall.cpp :730 ] [call:4700878282458380] Delete current invite session [1688635836.949|26079|call.cpp :247 ] [call:4700878282458380] state change 1/1, cnx 4/0, code 103 [1688635836.949|26079|call.cpp :278 ] [call:4700878282458380] emit client call state change HUNGUP, code 103 [1688635837.107|26069|audio_player.cpp :200 ] OpenSL playback start [1688635837.113|26079|manager.cpp :1648] [call:4700878282458380] Remove local audio [1688635837.113|26079|ringbufferpool.cpp :262 ] Unbind call 4700878282458380 from all bound calls [1688635837.115|26079|sipcall.cpp :1589] [call:4700878282458380] removeCall() [1688635837.116|26079|sdp.cpp :139 ] Set active local session to [0x0]. Was [0xb400006f2ee74018] [1688635837.116|26079|call_factory.cpp :72 ] Removing call 4700878282458380 [1688635837.116|26079|call_factory.cpp :75 ] Remaining 0 call [1688635837.116|26079|call.cpp :247 ] [call:4700878282458380] state change 1/6, cnx 0/0, code 0 [1688635837.116|26079|call.cpp :278 ] [call:4700878282458380] emit client call state change OVER, code 0 [1688635837.117|26079|sipcall.cpp :456 ] [call:4700878282458380] Setting transport to [0x0] [1688635837.159|26069|opensllayer.cpp :111 ] OpenSL audio layer started [1688635837.160|26069|audio_rtp_session.cpp :229 ] [0xb400006f3f049560] Stopping receiver [1688635837.160|26069|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3f049560 - call Id 4700878282458380 [1688635837.160|26069|ringbuffer.cpp :60 ] Destroy RingBuffer 4700878282458380 [1688635837.160|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 0 [1688635837.160|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635837.161|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635837.343|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635837.349|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635837.349|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635837.349|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK2a52.270bfa7926a6f0414c700e6aea493d6c.0 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK711ee976 Max-Forwards: 69 From: "Anonymous" ;tag=as1358684f To: Contact: Call-ID: 5fd8db0e6aa6c1c54d439f7b08bdeeec@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 362 [1688635837.349|26082|sipcall.cpp :149 ] [call:8799028916789525] Create a new [INCOMING] SIP call with 1 media [1688635837.350|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3eeda660 - call Id 8799028916789525 [1688635837.350|26082|ringbuffer.cpp :55 ] Create new RingBuffer 8799028916789525 [1688635837.350|26082|sipcall.cpp :2007] [call:8799028916789525] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635837.350|26082|sipcall.cpp :2010] [call:8799028916789525] Created 1 Media streams [1688635837.350|26082|sipcall.cpp :456 ] [call:8799028916789525] Setting transport to [0xb400006eced067c8] [1688635837.351|26082|sipcall.cpp :473 ] [call:8799028916789525] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635837.351|26082|sipcall.cpp :1706] [call:8799028916789525] Set peer's User-Agent to [Asterisk] [1688635837.351|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635837.352|26082|call.cpp :247 ] [call:8799028916789525] state change 0/0, cnx 0/2, code 0 [1688635837.352|26082|call.cpp :278 ] [call:8799028916789525] emit client call state change CONNECTING, code 0 [1688635837.353|26082|sipcall.cpp :744 ] [call:8799028916789525] Set new invite session [0xb400006f2ee558c8] [1688635837.353|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635837.354|26082|sipvoiplink.cpp :901 ] [call:8799028916789525] INVITE@0xb400006f2ee558c8 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed4c498 status 100 (Trying) [1688635837.354|26082|call.cpp :247 ] [call:8799028916789525] state change 0/0, cnx 2/1, code 0 [1688635837.354|26082|call.cpp :278 ] [call:8799028916789525] emit client call state change INACTIVE, code 0 [1688635837.354|26082|sipvoiplink.cpp :901 ] [call:8799028916789525] INVITE@0xb400006f2ee558c8 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed4c498 status 180 (Ringing) [1688635837.355|26082|sipcall.cpp :1811] [call:8799028916789525] Peer ringing [1688635837.355|26082|call.cpp :247 ] [call:8799028916789525] state change 0/0, cnx 1/3, code 0 [1688635837.355|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635837.355|26082|call.cpp :278 ] [call:8799028916789525] emit client call state change INCOMING, code 0 [1688635837.356|26082|call.cpp :247 ] [call:8799028916789525] state change 0/0, cnx 3/3, code 0 [1688635837.356|26082|manager.cpp :2483] Incoming call 8799028916789525 on account 1234 with 1 media [1688635837.372|26082|call.cpp :247 ] [call:8799028916789525] state change 0/0, cnx 3/3, code 0 [1688635837.372|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635837.373|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635837.384|26082|audio_player.cpp :200 ] OpenSL playback start [1688635837.443|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635837.462|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635837.463|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635837.465|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635837.466|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635837.653|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635837.661|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635837.661|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635837.661|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK2a52.270bfa7926a6f0414c700e6aea493d6c.1 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK711ee976 Max-Forwards: 69 From: "Anonymous" ;tag=as1358684f To: Contact: Call-ID: 5fd8db0e6aa6c1c54d439f7b08bdeeec@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 362 [1688635837.662|26082|sipcall.cpp :149 ] [call:2724049956604191] Create a new [INCOMING] SIP call with 1 media [1688635837.662|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3ef64ce0 - call Id 2724049956604191 [1688635837.662|26082|ringbuffer.cpp :55 ] Create new RingBuffer 2724049956604191 [1688635837.662|26082|sipcall.cpp :2007] [call:2724049956604191] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635837.662|26082|sipcall.cpp :2010] [call:2724049956604191] Created 1 Media streams [1688635837.662|26082|sipcall.cpp :456 ] [call:2724049956604191] Setting transport to [0xb400006eced067c8] [1688635837.662|26082|sipcall.cpp :473 ] [call:2724049956604191] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635837.662|26082|sipcall.cpp :1706] [call:2724049956604191] Set peer's User-Agent to [Asterisk] [1688635837.662|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635837.662|26082|call.cpp :247 ] [call:2724049956604191] state change 0/0, cnx 0/2, code 0 [1688635837.662|26082|call.cpp :278 ] [call:2724049956604191] emit client call state change CONNECTING, code 0 [1688635837.664|26082|sipcall.cpp :744 ] [call:2724049956604191] Set new invite session [0xb400006f2eeae398] [1688635837.664|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635837.664|26082|sipvoiplink.cpp :901 ] [call:2724049956604191] INVITE@0xb400006f2eeae398 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed892e8 status 100 (Trying) [1688635837.665|26082|call.cpp :247 ] [call:2724049956604191] state change 0/0, cnx 2/1, code 0 [1688635837.665|26082|call.cpp :278 ] [call:2724049956604191] emit client call state change INACTIVE, code 0 [1688635837.665|26082|sipvoiplink.cpp :901 ] [call:2724049956604191] INVITE@0xb400006f2eeae398 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed892e8 status 180 (Ringing) [1688635837.666|26082|sipcall.cpp :1811] [call:2724049956604191] Peer ringing [1688635837.666|26082|call.cpp :247 ] [call:2724049956604191] state change 0/0, cnx 1/3, code 0 [1688635837.666|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635837.666|26082|call.cpp :278 ] [call:2724049956604191] emit client call state change INCOMING, code 0 [1688635837.666|26082|call.cpp :247 ] [call:2724049956604191] state change 0/0, cnx 3/3, code 0 [1688635837.667|26082|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635837.667|26082|audio_player.cpp :239 ] OpenSL playback stop [1688635837.667|26082|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635837.668|26082|manager.cpp :2483] Incoming call 2724049956604191 on account 1234 with 1 media [1688635837.677|26069|sipcall.cpp :2224] [call:2724049956604191] Stopping all media [1688635837.677|26069|audio_rtp_session.cpp :229 ] [0xb400006f3ef64ce0] Stopping receiver [1688635837.677|26069|sipcall.cpp :756 ] [call:2724049956604191] Terminate SIP session [1688635837.677|26082|call.cpp :247 ] [call:2724049956604191] state change 0/0, cnx 3/3, code 0 [1688635837.677|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635837.677|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635837.677|26069|sipvoiplink.cpp :901 ] [call:2724049956604191] INVITE@0xb400006f2eeae398 state changed to 6 (DISCONNCTD): cause=486, tsx@0xb400006f4ed892e8 status 486 (Busy Here) [1688635837.677|26069|call.cpp :247 ] [call:2724049956604191] state change 0/3, cnx 3/0, code 0 [1688635837.678|26069|call.cpp :278 ] [call:2724049956604191] emit client call state change BUSY, code 0 [1688635837.861|26082|audio_player.cpp :200 ] OpenSL playback start [1688635837.867|26069|sipcall.cpp :730 ] [call:2724049956604191] Delete current invite session [1688635837.867|26069|call.cpp :247 ] [call:2724049956604191] state change 3/3, cnx 0/0, code 103 [1688635837.867|26069|sipcall.cpp :1589] [call:2724049956604191] removeCall() [1688635837.867|26069|call_factory.cpp :72 ] Removing call 2724049956604191 [1688635837.867|26069|call_factory.cpp :75 ] Remaining 1 call [1688635837.868|26069|call.cpp :247 ] [call:2724049956604191] state change 3/6, cnx 0/0, code 0 [1688635837.868|26069|call.cpp :278 ] [call:2724049956604191] emit client call state change OVER, code 0 [1688635837.868|26069|sipcall.cpp :456 ] [call:2724049956604191] Setting transport to [0x0] [1688635837.868|26069|manager.cpp :1648] [call:2724049956604191] Remove local audio [1688635837.868|26079|manager.cpp :1973] [call:2724049956604191] Busy [1688635837.868|26069|ringbufferpool.cpp :262 ] Unbind call 2724049956604191 from all bound calls [1688635837.868|26069|manager.cpp :1098] Could not hang up non-existant call 2724049956604191 [1688635837.868|26079|sipcall.cpp :1589] [call:2724049956604191] removeCall() [1688635837.869|26079|call_factory.cpp :72 ] Removing call 2724049956604191 [1688635837.869|26079|call_factory.cpp :75 ] Remaining 1 call [1688635837.869|26079|call.cpp :247 ] [call:2724049956604191] state change 6/6, cnx 0/0, code 0 [1688635837.873|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635837.881|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635837.882|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635837.883|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635837.883|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635837.984|26082|audio_rtp_session.cpp :229 ] [0xb400006f3ef64ce0] Stopping receiver [1688635837.984|26082|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3ef64ce0 - call Id 2724049956604191 [1688635837.984|26082|ringbuffer.cpp :60 ] Destroy RingBuffer 2724049956604191 [1688635838.647|26069|manager.cpp :1049] Answer call 8799028916789525 [1688635838.647|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635838.647|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635838.647|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635838.648|26069|sipcall.cpp :889 ] [call:8799028916789525] Answering incoming call with following media: [1688635838.649|26069|sipcall.cpp :892 ] [call:8799028916789525] Media @0 - type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635838.649|26069|sipcall.cpp :2336] [call:8799028916789525] [audio_0] already un-muted [1688635838.649|26069|sdp.cpp :606 ] Processing received offer for [Call ID 8799028916789525] with 1 media [1688635838.649|26069|sdp.cpp :503 ] [SDP OFFER] Remote session: v=0 o=root 988739689 988739689 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 42420 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635838.650|26069|sdp.cpp :263 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES]] [1688635838.650|26069|sdp.cpp :503 ] [SDP ANSWER] Local session: v=0 o=localhost 3897624638 0 IN IP4 77.119.82.20 s=Call ID 8799028916789525 c=IN IP4 77.119.82.20 t=0 0 m=audio 21560 RTP/SAVP 104 9 2 112 111 110 8 0 101 a=rtpmap:104 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:112 speex/32000 a=rtpmap:111 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:21561 IN IP4 77.119.82.20 a=sendrecv [1688635838.650|26069|sipcall.cpp :3463] [call:8799028916789525] No ICE username fragment attribute in remote SDP [1688635838.650|26069|sipvoiplink.cpp :1121] [call:8799028916789525] INVITE@0xb400006f2ee558c8 media update: status 0 [1688635838.651|26069|sdp.cpp :139 ] Set active local session to [0xb400006f2eeac568]. Was [0x0] [1688635838.651|26069|sdp.cpp :503 ] [SDP ANSWER] Local active session: v=0 o=localhost 3897624638 1 IN IP4 77.119.82.20 s=Call ID 8799028916789525 c=IN IP4 77.119.82.20 t=0 0 m=audio 0 RTP/AVP 8 111 0 3 110 101 [1688635838.651|26069|sdp.cpp :147 ] Set active remote session to [0xb400006f2ee32c78]. Was [0x0] [1688635838.651|26069|sdp.cpp :503 ] [SDP ANSWER] Remote active session: v=0 o=root 988739689 988739689 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 42420 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635838.651|26069|sipcall.cpp :970 ] [call:8799028916789525] Answering with contact header: [1688635838.652|26069|sipvoiplink.cpp :901 ] [call:8799028916789525] INVITE@0xb400006f2ee558c8 state changed to 4 (CONNECTING): cause=0, tsx@0xb400006f4ed4c498 status 200 (OK) [1688635838.652|26069|call.cpp :247 ] [call:8799028916789525] state change 0/1, cnx 3/4, code 0 [1688635838.652|26069|call.cpp :278 ] [call:8799028916789525] emit client call state change CURRENT, code 0 [1688635838.671|26069|manager.cpp :605 ] ----- Switch current call id to '8799028916789525' ----- [1688635838.671|26069|manager.cpp :1616] Add audio to call 8799028916789525 [1688635838.671|26079|sipcall.cpp :2607] [call:8799028916789525] Media negotiation complete [1688635838.671|26069|manager.cpp :1627] [call:8799028916789525] Attach audio [1688635838.671|26069|ringbufferpool.cpp :174 ] Bind call 8799028916789525 to call audiolayer_id [1688635838.671|26069|ringbufferpool.cpp :155 ] Bind rbuf '8799028916789525' to callid 'audiolayer_id' [1688635838.671|26079|sipcall.cpp :3463] [call:8799028916789525] No ICE username fragment attribute in remote SDP [1688635838.672|26069|ringbufferpool.cpp :155 ] Bind rbuf 'audiolayer_id' to callid '8799028916789525' [1688635838.672|26079|sipcall.cpp :2050] [call:8799028916789525] updating negotiated media [1688635838.672|26069|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635838.672|26069|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635838.672|26079|sipcall.cpp :2070] [call:8799028916789525] [SDP:slot#0] The media is disabled, skipping [1688635838.672|26079|sipcall.cpp :2633] [call:8799028916789525] ICE media disabled, using default media ports [1688635838.672|26079|sipcall.cpp :2224] [call:8799028916789525] Stopping all media [1688635838.672|26079|audio_rtp_session.cpp :229 ] [0xb400006f3eeda660] Stopping receiver [1688635838.672|26079|sipcall.cpp :2146] [call:8799028916789525] Starting all media [1688635838.672|26079|sipcall.cpp :2155] [call:8799028916789525] Crypto (SRTP) is negotiated over an insecure signaling transport [1688635838.673|26079|audio_rtp_session.cpp :229 ] [0xb400006f3eeda660] Stopping receiver [1688635838.673|26079|sipcall.cpp :2259] [call:8799028916789525] Updating remote media [1688635838.709|26082|sipcall.cpp :1652] [call:8799028916789525] onAnswered() [1688635838.712|26082|sipvoiplink.cpp :901 ] [call:8799028916789525] INVITE@0xb400006f2ee558c8 state changed to 6 (DISCONNCTD): cause=200, tsx@0xb400006f4ed4c498 status 200 (OK) [1688635838.712|26079|manager.cpp :1951] [call:8799028916789525] Peer hung up [1688635838.713|26079|sipcall.cpp :2224] [call:8799028916789525] Stopping all media [1688635838.713|26079|audio_rtp_session.cpp :229 ] [0xb400006f3eeda660] Stopping receiver [1688635838.713|26079|sipcall.cpp :756 ] [call:8799028916789525] Terminate SIP session [1688635838.714|26079|sipcall.cpp :730 ] [call:8799028916789525] Delete current invite session [1688635838.714|26079|call.cpp :247 ] [call:8799028916789525] state change 1/1, cnx 4/0, code 103 [1688635838.714|26079|call.cpp :278 ] [call:8799028916789525] emit client call state change HUNGUP, code 103 [1688635838.874|26069|audio_player.cpp :200 ] OpenSL playback start [1688635838.878|26079|manager.cpp :1648] [call:8799028916789525] Remove local audio [1688635838.878|26079|ringbufferpool.cpp :262 ] Unbind call 8799028916789525 from all bound calls [1688635838.879|26079|sipcall.cpp :1589] [call:8799028916789525] removeCall() [1688635838.879|26079|sdp.cpp :139 ] Set active local session to [0x0]. Was [0xb400006f2eeac568] [1688635838.879|26079|call_factory.cpp :72 ] Removing call 8799028916789525 [1688635838.879|26079|call_factory.cpp :75 ] Remaining 0 call [1688635838.879|26079|call.cpp :247 ] [call:8799028916789525] state change 1/6, cnx 0/0, code 0 [1688635838.879|26079|call.cpp :278 ] [call:8799028916789525] emit client call state change OVER, code 0 [1688635838.880|26079|sipcall.cpp :456 ] [call:8799028916789525] Setting transport to [0x0] [1688635838.933|26069|opensllayer.cpp :111 ] OpenSL audio layer started [1688635838.938|26069|audio_rtp_session.cpp :229 ] [0xb400006f3eeda660] Stopping receiver [1688635838.938|26069|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3eeda660 - call Id 8799028916789525 [1688635838.938|26069|ringbuffer.cpp :60 ] Destroy RingBuffer 8799028916789525 [1688635838.940|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 0 [1688635838.940|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635838.941|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635839.373|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635839.379|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635839.379|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635839.379|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK818.d5a4ca8c287461292166b06248996da5.0 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK132e98b8 Max-Forwards: 69 From: "Anonymous" ;tag=as1d8d9075 To: Contact: Call-ID: 1d3f123c413eb45f7e3007db679a00b3@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 362 [1688635839.380|26082|sipcall.cpp :149 ] [call:625243519853867] Create a new [INCOMING] SIP call with 1 media [1688635839.380|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3ef61fd0 - call Id 625243519853867 [1688635839.380|26082|ringbuffer.cpp :55 ] Create new RingBuffer 625243519853867 [1688635839.380|26082|sipcall.cpp :2007] [call:625243519853867] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635839.380|26082|sipcall.cpp :2010] [call:625243519853867] Created 1 Media streams [1688635839.380|26082|sipcall.cpp :456 ] [call:625243519853867] Setting transport to [0xb400006eced067c8] [1688635839.380|26082|sipcall.cpp :473 ] [call:625243519853867] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635839.380|26082|sipcall.cpp :1706] [call:625243519853867] Set peer's User-Agent to [Asterisk] [1688635839.380|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635839.381|26082|call.cpp :247 ] [call:625243519853867] state change 0/0, cnx 0/2, code 0 [1688635839.381|26082|call.cpp :278 ] [call:625243519853867] emit client call state change CONNECTING, code 0 [1688635839.382|26082|sipcall.cpp :744 ] [call:625243519853867] Set new invite session [0xb400006f2ee95288] [1688635839.382|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635839.383|26082|sipvoiplink.cpp :901 ] [call:625243519853867] INVITE@0xb400006f2ee95288 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed72aa8 status 100 (Trying) [1688635839.383|26082|call.cpp :247 ] [call:625243519853867] state change 0/0, cnx 2/1, code 0 [1688635839.383|26082|call.cpp :278 ] [call:625243519853867] emit client call state change INACTIVE, code 0 [1688635839.384|26082|sipvoiplink.cpp :901 ] [call:625243519853867] INVITE@0xb400006f2ee95288 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed72aa8 status 180 (Ringing) [1688635839.384|26082|sipcall.cpp :1811] [call:625243519853867] Peer ringing [1688635839.384|26082|call.cpp :247 ] [call:625243519853867] state change 0/0, cnx 1/3, code 0 [1688635839.384|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635839.384|26082|call.cpp :278 ] [call:625243519853867] emit client call state change INCOMING, code 0 [1688635839.387|26082|call.cpp :247 ] [call:625243519853867] state change 0/0, cnx 3/3, code 0 [1688635839.387|26082|manager.cpp :2483] Incoming call 625243519853867 on account 1234 with 1 media [1688635839.403|26082|call.cpp :247 ] [call:625243519853867] state change 0/0, cnx 3/3, code 0 [1688635839.403|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635839.403|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635839.415|26082|audio_player.cpp :200 ] OpenSL playback start [1688635839.463|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635839.472|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635839.473|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635839.474|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635839.474|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635839.578|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635839.583|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635839.583|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635839.583|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK818.d5a4ca8c287461292166b06248996da5.1 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK132e98b8 Max-Forwards: 69 From: "Anonymous" ;tag=as1d8d9075 To: Contact: Call-ID: 1d3f123c413eb45f7e3007db679a00b3@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 362 [1688635839.583|26082|sipcall.cpp :149 ] [call:3916342861316615] Create a new [INCOMING] SIP call with 1 media [1688635839.584|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3ed250d0 - call Id 3916342861316615 [1688635839.584|26082|ringbuffer.cpp :55 ] Create new RingBuffer 3916342861316615 [1688635839.584|26082|sipcall.cpp :2007] [call:3916342861316615] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635839.584|26082|sipcall.cpp :2010] [call:3916342861316615] Created 1 Media streams [1688635839.584|26082|sipcall.cpp :456 ] [call:3916342861316615] Setting transport to [0xb400006eced067c8] [1688635839.584|26082|sipcall.cpp :473 ] [call:3916342861316615] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635839.584|26082|sipcall.cpp :1706] [call:3916342861316615] Set peer's User-Agent to [Asterisk] [1688635839.584|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635839.584|26082|call.cpp :247 ] [call:3916342861316615] state change 0/0, cnx 0/2, code 0 [1688635839.584|26082|call.cpp :278 ] [call:3916342861316615] emit client call state change CONNECTING, code 0 [1688635839.586|26082|sipcall.cpp :744 ] [call:3916342861316615] Set new invite session [0xb400006f2ee904e8] [1688635839.586|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635839.586|26082|sipvoiplink.cpp :901 ] [call:3916342861316615] INVITE@0xb400006f2ee904e8 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed838d8 status 100 (Trying) [1688635839.586|26082|call.cpp :247 ] [call:3916342861316615] state change 0/0, cnx 2/1, code 0 [1688635839.586|26082|call.cpp :278 ] [call:3916342861316615] emit client call state change INACTIVE, code 0 [1688635839.587|26082|sipvoiplink.cpp :901 ] [call:3916342861316615] INVITE@0xb400006f2ee904e8 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed838d8 status 180 (Ringing) [1688635839.587|26082|sipcall.cpp :1811] [call:3916342861316615] Peer ringing [1688635839.587|26082|call.cpp :247 ] [call:3916342861316615] state change 0/0, cnx 1/3, code 0 [1688635839.587|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635839.587|26082|call.cpp :278 ] [call:3916342861316615] emit client call state change INCOMING, code 0 [1688635839.588|26082|call.cpp :247 ] [call:3916342861316615] state change 0/0, cnx 3/3, code 0 [1688635839.588|26082|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635839.588|26082|audio_player.cpp :239 ] OpenSL playback stop [1688635839.589|26082|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635839.590|26082|manager.cpp :2483] Incoming call 3916342861316615 on account 1234 with 1 media [1688635839.599|26069|sipcall.cpp :2224] [call:3916342861316615] Stopping all media [1688635839.599|26069|audio_rtp_session.cpp :229 ] [0xb400006f3ed250d0] Stopping receiver [1688635839.599|26069|sipcall.cpp :756 ] [call:3916342861316615] Terminate SIP session [1688635839.599|26069|sipvoiplink.cpp :901 ] [call:3916342861316615] INVITE@0xb400006f2ee904e8 state changed to 6 (DISCONNCTD): cause=486, tsx@0xb400006f4ed838d8 status 486 (Busy Here) [1688635839.600|26069|call.cpp :247 ] [call:3916342861316615] state change 0/3, cnx 3/0, code 0 [1688635839.600|26069|call.cpp :278 ] [call:3916342861316615] emit client call state change BUSY, code 0 [1688635839.639|26069|sipcall.cpp :730 ] [call:3916342861316615] Delete current invite session [1688635839.639|26069|call.cpp :247 ] [call:3916342861316615] state change 3/3, cnx 0/0, code 103 [1688635839.640|26069|sipcall.cpp :1589] [call:3916342861316615] removeCall() [1688635839.640|26069|call_factory.cpp :72 ] Removing call 3916342861316615 [1688635839.640|26069|call_factory.cpp :75 ] Remaining 1 call [1688635839.640|26069|call.cpp :247 ] [call:3916342861316615] state change 3/6, cnx 0/0, code 0 [1688635839.640|26069|call.cpp :278 ] [call:3916342861316615] emit client call state change OVER, code 0 [1688635839.641|26069|sipcall.cpp :456 ] [call:3916342861316615] Setting transport to [0x0] [1688635839.641|26069|manager.cpp :1648] [call:3916342861316615] Remove local audio [1688635839.641|26069|ringbufferpool.cpp :262 ] Unbind call 3916342861316615 from all bound calls [1688635839.641|26082|call.cpp :247 ] [call:3916342861316615] state change 6/6, cnx 0/3, code 0 [1688635839.641|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635839.641|26069|manager.cpp :1098] Could not hang up non-existant call 3916342861316615 [1688635839.641|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635839.641|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635839.641|26079|manager.cpp :1973] [call:3916342861316615] Busy [1688635839.642|26079|sipcall.cpp :1589] [call:3916342861316615] removeCall() [1688635839.642|26079|call_factory.cpp :72 ] Removing call 3916342861316615 [1688635839.642|26079|call_factory.cpp :75 ] Remaining 1 call [1688635839.642|26079|call.cpp :247 ] [call:3916342861316615] state change 6/6, cnx 3/3, code 0 [1688635839.845|26082|audio_player.cpp :200 ] OpenSL playback start [1688635839.861|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635839.873|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635839.877|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635839.879|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635839.883|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635840.700|26082|audio_rtp_session.cpp :229 ] [0xb400006f3ed250d0] Stopping receiver [1688635840.700|26082|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3ed250d0 - call Id 3916342861316615 [1688635840.700|26082|ringbuffer.cpp :60 ] Destroy RingBuffer 3916342861316615 [1688635840.211|26069|manager.cpp :1049] Answer call 625243519853867 [1688635840.211|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635840.211|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635840.211|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635840.215|26069|sipcall.cpp :889 ] [call:625243519853867] Answering incoming call with following media: [1688635840.215|26069|sipcall.cpp :892 ] [call:625243519853867] Media @0 - type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635840.216|26069|sipcall.cpp :2336] [call:625243519853867] [audio_0] already un-muted [1688635840.216|26069|sdp.cpp :606 ] Processing received offer for [Call ID 625243519853867] with 1 media [1688635840.216|26069|sdp.cpp :503 ] [SDP OFFER] Remote session: v=0 o=root 670812692 670812692 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 49672 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635840.216|26069|sdp.cpp :263 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES]] [1688635840.216|26069|sdp.cpp :503 ] [SDP ANSWER] Local session: v=0 o=localhost 3897624640 0 IN IP4 77.119.82.20 s=Call ID 625243519853867 c=IN IP4 77.119.82.20 t=0 0 m=audio 19986 RTP/SAVP 104 9 2 112 111 110 8 0 101 a=rtpmap:104 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:112 speex/32000 a=rtpmap:111 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:19987 IN IP4 77.119.82.20 a=sendrecv [1688635840.216|26069|sipcall.cpp :3463] [call:625243519853867] No ICE username fragment attribute in remote SDP [1688635840.216|26069|sipvoiplink.cpp :1121] [call:625243519853867] INVITE@0xb400006f2ee95288 media update: status 0 [1688635840.216|26069|sdp.cpp :139 ] Set active local session to [0xb400006f2ee6a088]. Was [0x0] [1688635840.216|26069|sdp.cpp :503 ] [SDP ANSWER] Local active session: v=0 o=localhost 3897624640 1 IN IP4 77.119.82.20 s=Call ID 625243519853867 c=IN IP4 77.119.82.20 t=0 0 m=audio 0 RTP/AVP 8 111 0 3 110 101 [1688635840.216|26069|sdp.cpp :147 ] Set active remote session to [0xb400006f2ee98ee8]. Was [0x0] [1688635840.216|26069|sdp.cpp :503 ] [SDP ANSWER] Remote active session: v=0 o=root 670812692 670812692 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 49672 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635840.216|26069|sipcall.cpp :970 ] [call:625243519853867] Answering with contact header: [1688635840.217|26069|sipvoiplink.cpp :901 ] [call:625243519853867] INVITE@0xb400006f2ee95288 state changed to 4 (CONNECTING): cause=0, tsx@0xb400006f4ed72aa8 status 200 (OK) [1688635840.217|26069|call.cpp :247 ] [call:625243519853867] state change 0/1, cnx 3/4, code 0 [1688635840.217|26069|call.cpp :278 ] [call:625243519853867] emit client call state change CURRENT, code 0 [1688635840.225|26069|manager.cpp :605 ] ----- Switch current call id to '625243519853867' ----- [1688635840.225|26069|manager.cpp :1616] Add audio to call 625243519853867 [1688635840.225|26069|manager.cpp :1627] [call:625243519853867] Attach audio [1688635840.225|26079|sipcall.cpp :2607] [call:625243519853867] Media negotiation complete [1688635840.225|26069|ringbufferpool.cpp :174 ] Bind call 625243519853867 to call audiolayer_id [1688635840.225|26069|ringbufferpool.cpp :155 ] Bind rbuf '625243519853867' to callid 'audiolayer_id' [1688635840.225|26069|ringbufferpool.cpp :155 ] Bind rbuf 'audiolayer_id' to callid '625243519853867' [1688635840.225|26069|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635840.225|26069|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635840.225|26079|sipcall.cpp :3463] [call:625243519853867] No ICE username fragment attribute in remote SDP [1688635840.226|26079|sipcall.cpp :2050] [call:625243519853867] updating negotiated media [1688635840.226|26079|sipcall.cpp :2070] [call:625243519853867] [SDP:slot#0] The media is disabled, skipping [1688635840.227|26079|sipcall.cpp :2633] [call:625243519853867] ICE media disabled, using default media ports [1688635840.227|26079|sipcall.cpp :2224] [call:625243519853867] Stopping all media [1688635840.227|26079|audio_rtp_session.cpp :229 ] [0xb400006f3ef61fd0] Stopping receiver [1688635840.228|26079|sipcall.cpp :2146] [call:625243519853867] Starting all media [1688635840.228|26079|sipcall.cpp :2155] [call:625243519853867] Crypto (SRTP) is negotiated over an insecure signaling transport [1688635840.228|26079|audio_rtp_session.cpp :229 ] [0xb400006f3ef61fd0] Stopping receiver [1688635840.229|26079|sipcall.cpp :2259] [call:625243519853867] Updating remote media [1688635840.272|26082|sipcall.cpp :1652] [call:625243519853867] onAnswered() [1688635840.274|26082|sipvoiplink.cpp :901 ] [call:625243519853867] INVITE@0xb400006f2ee95288 state changed to 6 (DISCONNCTD): cause=200, tsx@0xb400006f4ed72aa8 status 200 (OK) [1688635840.274|26079|manager.cpp :1951] [call:625243519853867] Peer hung up [1688635840.274|26079|sipcall.cpp :2224] [call:625243519853867] Stopping all media [1688635840.274|26079|audio_rtp_session.cpp :229 ] [0xb400006f3ef61fd0] Stopping receiver [1688635840.275|26079|sipcall.cpp :756 ] [call:625243519853867] Terminate SIP session [1688635840.275|26079|sipcall.cpp :730 ] [call:625243519853867] Delete current invite session [1688635840.275|26079|call.cpp :247 ] [call:625243519853867] state change 1/1, cnx 4/0, code 103 [1688635840.275|26079|call.cpp :278 ] [call:625243519853867] emit client call state change HUNGUP, code 103 [1688635840.435|26069|audio_player.cpp :200 ] OpenSL playback start [1688635840.438|26079|manager.cpp :1648] [call:625243519853867] Remove local audio [1688635840.438|26079|ringbufferpool.cpp :262 ] Unbind call 625243519853867 from all bound calls [1688635840.438|26079|sipcall.cpp :1589] [call:625243519853867] removeCall() [1688635840.438|26079|sdp.cpp :139 ] Set active local session to [0x0]. Was [0xb400006f2ee6a088] [1688635840.438|26079|call_factory.cpp :72 ] Removing call 625243519853867 [1688635840.438|26079|call_factory.cpp :75 ] Remaining 0 call [1688635840.438|26079|call.cpp :247 ] [call:625243519853867] state change 1/6, cnx 0/0, code 0 [1688635840.438|26079|call.cpp :278 ] [call:625243519853867] emit client call state change OVER, code 0 [1688635840.439|26079|sipcall.cpp :456 ] [call:625243519853867] Setting transport to [0x0] [1688635840.623|26069|opensllayer.cpp :111 ] OpenSL audio layer started [1688635840.623|26069|audio_rtp_session.cpp :229 ] [0xb400006f3ef61fd0] Stopping receiver [1688635840.623|26069|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3ef61fd0 - call Id 625243519853867 [1688635840.624|26069|ringbuffer.cpp :60 ] Destroy RingBuffer 625243519853867 [1688635840.624|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 0 [1688635840.624|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635840.625|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635840.708|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635840.718|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635840.718|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635840.719|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK09cc.f2f9024142c67a48eac0708bbda1f894.0 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK48a8b6db Max-Forwards: 69 From: "Anonymous" ;tag=as1c96cf3b To: Contact: Call-ID: 1171ef825bfbcf3b1a7289796b8e37c0@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 364 [1688635840.719|26082|sipcall.cpp :149 ] [call:5798855410039169] Create a new [INCOMING] SIP call with 1 media [1688635840.720|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3efc3c20 - call Id 5798855410039169 [1688635840.720|26082|ringbuffer.cpp :55 ] Create new RingBuffer 5798855410039169 [1688635840.721|26082|sipcall.cpp :2007] [call:5798855410039169] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635840.721|26082|sipcall.cpp :2010] [call:5798855410039169] Created 1 Media streams [1688635840.722|26082|sipcall.cpp :456 ] [call:5798855410039169] Setting transport to [0xb400006eced067c8] [1688635840.722|26082|sipcall.cpp :473 ] [call:5798855410039169] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635840.722|26082|sipcall.cpp :1706] [call:5798855410039169] Set peer's User-Agent to [Asterisk] [1688635840.723|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635840.723|26082|call.cpp :247 ] [call:5798855410039169] state change 0/0, cnx 0/2, code 0 [1688635840.724|26082|call.cpp :278 ] [call:5798855410039169] emit client call state change CONNECTING, code 0 [1688635840.727|26082|sipcall.cpp :744 ] [call:5798855410039169] Set new invite session [0xb400006f2ee19718] [1688635840.727|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635840.728|26082|sipvoiplink.cpp :901 ] [call:5798855410039169] INVITE@0xb400006f2ee19718 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ed71a48 status 100 (Trying) [1688635840.729|26082|call.cpp :247 ] [call:5798855410039169] state change 0/0, cnx 2/1, code 0 [1688635840.729|26082|call.cpp :278 ] [call:5798855410039169] emit client call state change INACTIVE, code 0 [1688635840.731|26082|sipvoiplink.cpp :901 ] [call:5798855410039169] INVITE@0xb400006f2ee19718 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ed71a48 status 180 (Ringing) [1688635840.732|26082|sipcall.cpp :1811] [call:5798855410039169] Peer ringing [1688635840.732|26082|call.cpp :247 ] [call:5798855410039169] state change 0/0, cnx 1/3, code 0 [1688635840.732|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635840.733|26082|call.cpp :278 ] [call:5798855410039169] emit client call state change INCOMING, code 0 [1688635840.734|26082|call.cpp :247 ] [call:5798855410039169] state change 0/0, cnx 3/3, code 0 [1688635840.735|26082|manager.cpp :2483] Incoming call 5798855410039169 on account 1234 with 1 media [1688635840.775|26082|call.cpp :247 ] [call:5798855410039169] state change 0/0, cnx 3/3, code 0 [1688635840.776|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635840.776|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635840.859|26082|audio_player.cpp :200 ] OpenSL playback start [1688635840.916|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635840.927|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635840.927|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635840.929|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635840.930|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635841.360|26082|sipvoiplink.cpp :783 ] username = 123456, server = 12.345.6782, from = 12.345.6768 [1688635841.430|26082|sipaccount.cpp :1528] Matching account id in request with username 123456 [1688635841.440|26082|sipvoiplink.cpp :374 ] Received a SIP INVITE request [1688635841.440|26082|sip_utils.cpp :281 ] Message headers: Record-Route: Via: SIP/2.0/UDP 12.345.6782;received=12.345.6782;branch=z9hG4bK09cc.f2f9024142c67a48eac0708bbda1f894.1 Via: SIP/2.0/UDP 12.345.6768:5060;rport=5060;received=12.345.6768;branch=z9hG4bK48a8b6db Max-Forwards: 69 From: "Anonymous" ;tag=as1c96cf3b To: Contact: Call-ID: 1171ef825bfbcf3b1a7289796b8e37c0@12.345.6768:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 06 Jul 2023 09:30:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 364 [1688635841.440|26082|sipcall.cpp :149 ] [call:2155136010844804] Create a new [INCOMING] SIP call with 1 media [1688635841.440|26082|audio_rtp_session.cpp :58 ] Created Audio RTP session: 0xb400006f3ed57580 - call Id 2155136010844804 [1688635841.440|26082|ringbuffer.cpp :55 ] Create new RingBuffer 2155136010844804 [1688635841.440|26082|sipcall.cpp :2007] [call:2155136010844804] Added media @0: type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635841.440|26082|sipcall.cpp :2010] [call:2155136010844804] Created 1 Media streams [1688635841.450|26082|sipcall.cpp :456 ] [call:2155136010844804] Setting transport to [0xb400006eced067c8] [1688635841.450|26082|sipcall.cpp :473 ] [call:2155136010844804] Crypto (SRTP) is negotiated over an un-encrypted signaling channel [1688635841.450|26082|sipcall.cpp :1706] [call:2155136010844804] Set peer's User-Agent to [Asterisk] [1688635841.450|26082|sipcall.cpp :1724] Could not find the expected package name in peer's User-Agent [1688635841.450|26082|call.cpp :247 ] [call:2155136010844804] state change 0/0, cnx 0/2, code 0 [1688635841.450|26082|call.cpp :278 ] [call:2155136010844804] emit client call state change CONNECTING, code 0 [1688635841.470|26082|sipcall.cpp :744 ] [call:2155136010844804] Set new invite session [0xb400006f2edd4fb8] [1688635841.470|26082|sip_utils.cpp :222 ] Add header to SIP message: "User-Agent: Jami Daemon 13.7.0 (android)" [1688635841.470|26082|sipvoiplink.cpp :901 ] [call:2155136010844804] INVITE@0xb400006f2edd4fb8 state changed to 2 (INCOMING): cause=0, tsx@0xb400006f4ecd6978 status 100 (Trying) [1688635841.480|26082|call.cpp :247 ] [call:2155136010844804] state change 0/0, cnx 2/1, code 0 [1688635841.480|26082|call.cpp :278 ] [call:2155136010844804] emit client call state change INACTIVE, code 0 [1688635841.490|26082|sipvoiplink.cpp :901 ] [call:2155136010844804] INVITE@0xb400006f2edd4fb8 state changed to 3 (EARLY): cause=0, tsx@0xb400006f4ecd6978 status 180 (Ringing) [1688635841.490|26082|sipcall.cpp :1811] [call:2155136010844804] Peer ringing [1688635841.490|26082|call.cpp :247 ] [call:2155136010844804] state change 0/0, cnx 1/3, code 0 [1688635841.490|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635841.490|26082|call.cpp :278 ] [call:2155136010844804] emit client call state change INCOMING, code 0 [1688635841.500|26082|call.cpp :247 ] [call:2155136010844804] state change 0/0, cnx 3/3, code 0 [1688635841.500|26082|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635841.500|26082|audio_player.cpp :239 ] OpenSL playback stop [1688635841.500|26082|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635841.510|26082|manager.cpp :2483] Incoming call 2155136010844804 on account 1234 with 1 media [1688635841.620|26069|sipcall.cpp :2224] [call:2155136010844804] Stopping all media [1688635841.620|26069|audio_rtp_session.cpp :229 ] [0xb400006f3ed57580] Stopping receiver [1688635841.620|26069|sipcall.cpp :756 ] [call:2155136010844804] Terminate SIP session [1688635841.620|26069|sipvoiplink.cpp :901 ] [call:2155136010844804] INVITE@0xb400006f2edd4fb8 state changed to 6 (DISCONNCTD): cause=486, tsx@0xb400006f4ecd6978 status 486 (Busy Here) [1688635841.630|26069|call.cpp :247 ] [call:2155136010844804] state change 0/3, cnx 3/0, code 0 [1688635841.630|26069|call.cpp :278 ] [call:2155136010844804] emit client call state change BUSY, code 0 [1688635841.218|26069|sipcall.cpp :730 ] [call:2155136010844804] Delete current invite session [1688635841.218|26069|call.cpp :247 ] [call:2155136010844804] state change 3/3, cnx 0/0, code 103 [1688635841.218|26069|sipcall.cpp :1589] [call:2155136010844804] removeCall() [1688635841.219|26069|call_factory.cpp :72 ] Removing call 2155136010844804 [1688635841.219|26069|call_factory.cpp :75 ] Remaining 1 call [1688635841.219|26069|call.cpp :247 ] [call:2155136010844804] state change 3/6, cnx 0/0, code 0 [1688635841.219|26069|call.cpp :278 ] [call:2155136010844804] emit client call state change OVER, code 0 [1688635841.220|26069|sipcall.cpp :456 ] [call:2155136010844804] Setting transport to [0x0] [1688635841.221|26069|manager.cpp :1648] [call:2155136010844804] Remove local audio [1688635841.221|26082|call.cpp :247 ] [call:2155136010844804] state change 6/6, cnx 0/3, code 0 [1688635841.221|26069|ringbufferpool.cpp :262 ] Unbind call 2155136010844804 from all bound calls [1688635841.221|26082|call.cpp :101 ] Scheduling call timeout in 30 seconds [1688635841.221|26069|manager.cpp :1098] Could not hang up non-existant call 2155136010844804 [1688635841.221|26082|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635841.221|26082|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635841.221|26079|manager.cpp :1973] [call:2155136010844804] Busy [1688635841.222|26079|sipcall.cpp :1589] [call:2155136010844804] removeCall() [1688635841.222|26079|call_factory.cpp :72 ] Removing call 2155136010844804 [1688635841.222|26079|call_factory.cpp :75 ] Remaining 1 call [1688635841.223|26079|call.cpp :247 ] [call:2155136010844804] state change 6/6, cnx 3/3, code 0 [1688635841.235|26082|audio_player.cpp :200 ] OpenSL playback start [1688635841.244|26082|opensllayer.cpp :111 ] OpenSL audio layer started [1688635841.254|26082|media_decoder.cpp :157 ] Trying to open device /data/data/cx.ring/files/ringtones/default.opus with format , pixel format , size 0x0, rate 0.000000 [1688635841.255|26082|media_decoder.cpp :174 ] Using format and resolution 0x0 [1688635841.255|26082|media_decoder.cpp :551 ] Using libopus Opus (libopus) decoder for audio [1688635841.255|26082|media_decoder.cpp :559 ] Not using hardware decoding for opus [1688635841.367|26082|audio_rtp_session.cpp :229 ] [0xb400006f3ed57580] Stopping receiver [1688635841.368|26082|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3ed57580 - call Id 2155136010844804 [1688635841.368|26082|ringbuffer.cpp :60 ] Destroy RingBuffer 2155136010844804 [1688635842.230|26069|manager.cpp :1049] Answer call 5798855410039169 [1688635842.230|26069|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 2 [1688635842.230|26069|audio_player.cpp :239 ] OpenSL playback stop [1688635842.231|26069|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635842.232|26069|sipcall.cpp :889 ] [call:5798855410039169] Answering incoming call with following media: [1688635842.233|26069|sipcall.cpp :892 ] [call:5798855410039169] Media @0 - type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES] [1688635842.233|26069|sipcall.cpp :2336] [call:5798855410039169] [audio_0] already un-muted [1688635842.233|26069|sdp.cpp :606 ] Processing received offer for [Call ID 5798855410039169] with 1 media [1688635842.234|26069|sdp.cpp :503 ] [SDP OFFER] Remote session: v=0 o=root 1072200963 1072200963 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 62800 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635842.234|26069|sdp.cpp :263 ] Add media description [type [AUDIO] enabled [YES] muted [NO] label [audio_0] source [] secure [YES]] [1688635842.234|26069|sdp.cpp :503 ] [SDP ANSWER] Local session: v=0 o=localhost 3897624642 0 IN IP4 77.119.82.20 s=Call ID 5798855410039169 c=IN IP4 77.119.82.20 t=0 0 m=audio 29260 RTP/SAVP 104 9 2 112 111 110 8 0 101 a=rtpmap:104 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:112 speex/32000 a=rtpmap:111 speex/16000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:29261 IN IP4 77.119.82.20 a=sendrecv [1688635842.234|26069|sipcall.cpp :3463] [call:5798855410039169] No ICE username fragment attribute in remote SDP [1688635842.235|26069|sipvoiplink.cpp :1121] [call:5798855410039169] INVITE@0xb400006f2ee19718 media update: status 0 [1688635842.235|26069|sdp.cpp :139 ] Set active local session to [0xb400006f2ee55028]. Was [0x0] [1688635842.235|26069|sdp.cpp :503 ] [SDP ANSWER] Local active session: v=0 o=localhost 3897624642 1 IN IP4 77.119.82.20 s=Call ID 5798855410039169 c=IN IP4 77.119.82.20 t=0 0 m=audio 0 RTP/AVP 8 111 0 3 110 101 [1688635842.235|26069|sdp.cpp :147 ] Set active remote session to [0xb400006f2ee0bb28]. Was [0x0] [1688635842.235|26069|sdp.cpp :503 ] [SDP ANSWER] Remote active session: v=0 o=root 1072200963 1072200963 IN IP4 12.345.6783 s=Asterisk c=IN IP4 12.345.6783 t=0 0 m=audio 62800 RTP/AVP 8 111 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv a=nortpproxy:yes [1688635842.235|26069|sipcall.cpp :970 ] [call:5798855410039169] Answering with contact header: [1688635842.236|26069|sipvoiplink.cpp :901 ] [call:5798855410039169] INVITE@0xb400006f2ee19718 state changed to 4 (CONNECTING): cause=0, tsx@0xb400006f4ed71a48 status 200 (OK) [1688635842.236|26069|call.cpp :247 ] [call:5798855410039169] state change 0/1, cnx 3/4, code 0 [1688635842.236|26069|call.cpp :278 ] [call:5798855410039169] emit client call state change CURRENT, code 0 [1688635842.244|26069|manager.cpp :605 ] ----- Switch current call id to '5798855410039169' ----- [1688635842.244|26069|manager.cpp :1616] Add audio to call 5798855410039169 [1688635842.244|26069|manager.cpp :1627] [call:5798855410039169] Attach audio [1688635842.244|26069|ringbufferpool.cpp :174 ] Bind call 5798855410039169 to call audiolayer_id [1688635842.244|26069|ringbufferpool.cpp :155 ] Bind rbuf '5798855410039169' to callid 'audiolayer_id' [1688635842.245|26069|ringbufferpool.cpp :155 ] Bind rbuf 'audiolayer_id' to callid '5798855410039169' [1688635842.245|26069|opensllayer.cpp :61 ] Start OpenSL audio layer [1688635842.244|26079|sipcall.cpp :2607] [call:5798855410039169] Media negotiation complete [1688635842.245|26069|audio_player.cpp :105 ] Creating OpenSL playback stream {s16, 1 channels, 48000Hz} [1688635842.245|26079|sipcall.cpp :3463] [call:5798855410039169] No ICE username fragment attribute in remote SDP [1688635842.245|26079|sipcall.cpp :2050] [call:5798855410039169] updating negotiated media [1688635842.245|26079|sipcall.cpp :2070] [call:5798855410039169] [SDP:slot#0] The media is disabled, skipping [1688635842.246|26079|sipcall.cpp :2633] [call:5798855410039169] ICE media disabled, using default media ports [1688635842.246|26079|sipcall.cpp :2224] [call:5798855410039169] Stopping all media [1688635842.246|26079|audio_rtp_session.cpp :229 ] [0xb400006f3efc3c20] Stopping receiver [1688635842.247|26079|sipcall.cpp :2146] [call:5798855410039169] Starting all media [1688635842.247|26079|sipcall.cpp :2155] [call:5798855410039169] Crypto (SRTP) is negotiated over an insecure signaling transport [1688635842.247|26079|audio_rtp_session.cpp :229 ] [0xb400006f3efc3c20] Stopping receiver [1688635842.248|26079|sipcall.cpp :2259] [call:5798855410039169] Updating remote media [1688635842.307|26082|sipvoiplink.cpp :901 ] [call:5798855410039169] INVITE@0xb400006f2ee19718 state changed to 6 (DISCONNCTD): cause=200, tsx@0xb400006f4ed6d098 status 200 (OK) [1688635842.308|26079|manager.cpp :1951] [call:5798855410039169] Peer hung up [1688635842.308|26079|sipcall.cpp :2224] [call:5798855410039169] Stopping all media [1688635842.308|26079|audio_rtp_session.cpp :229 ] [0xb400006f3efc3c20] Stopping receiver [1688635842.308|26079|sipcall.cpp :756 ] [call:5798855410039169] Terminate SIP session [1688635842.309|26079|sipcall.cpp :730 ] [call:5798855410039169] Delete current invite session [1688635842.309|26079|call.cpp :247 ] [call:5798855410039169] state change 1/1, cnx 4/0, code 103 [1688635842.309|26079|call.cpp :278 ] [call:5798855410039169] emit client call state change HUNGUP, code 103 [1688635842.447|26069|audio_player.cpp :200 ] OpenSL playback start [1688635842.494|26069|opensllayer.cpp :111 ] OpenSL audio layer started [1688635842.507|26079|manager.cpp :1648] [call:5798855410039169] Remove local audio [1688635842.507|26079|ringbufferpool.cpp :262 ] Unbind call 5798855410039169 from all bound calls [1688635842.507|26079|opensllayer.cpp :121 ] Stopping OpenSL audio layer for type 0 [1688635842.507|26079|audio_player.cpp :239 ] OpenSL playback stop [1688635842.511|26079|audio_player.cpp :176 ] Destroying OpenSL playback stream [1688635842.520|26079|sipcall.cpp :1589] [call:5798855410039169] removeCall() [1688635842.520|26079|sdp.cpp :139 ] Set active local session to [0x0]. Was [0xb400006f2ee55028] [1688635842.520|26079|call_factory.cpp :72 ] Removing call 5798855410039169 [1688635842.520|26079|call_factory.cpp :75 ] Remaining 0 call [1688635842.520|26079|call.cpp :247 ] [call:5798855410039169] state change 1/6, cnx 0/0, code 0 [1688635842.520|26079|call.cpp :278 ] [call:5798855410039169] emit client call state change OVER, code 0 [1688635842.521|26079|sipcall.cpp :456 ] [call:5798855410039169] Setting transport to [0x0] [1688635842.521|26079|audio_rtp_session.cpp :229 ] [0xb400006f3efc3c20] Stopping receiver [1688635842.522|26079|audio_rtp_session.cpp :68 ] Destroyed Audio RTP session: 0xb400006f3efc3c20 - call Id 5798855410039169 [1688635842.522|26079|ringbuffer.cpp :60 ] Destroy RingBuffer 5798855410039169added BacklogMedium label
Maybe this patch (also) introduced some problems?
One additional question:
I tried to receive a call using phone 1 (SIM-card) -> phone 2 (SIP-account using Jami). If you use a SIM to make a call you shouldn't be able to answer the call with video.
But if you make a call SIP-Account 1 (with video) -> SIP Account 2 (using Jami) should you get the options "Decline", "Answer in audio" and "Answer in video"?
If yes, can you answer the call if you tap "Answer in video" (instead of audio)? Because I only saw the options "Decline" and "Answer in audio" - but I only got one 1SIP-account.
Edited by Elys
assigned to @pnicolas
removed BacklogMedium label
added BacklogHigh label
added SprintIn progress label
Scenario 1: Two devices Alice and Bob, both on QT, connected with SIP account : Alice call Bob, who can answer (OK).
Scenario 2: Bob move on Android : Alice call Bob, Bob try to answer but it doesn't work (KO).
Edited by Pierre NicolasIf both - Alice and Bob's SIP provider support video - can you answer the call in video? Android and/or Desktop?
(I was only able to test "Answer in audio" since SIM-cards (call: SIM -> SIP (using Jami)) don't support video?)
Edited by Elys
added SprintTo review label and removed SprintIn progress label
mentioned in commit 94f816d3
added SprintDone label and removed SprintTo review label
marked this issue as related to #1314
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