PJSIP Logging enabled <--- Received SIP request (862 bytes) from UDP:84.8.1.2:5061 ---> INVITE sip:12345@asterisktest.com SIP/2.0 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 CSeq: 1 INVITE From: "desktop" ;tag=2f6080b7 To: Via: SIP/2.0/UDP 84.8.1.2:5061;branch=z9hG4bK-333239-0e235e95104dc687bd943c0d98159341 Max-Forwards: 70 Contact: "desktop" User-Agent: Jitsi2.10.5550Linux Content-Type: application/sdp Content-Length: 308 v=0 o=desktop-jitsi.org 0 0 IN IP4 84.8.1.2 s=- c=IN IP4 84.8.1.2 t=0 0 m=audio 5042 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-xr:voip-metrics == Setting global variable 'SIPDOMAIN' to 'asterisktest.com' <--- Transmitting SIP response (389 bytes) to UDP:84.8.1.2:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.8.1.2:5061;rport=5061;received=84.8.1.2;branch=z9hG4bK-333239-0e235e95104dc687bd943c0d98159341 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 From: "desktop" ;tag=2f6080b7 To: CSeq: 1 INVITE Server: Asterisk PBX 15.0.0 Content-Length: 0 -- Executing [12345@from-proxy:1] Dial("PJSIP/desktop-00000002", "PJSIP/laptop") in new stack <--- Transmitting SIP request (1065 bytes) to UDP:84.8.1.2:15060 ---> INVITE sip:laptop@84.8.1.2:15060;transport=udp;registering_acc=asterisktest_iperitydev_com SIP/2.0 Via: SIP/2.0/UDP 178.6.1.2:5060;rport;branch=z9hG4bKPje9ffa8b0-31fd-4b07-915d-d4446a17145c From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: Contact: Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea CSeq: 13842 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 15.0.0 Content-Type: application/sdp Content-Length: 287 v=0 o=- 551735915 551735915 IN IP4 178.6.1.2 s=Asterisk c=IN IP4 178.6.1.2 t=0 0 m=audio 34562 RTP/AVP 9 8 0 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- Called PJSIP/laptop <--- Received SIP response (496 bytes) from UDP:84.8.1.2:15060 ---> SIP/2.0 180 Ringing CSeq: 13842 INVITE Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: ;tag=df7436f3 Via: SIP/2.0/UDP 178.6.1.2:5060;rport=5060;branch=z9hG4bKPje9ffa8b0-31fd-4b07-915d-d4446a17145c;received=178.6.1.2 Contact: "laptop" User-Agent: Jitsi2.11.5568Linux Content-Length: 0 -- PJSIP/laptop-00000003 is ringing <--- Transmitting SIP response (578 bytes) to UDP:84.8.1.2:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.8.1.2:5061;rport=5061;received=84.8.1.2;branch=z9hG4bK-333239-0e235e95104dc687bd943c0d98159341 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 From: "desktop" ;tag=2f6080b7 To: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 CSeq: 1 INVITE Server: Asterisk PBX 15.0.0 Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Content-Length: 0 <--- Received SIP response (496 bytes) from UDP:84.8.1.2:15060 ---> SIP/2.0 180 Ringing CSeq: 13842 INVITE Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: ;tag=df7436f3 Via: SIP/2.0/UDP 178.6.1.2:5060;rport=5060;branch=z9hG4bKPje9ffa8b0-31fd-4b07-915d-d4446a17145c;received=178.6.1.2 Contact: "laptop" User-Agent: Jitsi2.11.5568Linux Content-Length: 0 -- PJSIP/laptop-00000003 is ringing <--- Received SIP request (644 bytes) from UDP:84.8.1.2:15060 ---> OPTIONS sip:asterisktest.com SIP/2.0 Call-ID: 36d74c8ca60312392fb9941361ff7ec1@0:0:0:0:0:0:0:0 CSeq: 50 OPTIONS From: "laptop" ;tag=ebb0c352 To: "laptop" Via: SIP/2.0/UDP 84.8.1.2:15060;branch=z9hG4bK-383531-744816eff381f1cbdcbd1e8f8d934cee Max-Forwards: 70 Contact: "laptop" User-Agent: Jitsi2.11.5568Linux Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary Content-Length: 0 <--- Transmitting SIP response (936 bytes) to UDP:84.8.1.2:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.1.2:15060;rport=15060;received=84.8.1.2;branch=z9hG4bK-383531-744816eff381f1cbdcbd1e8f8d934cee Call-ID: 36d74c8ca60312392fb9941361ff7ec1@0:0:0:0:0:0:0:0 From: "laptop" ;tag=ebb0c352 To: "laptop" ;tag=z9hG4bK-383531-744816eff381f1cbdcbd1e8f8d934cee CSeq: 50 OPTIONS Accept: application/sdp, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Accept-Encoding: text/plain Accept-Language: en Server: Asterisk PBX 15.0.0 Content-Length: 0 <--- Received SIP response (496 bytes) from UDP:84.8.1.2:15060 ---> SIP/2.0 180 Ringing CSeq: 13842 INVITE Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: ;tag=df7436f3 Via: SIP/2.0/UDP 178.6.1.2:5060;rport=5060;branch=z9hG4bKPje9ffa8b0-31fd-4b07-915d-d4446a17145c;received=178.6.1.2 Contact: "laptop" User-Agent: Jitsi2.11.5568Linux Content-Length: 0 -- PJSIP/laptop-00000003 is ringing <--- Received SIP response (693 bytes) from UDP:84.8.1.2:15060 ---> SIP/2.0 200 OK CSeq: 13842 INVITE Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: ;tag=df7436f3 Via: SIP/2.0/UDP 178.6.1.2:5060;rport=5060;branch=z9hG4bKPje9ffa8b0-31fd-4b07-915d-d4446a17145c;received=178.6.1.2 Contact: "laptop" User-Agent: Jitsi2.11.5568Linux Content-Type: application/sdp Content-Length: 169 v=0 o=laptop-jitsi.org 0 0 IN IP4 192.168.2.27 s=- c=IN IP4 192.168.2.27 t=0 0 m=audio 5030 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <--- Transmitting SIP request (454 bytes) to UDP:84.8.1.2:15060 ---> ACK sip:laptop@84.8.1.2:15060 SIP/2.0 Via: SIP/2.0/UDP 178.6.1.2:5060;rport;branch=z9hG4bKPj151a3efb-f3a5-4df1-b1f8-430828f92e90 From: "desktop" ;tag=ca2ba3c4-c893-495a-b309-d01c49a93f1e To: ;tag=df7436f3 Call-ID: ed5ccda6-9bc6-48eb-85fd-14e92cb8e6ea CSeq: 13842 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 15.0.0 Content-Length: 0 -- PJSIP/laptop-00000003 answered PJSIP/desktop-00000002 <--- Transmitting SIP response (878 bytes) to UDP:84.8.1.2:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.1.2:5061;rport=5061;received=84.8.1.2;branch=z9hG4bK-333239-0e235e95104dc687bd943c0d98159341 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 From: "desktop" ;tag=2f6080b7 To: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 CSeq: 1 INVITE Server: Asterisk PBX 15.0.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Contact: Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 223 v=0 o=- 0 2 IN IP4 178.6.1.2 s=Asterisk c=IN IP4 178.6.1.2 t=0 0 m=audio 16460 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- Channel PJSIP/laptop-00000003 joined 'simple_bridge' basic-bridge <3d5407c7-f048-4331-ba7e-8fbbed229167> -- Channel PJSIP/desktop-00000002 joined 'simple_bridge' basic-bridge <3d5407c7-f048-4331-ba7e-8fbbed229167> <--- Received SIP request (542 bytes) from UDP:84.8.1.2:5061 ---> ACK sip:178.6.1.2:5060 SIP/2.0 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 CSeq: 1 ACK Via: SIP/2.0/UDP 84.8.1.2:5061;branch=z9hG4bK-333239-467e76e3c3c2c1a3d35a2cad907f0a7a From: "desktop" ;tag=2f6080b7 To: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 Max-Forwards: 70 Contact: "desktop" User-Agent: Jitsi2.10.5550Linux Content-Length: 0 <--- Transmitting SIP request (986 bytes) to UDP:84.8.1.2:5061 ---> INVITE sip:desktop@84.8.1.2:5061;transport=udp;registering_acc=asterisktest_iperitydev_com SIP/2.0 Via: SIP/2.0/UDP 178.6.1.2:5060;rport;branch=z9hG4bKPjb94043fa-d3b7-4d06-b1cd-d9f6aaed6c4e From: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 To: "desktop" ;tag=2f6080b7 Contact: Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 CSeq: 30517 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 15.0.0 Content-Type: application/sdp Content-Length: 223 v=0 o=- 0 3 IN IP4 178.6.1.2 s=Asterisk c=IN IP4 178.6.1.2 t=0 0 m=audio 16460 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (550 bytes) from UDP:84.8.1.2:5061 ---> SIP/2.0 500 Server Internal Error CSeq: 30517 INVITE Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 From: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 To: "desktop" ;tag=2f6080b7 Via: SIP/2.0/UDP 178.6.1.2:5060;rport=5060;branch=z9hG4bKPjb94043fa-d3b7-4d06-b1cd-d9f6aaed6c4e;received=178.6.1.2 Contact: "desktop" User-Agent: Jitsi2.10.5550Linux Content-Length: 0 <--- Transmitting SIP request (493 bytes) to UDP:84.8.1.2:5061 ---> ACK sip:desktop@84.8.1.2:5061;transport=udp;registering_acc=asterisktest_iperitydev_com SIP/2.0 Via: SIP/2.0/UDP 178.6.1.2:5060;rport;branch=z9hG4bKPjb94043fa-d3b7-4d06-b1cd-d9f6aaed6c4e From: ;tag=b322b869-4d16-4e1e-a45b-7dc9c6888f65 To: "desktop" ;tag=2f6080b7 Call-ID: dcd70fbf15ab56e26c60bb2b8cf4eb39@0:0:0:0:0:0:0:0 CSeq: 30517 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 15.0.0 Content-Length: 0 <--- Received SIP request (706 bytes) from UDP:84.8.1.2:5061 ---> OPTIONS sip:asterisktest.com SIP/2.0 Call-ID: d31607a6c01240edcbd60adf87e63c86@0:0:0:0:0:0:0:0 CSeq: 51 OPTIONS From: "desktop" ;tag=b764c0a7 To: "desktop" Via: SIP/2.0/UDP 84.8.1.2:5061;branch=z9hG4bK-333239-297a9c606ae378424e3c679c72624b8f Max-Forwards: 70 Contact: "desktop" User-Agent: Jitsi2.10.5550Linux Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary Content-Length: 0 <--- Transmitting SIP response (938 bytes) to UDP:84.8.1.2:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.8.1.2:5061;rport=5061;received=84.8.1.2;branch=z9hG4bK-333239-297a9c606ae378424e3c679c72624b8f Call-ID: d31607a6c01240edcbd60adf87e63c86@0:0:0:0:0:0:0:0 From: "desktop" ;tag=b764c0a7 To: "desktop" ;tag=z9hG4bK-333239-297a9c606ae378424e3c679c72624b8f CSeq: 51 OPTIONS Accept: application/sdp, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Accept-Encoding: text/plain Accept-Language: en Server: Asterisk PBX 15.0.0 Content-Length: 0 asterisktest*CLI>