<--- Received SIP request (563 bytes) from UDP:127.0.0.1:6789 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPjc61f170c-89bd-4fbb-8652-b537134f9f68 Route: Max-Forwards: 70 From: ;tag=ed1480e5-75be-47cc-9cb3-44dac9cc0573 To: Call-ID: cefc7bd0-99db-4d85-b16a-ad0634d841b1 CSeq: 32716 REGISTER User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <--- Transmitting SIP response (554 bytes) to UDP:127.0.0.1:6789 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:6789;rport=6789;received=127.0.0.1;branch=z9hG4bKPjc61f170c-89bd-4fbb-8652-b537134f9f68 Call-ID: cefc7bd0-99db-4d85-b16a-ad0634d841b1 From: ;tag=ed1480e5-75be-47cc-9cb3-44dac9cc0573 To: ;tag=z9hG4bKPjc61f170c-89bd-4fbb-8652-b537134f9f68 CSeq: 32716 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1487847165/6d8f33adfdf90e7a23c80ff758a9af65",opaque="31fdfe4625d25a30",algorithm=md5,qop="auth" Server: Asterisk PBX 14.2.1 Content-Length: 0 <--- Received SIP request (851 bytes) from UDP:127.0.0.1:6789 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPjb3287351-e010-4b68-b9ac-1133777ceaee Route: Max-Forwards: 70 From: ;tag=ed1480e5-75be-47cc-9cb3-44dac9cc0573 To: Call-ID: cefc7bd0-99db-4d85-b16a-ad0634d841b1 CSeq: 32717 REGISTER User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1000", realm="asterisk", nonce="1487847165/6d8f33adfdf90e7a23c80ff758a9af65", uri="sip:127.0.0.1", response="a23b55bec9d75ff547785f886d55b48b", algorithm=md5, cnonce="bc0cbdb3-2dc4-4dbf-a170-a29572e4ffcd", opaque="31fdfe4625d25a30", qop=auth, nc=00000001 Content-Length: 0 -- Added contact 'sip:1000@127.0.0.1:6789;ob' to AOR '1000' with expiration of 300 seconds == Contact 1000/sip:1000@127.0.0.1:6789;ob has been created <--- Transmitting SIP response (499 bytes) to UDP:127.0.0.1:6789 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:6789;rport=6789;received=127.0.0.1;branch=z9hG4bKPjb3287351-e010-4b68-b9ac-1133777ceaee Call-ID: cefc7bd0-99db-4d85-b16a-ad0634d841b1 From: ;tag=ed1480e5-75be-47cc-9cb3-44dac9cc0573 To: ;tag=z9hG4bKPjb3287351-e010-4b68-b9ac-1133777ceaee CSeq: 32717 REGISTER Date: Thu, 23 Feb 2017 10:52:45 GMT Contact: ;expires=299 Expires: 300 Server: Asterisk PBX 14.2.1 Content-Length: 0 == Endpoint 1000 is now Reachable -- Contact 1000/sip:1000@127.0.0.1:6789;ob is now Unknown. RTT: 0.000 msec <--- Received SIP request (563 bytes) from UDP:127.0.0.1:4789 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:4789;rport;branch=z9hG4bKPja172c9c5-a64d-4744-82a6-3178645abe95 Route: Max-Forwards: 70 From: ;tag=9e0eb6ca-1fb9-40a9-871b-74ff55bf78e3 To: Call-ID: 52185920-a4ff-4864-9b62-5eb379f5ce49 CSeq: 59012 REGISTER User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <--- Transmitting SIP response (554 bytes) to UDP:127.0.0.1:4789 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:4789;rport=4789;received=127.0.0.1;branch=z9hG4bKPja172c9c5-a64d-4744-82a6-3178645abe95 Call-ID: 52185920-a4ff-4864-9b62-5eb379f5ce49 From: ;tag=9e0eb6ca-1fb9-40a9-871b-74ff55bf78e3 To: ;tag=z9hG4bKPja172c9c5-a64d-4744-82a6-3178645abe95 CSeq: 59012 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1487847176/d2d8df6dfe0e978678a11a0e5a7af208",opaque="73df6d1b7a733eb2",algorithm=md5,qop="auth" Server: Asterisk PBX 14.2.1 Content-Length: 0 <--- Received SIP request (851 bytes) from UDP:127.0.0.1:4789 ---> REGISTER sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:4789;rport;branch=z9hG4bKPjaf470eb0-a85f-4f25-8bc1-1d5d8fb06c80 Route: Max-Forwards: 70 From: ;tag=9e0eb6ca-1fb9-40a9-871b-74ff55bf78e3 To: Call-ID: 52185920-a4ff-4864-9b62-5eb379f5ce49 CSeq: 59013 REGISTER User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Contact: Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1001", realm="asterisk", nonce="1487847176/d2d8df6dfe0e978678a11a0e5a7af208", uri="sip:127.0.0.1", response="665417975cd313062d9274c6b468e1ab", algorithm=md5, cnonce="70864eba-e35a-4c41-a5e2-ced716f6c818", opaque="73df6d1b7a733eb2", qop=auth, nc=00000001 Content-Length: 0 -- Added contact 'sip:1001@127.0.0.1:4789;ob' to AOR '1001' with expiration of 300 seconds <--- Transmitting SIP response (499 bytes) to UDP:127.0.0.1:4789 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:4789;rport=4789;received=127.0.0.1;branch=z9hG4bKPjaf470eb0-a85f-4f25-8bc1-1d5d8fb06c80 Call-ID: 52185920-a4ff-4864-9b62-5eb379f5ce49 From: ;tag=9e0eb6ca-1fb9-40a9-871b-74ff55bf78e3 To: ;tag=z9hG4bKPjaf470eb0-a85f-4f25-8bc1-1d5d8fb06c80 CSeq: 59013 REGISTER Date: Thu, 23 Feb 2017 10:52:56 GMT Contact: ;expires=299 Expires: 300 Server: Asterisk PBX 14.2.1 Content-Length: 0 == Contact 1001/sip:1001@127.0.0.1:4789;ob has been created == Endpoint 1001 is now Reachable -- Contact 1001/sip:1001@127.0.0.1:4789;ob is now Unknown. RTT: 0.000 msec <--- Received SIP request (1184 bytes) from UDP:127.0.0.1:6789 ---> INVITE sip:1001@asterisk SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPjf9abc2ee-b544-4468-9115-be401d1d5bcb Max-Forwards: 70 From: sip:1000@127.0.0.1;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: sip:1001@asterisk Contact: Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 CSeq: 24363 INVITE Route: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Content-Type: application/sdp Content-Length: 523 v=0 o=- 3696835979 3696835979 IN IP4 10.0.2.15 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 7803 RTP/AVP 98 97 99 104 3 0 8 9 120 96 c=IN IP4 10.0.2.15 b=TIAS:96000 a=rtcp:7804 IN IP4 10.0.2.15 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 <--- Transmitting SIP response (551 bytes) to UDP:127.0.0.1:6789 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:6789;rport=6789;received=127.0.0.1;branch=z9hG4bKPjf9abc2ee-b544-4468-9115-be401d1d5bcb Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 From: ;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: ;tag=z9hG4bKPjf9abc2ee-b544-4468-9115-be401d1d5bcb CSeq: 24363 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1487847179/f018f75505a964d05973d83fc8d22c61",opaque="54978d7b597ee933",algorithm=md5,qop="auth" Server: Asterisk PBX 14.2.1 Content-Length: 0 <--- Received SIP request (394 bytes) from UDP:127.0.0.1:6789 ---> ACK sip:1001@asterisk SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPjf9abc2ee-b544-4468-9115-be401d1d5bcb Max-Forwards: 70 From: sip:1000@127.0.0.1;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: sip:1001@asterisk;tag=z9hG4bKPjf9abc2ee-b544-4468-9115-be401d1d5bcb Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 CSeq: 24363 ACK Route: Content-Length: 0 <--- Received SIP request (1476 bytes) from UDP:127.0.0.1:6789 ---> INVITE sip:1001@asterisk SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPj1e0b09a2-8f02-43df-ae2d-1ffc15ee3359 Max-Forwards: 70 From: sip:1000@127.0.0.1;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: sip:1001@asterisk Contact: Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 CSeq: 24364 INVITE Route: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.6 Linux-4.4.0.64/x86_64/glibc-2.23 Authorization: Digest username="1000", realm="asterisk", nonce="1487847179/f018f75505a964d05973d83fc8d22c61", uri="sip:1001@asterisk", response="2ad38c51a094d21613a556a224a2e157", algorithm=md5, cnonce="d2ade312-6c05-413f-ba3c-6c2f6b7ec16e", opaque="54978d7b597ee933", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 523 v=0 o=- 3696835979 3696835979 IN IP4 10.0.2.15 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 7803 RTP/AVP 98 97 99 104 3 0 8 9 120 96 c=IN IP4 10.0.2.15 b=TIAS:96000 a=rtcp:7804 IN IP4 10.0.2.15 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 <--- Transmitting SIP response (348 bytes) to UDP:127.0.0.1:6789 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:6789;rport=6789;received=127.0.0.1;branch=z9hG4bKPj1e0b09a2-8f02-43df-ae2d-1ffc15ee3359 Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 From: ;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: CSeq: 24364 INVITE Server: Asterisk PBX 14.2.1 Content-Length: 0 -- Executing [1001@internal:1] Verbose("PJSIP/1000-00000000", "1, Call for Extension ") in new stack Call for Extension -- Executing [1001@internal:2] Dial("PJSIP/1000-00000000", "PJSIP/1001") in new stack -- Called PJSIP/1001 <--- Transmitting SIP request (872 bytes) to UDP:127.0.0.1:4789 ---> INVITE sip:1001@127.0.0.1:4789;ob SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj49c7f8a1-266e-4cf7-aa92-a47f3e88d02f From: ;tag=12756dd3-a016-48de-809f-99b786ed3fb9 To: Contact: Call-ID: 9b048690-460d-49d9-8d9b-7c17244105d5 CSeq: 18200 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 14.2.1 Content-Type: application/sdp Content-Length: 231 v=0 o=- 1883625038 1883625038 IN IP4 127.0.0.1 s=Asterisk c=IN IP4 127.0.0.1 t=0 0 m=audio 29690 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (323 bytes) from UDP:127.0.0.1:4789 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;received=127.0.0.1;branch=z9hG4bKPj49c7f8a1-266e-4cf7-aa92-a47f3e88d02f Call-ID: 9b048690-460d-49d9-8d9b-7c17244105d5 From: ;tag=12756dd3-a016-48de-809f-99b786ed3fb9 To: CSeq: 18200 INVITE Content-Length: 0 <--- Received SIP response (898 bytes) from UDP:127.0.0.1:4789 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;received=127.0.0.1;branch=z9hG4bKPj49c7f8a1-266e-4cf7-aa92-a47f3e88d02f Call-ID: 9b048690-460d-49d9-8d9b-7c17244105d5 From: ;tag=12756dd3-a016-48de-809f-99b786ed3fb9 To: ;tag=501a643c-cd20-4696-b1a0-fa51f73efd54 CSeq: 18200 INVITE Contact: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 266 v=0 o=- 3696835979 3696835980 IN IP4 10.0.2.15 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 5805 RTP/AVP 0 101 c=IN IP4 10.0.2.15 b=TIAS:96000 a=rtcp:5806 IN IP4 10.0.2.15 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <--- Transmitting SIP request (408 bytes) to UDP:127.0.0.1:4789 ---> ACK sip:1001@127.0.0.1:4789;ob SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPjf171ead7-8f00-4cb9-8178-2058da758cfa From: ;tag=12756dd3-a016-48de-809f-99b786ed3fb9 To: ;tag=501a643c-cd20-4696-b1a0-fa51f73efd54 Call-ID: 9b048690-460d-49d9-8d9b-7c17244105d5 CSeq: 18200 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 -- PJSIP/1001-00000001 answered PJSIP/1000-00000000 <--- Transmitting SIP response (896 bytes) to UDP:127.0.0.1:6789 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:6789;rport=6789;received=127.0.0.1;branch=z9hG4bKPj1e0b09a2-8f02-43df-ae2d-1ffc15ee3359 Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 From: ;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: ;tag=6d063129-a767-49fe-ae26-896c1c2ee3b1 CSeq: 24364 INVITE Server: Asterisk PBX 14.2.1 Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 234 v=0 o=- 3696835979 3696835981 IN IP4 127.0.0.1 s=Asterisk c=IN IP4 127.0.0.1 t=0 0 m=audio 24342 RTP/AVP 120 96 a=rtpmap:120 opus/48000/2 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:20 a=sendrecv -- Channel PJSIP/1001-00000001 joined 'simple_bridge' basic-bridge -- Channel PJSIP/1000-00000000 joined 'simple_bridge' basic-bridge > 0x7f6eb0024a90 -- Probation passed - setting RTP source address to 127.0.0.1:5805 Got RTP packet from 127.0.0.1:5805 (type 00, seq 026419, ts 000160, len 000160) <--- Received SIP request (359 bytes) from UDP:127.0.0.1:6789 ---> ACK sip:127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:6789;rport;branch=z9hG4bKPj155bccf2-e7f6-4c22-836f-01edb11b4238 Max-Forwards: 70 From: sip:1000@127.0.0.1;tag=848ade1d-c97b-4b9e-8ed9-0934f7c4b024 To: sip:1001@asterisk;tag=6d063129-a767-49fe-ae26-896c1c2ee3b1 Call-ID: 3afa4b99-c80d-4367-aaa9-43a0c6fdabb0 CSeq: 24364 ACK Content-Length: 0 Segmentation fault (core dumped)