[Jan 22 18:10:25] Asterisk 13.7.0 built by root @ debian1 on a i686 running Linux on 2016-01-21 14:51:32 UTC [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP request (873 bytes) from UDP:192.168.1.13:5060 ---> INVITE sip:123@192.168.1.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK611163411562530295;rport From: ALICE ;tag=672914266 To: "123" Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 1 INVITE Contact: Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 339 v=0 o=BOB 26803467 17124300 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10018 RTP/AVP 0 8 18 4 2 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP response (515 bytes) to UDP:192.168.1.13:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.13:5060;rport=5060;received=192.168.1.13;branch=z9hG4bK611163411562530295 Call-ID: 139211552013260-127592499222417@192.168.1.13 From: "ALICE" ;tag=672914266 To: "123" ;tag=z9hG4bK611163411562530295 CSeq: 1 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1453507830/809a83164d9dffcc7a7f86cc19d66070",opaque="5b08c4672aacb1ab",algorithm=md5,qop="auth" Server: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP request (334 bytes) from UDP:192.168.1.13:5060 ---> ACK sip:123@192.168.1.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK611163411562530295;rport From: "ALICE" ;tag=672914266 To: "123" ;tag=z9hG4bK611163411562530295 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP request (1142 bytes) from UDP:192.168.1.13:5060 ---> INVITE sip:123@192.168.1.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK2526050211618511208;rport From: ALICE ;tag=672914266 To: "123" Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 2 INVITE Contact: Authorization: Digest username="ALICE", realm="asterisk", nonce="1453507830/809a83164d9dffcc7a7f86cc19d66070", uri="sip:123@192.168.1.24", response="08f6ed327df560c1b27c4f2397b456f9", algorithm=MD5, cnonce="ebcb82d2", opaque="5b08c4672aacb1ab", qop=auth, nc=00000001 Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 339 v=0 o=BOB 26803467 17124300 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10018 RTP/AVP 0 8 18 4 2 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [Jan 22 18:10:30] VERBOSE[3199] res_pjsip_logger.c: <--- Transmitting SIP response (333 bytes) to UDP:192.168.1.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.13:5060;rport=5060;received=192.168.1.13;branch=z9hG4bK2526050211618511208 Call-ID: 139211552013260-127592499222417@192.168.1.13 From: "ALICE" ;tag=672914266 To: "123" CSeq: 2 INVITE Server: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (913 bytes) to UDP:192.168.1.21:5060 ---> INVITE sip:123@192.168.1.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj89cfbb1a-1ce0-4479-a2d7-f7f47653577d From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: Contact: Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15738 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Type: application/sdp Content-Length: 261 v=0 o=- 1678942029 1678942029 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.24 t=0 0 m=audio 11676 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (567 bytes) from UDP:192.168.1.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bKPj89cfbb1a-1ce0-4479-a2d7-f7f47653577d;received=192.168.1.24;rport=5060 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15738 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (884 bytes) from UDP:192.168.1.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bKPj89cfbb1a-1ce0-4479-a2d7-f7f47653577d;received=192.168.1.24;rport=5060 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15738 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 243 v=0 o=root 1043598828 1043598828 IN IP4 192.168.1.21 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 192.168.1.21 t=0 0 m=audio 11080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (870 bytes) from UDP:192.168.1.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bKPj89cfbb1a-1ce0-4479-a2d7-f7f47653577d;received=192.168.1.24;rport=5060 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15738 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 243 v=0 o=root 1043598828 1043598828 IN IP4 192.168.1.21 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 192.168.1.21 t=0 0 m=audio 11080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (395 bytes) to UDP:192.168.1.21:5060 ---> ACK sip:123@192.168.1.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPjc99bbd27-d0a0-4506-abbf-10a0822d63f1 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15738 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP response (796 bytes) to UDP:192.168.1.13:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.13:5060;rport=5060;received=192.168.1.13;branch=z9hG4bK2526050211618511208 Call-ID: 139211552013260-127592499222417@192.168.1.13 From: "ALICE" ;tag=672914266 To: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 CSeq: 2 INVITE Server: Asterisk PBX 13.7.0 Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Content-Type: application/sdp Content-Length: 233 v=0 o=- 26803467 17124302 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.24 t=0 0 m=audio 15888 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 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP response (830 bytes) to UDP:192.168.1.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.13:5060;rport=5060;received=192.168.1.13;branch=z9hG4bK2526050211618511208 Call-ID: 139211552013260-127592499222417@192.168.1.13 From: "ALICE" ;tag=672914266 To: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 CSeq: 2 INVITE Server: Asterisk PBX 13.7.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Contact: Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 233 v=0 o=- 26803467 17124302 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.24 t=0 0 m=audio 15888 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 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (923 bytes) to UDP:192.168.1.21:5060 ---> INVITE sip:123@192.168.1.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj02f9aba4-a061-41a6-b81b-dd3c997b84f8 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Contact: Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15739 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uas Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Type: application/sdp Content-Length: 237 v=0 o=- 1678942029 1678942030 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.13 t=0 0 m=audio 10018 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 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (557 bytes) from UDP:192.168.1.21:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.1.24:5060;branch=z9hG4bKPj02f9aba4-a061-41a6-b81b-dd3c997b84f8;received=192.168.1.24;rport=5060 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15739 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (395 bytes) to UDP:192.168.1.21:5060 ---> ACK sip:123@192.168.1.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj02f9aba4-a061-41a6-b81b-dd3c997b84f8 From: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a To: ;tag=as7ef884fe Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 15739 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP request (372 bytes) from UDP:192.168.1.13:5060 ---> ACK sip:192.168.1.24:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK1057481961150924835 From: ALICE ;tag=672914266 To: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (911 bytes) to UDP:192.168.1.13:5060 ---> INVITE sip:ALICE@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj669d4bf8-4748-4af0-8e1a-0f80646c0755 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Contact: Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24771 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Type: application/sdp Content-Length: 233 v=0 o=- 26803467 17124303 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.21 t=0 0 m=audio 11080 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 [Jan 22 18:10:30] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (770 bytes) from UDP:192.168.1.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj669d4bf8-4748-4af0-8e1a-0f80646c0755 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24771 INVITE Contact: Supported: 100rel, replaces, timer Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 214 v=0 o=BOB 69312114 16415161 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10018 RTP/AVP 9 101 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [Jan 22 18:10:30] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (410 bytes) to UDP:192.168.1.13:5060 ---> ACK sip:ALICE@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj6c585bf6-3098-4c32-b6ab-13ced19aa4c0 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24771 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:59] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:192.168.1.21:5060 ---> BYE sip:asterisk@192.168.1.24:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK5012e50a;rport Max-Forwards: 70 From: ;tag=as7ef884fe To: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a Call-ID: 454768bc-cf90-457a-97f5-078522346296 CSeq: 102 BYE User-Agent: Asterisk PBX 13.7.0-rc3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 [Jan 22 18:10:59] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP response (345 bytes) to UDP:192.168.1.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.21:5060;rport=5060;received=192.168.1.21;branch=z9hG4bK5012e50a Call-ID: 454768bc-cf90-457a-97f5-078522346296 From: ;tag=as7ef884fe To: "ALICE" ;tag=16a56f3b-3363-44d1-8c6c-d6ae87e8095a CSeq: 102 BYE Server: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:59] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (911 bytes) to UDP:192.168.1.13:5060 ---> INVITE sip:ALICE@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj909335b1-bcb1-4344-a5de-1de0277122f2 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Contact: Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24772 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Type: application/sdp Content-Length: 233 v=0 o=- 26803467 17124304 IN IP4 192.168.1.24 s=Asterisk c=IN IP4 192.168.1.24 t=0 0 m=audio 15888 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 [Jan 22 18:10:59] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (770 bytes) from UDP:192.168.1.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj909335b1-bcb1-4344-a5de-1de0277122f2 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24772 INVITE Contact: Supported: 100rel, replaces, timer Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 214 v=0 o=BOB 13322969 17888230 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10018 RTP/AVP 9 101 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [Jan 22 18:10:59] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (410 bytes) to UDP:192.168.1.13:5060 ---> ACK sip:ALICE@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj00b9e693-a244-4e3b-8141-50b872fa160b From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24772 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:59] VERBOSE[3198] res_pjsip_logger.c: <--- Transmitting SIP request (410 bytes) to UDP:192.168.1.13:5060 ---> BYE sip:ALICE@192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj7eab9676-817a-49b9-a1d1-1931e8708d07 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24773 BYE Max-Forwards: 70 User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 [Jan 22 18:10:59] VERBOSE[3203] res_pjsip_logger.c: <--- Received SIP response (357 bytes) from UDP:192.168.1.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj7eab9676-817a-49b9-a1d1-1931e8708d07 From: "123" ;tag=34cd55aa-9df1-48ce-a2dd-12b9311243f8 To: "ALICE" ;tag=672914266 Call-ID: 139211552013260-127592499222417@192.168.1.13 CSeq: 24773 BYE Server: Voip Phone 1.0 Content-Length: 0 [Jan 22 18:11:05] VERBOSE[3251] asterisk.c: Asterisk cleanly ending (0). [Jan 22 18:11:05] VERBOSE[3251] asterisk.c: Executing last minute cleanups