set debug 4 Core debug was 5 and is now 4 The 'set debug' command is deprecated and will be removed in a future release. Please use 'core set debug' instead. *CLI> set verbose 4 Verbosity was 5 and is now 4 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. *CLI> [Jan 30 18:42:46] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '7edd20344f786c154064055c22061aa8@84.234.24.35' [Jan 30 18:42:46] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 7edd20344f786c154064055c22061aa8@84.234.24.35 [Jan 30 18:42:46] Really destroying SIP dialog '7edd20344f786c154064055c22061aa8@84.234.24.35' Method: REGISTER [Jan 30 18:42:46] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '1ca036d136e94b54794fce437567dd2a@84.234.24.35' [Jan 30 18:42:46] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 1ca036d136e94b54794fce437567dd2a@84.234.24.35 [Jan 30 18:42:46] Really destroying SIP dialog '1ca036d136e94b54794fce437567dd2a@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '72cc7485690ac780330647ca3966cd55@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 72cc7485690ac780330647ca3966cd55@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '72cc7485690ac780330647ca3966cd55@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '5575c9cd0c5f152105aee3647b16c08f@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '5575c9cd0c5f152105aee3647b16c08f@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '016ec07d16e0aeca1526e49a01d3e954@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '016ec07d16e0aeca1526e49a01d3e954@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '68c690c351d1584304d7a00d3e3e6d62@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '68c690c351d1584304d7a00d3e3e6d62@84.234.24.35' Method: REGISTER [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '49c55cb9562471da5bd3553f56ea15ce@84.234.24.35' [Jan 30 18:42:47] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 [Jan 30 18:42:47] Really destroying SIP dialog '49c55cb9562471da5bd3553f56ea15ce@84.234.24.35' Method: REGISTER [Jan 30 18:42:48] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> [Jan 30 18:43:02] <--- SIP read from 83.105.95.134:43652 ---> INVITE sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK9bcd03308d37b404 From: "James B Warrington" ;tag=2c52c682e2e2191d To: Contact: Supported: replaces, timer Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12979 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8000 8000 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@www.jb-consultancy.com SIP/2.0 (46) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK9bcd03308d37b404 (63) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=2c52c682e2e2191d (81) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Supported: replaces, timer (26) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 61b3f03b07b4c951@10.0.0.213 (36) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 12979 INVITE (18) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 331 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: (0) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=1001 8000 8000 IN IP4 10.0.0.213 (34) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 30 18:43:02] --- (13 headers 16 lines) --- [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 61b3f03b07b4c951@10.0.0.213 - INVITE (With RTP) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -replaces- [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: replaces [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Jan 30 18:43:02] Sending to 10.0.0.213 : 4068 (no NAT) [Jan 30 18:43:02] Using INVITE request as basis request - 61b3f03b07b4c951@10.0.0.213 [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:02] <--- Reliably Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK9bcd03308d37b404;received=83.105.95.134 From: "James B Warrington" ;tag=2c52c682e2e2191d To: ;tag=as30e96139 Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66c47a43" Content-Length: 0 <------------> [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #137 [Jan 30 18:43:02] Scheduling destruction of SIP dialog '61b3f03b07b4c951@10.0.0.213' in 32000 ms (Method: INVITE) [Jan 30 18:43:02] Found user '1001' [Jan 30 18:43:02] <--- SIP read from 83.105.95.134:43652 ---> ACK sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK9bcd03308d37b404 From: "James B Warrington" ;tag=2c52c682e2e2191d To: ;tag=as30e96139 Contact: Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12979 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1110@www.jb-consultancy.com SIP/2.0 (43) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK9bcd03308d37b404 (63) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=2c52c682e2e2191d (81) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=as30e96139 (52) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 61b3f03b07b4c951@10.0.0.213 (36) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 12979 ACK (15) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (21) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:02] --- (11 headers 0 lines) --- [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #137 [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '61b3f03b07b4c951@10.0.0.213' of Response 12979: Match Not Found [Jan 30 18:43:02] <--- SIP read from 83.105.95.134:43652 ---> INVITE sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKd67c2856e3768bc6 From: "James B Warrington" ;tag=2c52c682e2e2191d To: Contact: Supported: replaces, timer Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="66c47a43", response="5847f4533e19f0a792da5b02ddeb1e15" Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12980 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8000 8001 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@www.jb-consultancy.com SIP/2.0 (46) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKd67c2856e3768bc6 (63) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=2c52c682e2e2191d (81) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Supported: replaces, timer (26) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="66c47a43", response="5847f4533e19f0a792da5b02ddeb1e15" (178) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Call-ID: 61b3f03b07b4c951@10.0.0.213 (36) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: CSeq: 12980 INVITE (18) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Content-Length: 331 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: (0) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=1001 8000 8001 IN IP4 10.0.0.213 (34) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 30 18:43:02] --- (14 headers 16 lines) --- [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:02] Sending to 83.105.95.134 : 43652 (NAT) [Jan 30 18:43:02] Using INVITE request as basis request - 61b3f03b07b4c951@10.0.0.213 [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:02] Found user '1001' [Jan 30 18:43:02] Found RTP audio format 0 [Jan 30 18:43:02] Found RTP audio format 3 [Jan 30 18:43:02] Found RTP audio format 4 [Jan 30 18:43:02] Found RTP audio format 8 [Jan 30 18:43:02] Found RTP audio format 18 [Jan 30 18:43:02] Found RTP audio format 2 [Jan 30 18:43:02] Found RTP audio format 101 [Jan 30 18:43:02] Peer audio RTP is at port 10.0.0.213:6008 [Jan 30 18:43:02] Found description format PCMU for ID 0 [Jan 30 18:43:02] Found description format GSM for ID 3 [Jan 30 18:43:02] Found description format G723 for ID 4 [Jan 30 18:43:02] Found description format PCMA for ID 8 [Jan 30 18:43:02] Found description format G729 for ID 18 [Jan 30 18:43:02] Found description format G726-32 for ID 2 [Jan 30 18:43:02] Found description format telephone-event for ID 101 [Jan 30 18:43:02] Got unsupported a:fmtp in SDP offer [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel [Jan 30 18:43:02] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:43:02] Peer audio RTP is at port 10.0.0.213:6008 [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:13326 handle_request_invite: Checking SIP call limits for device 1001 [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3066 update_call_counter: Call from peer '1001' is 1 out of 200 [Jan 30 18:43:02] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:02] Looking for 1110 in int_jbc (domain www.jb-consultancy.com) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:7922 build_route: build_route: Contact hop: [Jan 30 18:43:02] list_route: hop: [Jan 30 18:43:02] DEBUG[10962]: chan_sip.c:13401 handle_request_invite: SIP/1001-081ca4b0: New call is still down.... Trying... [Jan 30 18:43:02] <--- Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKd67c2856e3768bc6;received=83.105.95.134 From: "James B Warrington" ;tag=2c52c682e2e2191d To: Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12980 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:02] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081ca4b0 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:02] DEBUG[10976]: pbx.c:1770 pbx_extension_helper: Launching 'Set' [Jan 30 18:43:02] -- Executing [1110@int_jbc:1] Set("SIP/1001-081ca4b0", "OUTTRUNK=JBC_OUT") in new stack [Jan 30 18:43:02] DEBUG[10976]: pbx.c:1770 pbx_extension_helper: Launching 'SIPAddHeader' [Jan 30 18:43:02] -- Executing [1110@int_jbc:2] SIPAddHeader("SIP/1001-081ca4b0", ""Alert-Info: "") in new stack [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:16882 sip_addheader: SIP Header added ""Alert-Info: "" as _SIPADDHEADER01 [Jan 30 18:43:02] DEBUG[10976]: pbx.c:1770 pbx_extension_helper: Launching 'Dial' [Jan 30 18:43:02] -- Executing [1110@int_jbc:3] Dial("SIP/1001-081ca4b0", "SIP/1110|30|tT") in new stack [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:15191 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3792 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-3. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3281 ast_channel_inherit_variables: Copying soft-transferable variable SIPADDHEADER01. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-2. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable OUTTRUNK. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-1. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 30 18:43:02] DEBUG[10976]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:2828 sip_call: Outgoing Call for 1110 [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:3066 update_call_counter: Call to peer '1110' is 1 out of 200 [Jan 30 18:43:02] DEBUG[10976]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 6 (Ringing) [Jan 30 18:43:02] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:02] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:2842 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:6786 transmit_invite: Adding SIP Header "Alert-Info" with content :: [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:6149 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 30 18:43:02] Audio is at 84.234.24.35 port 15346 [Jan 30 18:43:02] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:02] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:02] Adding codec 0x2 (gsm) to SDP [Jan 30 18:43:02] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@10.0.0.220 SIP/2.0 (34) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport (63) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 3: To: (25) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 4: Contact: (32) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:02 GMT (35) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 12: Alert-Info: (26) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 13: Content-Type: application/sdp (29) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 14: Content-Length: 287 (19) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4547 parse_request: Header 15: (0) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: o=root 10976 10976 IN IP4 84.234.24.35 (38) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: s=session (9) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: m=audio 15346 RTP/AVP 0 8 3 101 (31) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:02] Reliably Transmitting (NAT) to 83.105.95.134:44868: INVITE sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport From: "James B Warrington" ;tag=as6dfc924b To: Contact: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Alert-Info: Content-Type: application/sdp Content-Length: 287 v=0 o=root 10976 10976 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 15346 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 30 18:43:02] DEBUG[10976]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #139 [Jan 30 18:43:02] -- Called 1110 [Jan 30 18:43:03] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 100 Trying Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=52cb89920d299d7 (45) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport (63) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: (0) [Jan 30 18:43:03] --- (8 headers 0 lines) --- [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:2120 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #139 - INVITE (got response) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' Request 102: Found [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 100 to standard invite [Jan 30 18:43:03] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 180 Ringing Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport Content-Length: 0 Allow-Events: talk, hold, conference Contact: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=52cb89920d299d7 (45) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport (63) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Allow-Events: talk, hold, conference (36) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: (30) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:03] --- (10 headers 0 lines) --- [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' Request 102: Found [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 180 to standard invite [Jan 30 18:43:03] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081d5ee8 [Jan 30 18:43:03] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:03] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:03] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 6 (Ringing) [Jan 30 18:43:03] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:03] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:03] -- SIP/1110-081d5ee8 is ringing [Jan 30 18:43:03] <--- Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKd67c2856e3768bc6;received=83.105.95.134 From: "James B Warrington" ;tag=2c52c682e2e2191d To: ;tag=as0c33fcab Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12980 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:03] <--- SIP read from 89.213.46.56:65406 ---> NOTIFY sip:www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-bbc9eacc From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11169 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:www.jb-consultancy.com SIP/2.0 (41) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-bbc9eacc (59) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "Ext 1000" ;tag=9404e41c296a3eb8o0 (73) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 75a4866c-d81e80a8@192.168.1.211 (40) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 11169 NOTIFY (18) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Event: keep-alive (17) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Linksys/SPA941-5.1.5 (32) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (17) [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:03] --- (10 headers 0 lines) --- [Jan 30 18:43:03] <--- Transmitting (no NAT) to 89.213.46.56:65406 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-bbc9eacc;received=89.213.46.56 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: ;tag=as1b109520 Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11169 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:43:03] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 [Jan 30 18:43:10] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport Content-Length: 249 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 825697346 IN IP4 10.0.0.220 s=SIP Call c=IN IP4 10.0.0.220 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=52cb89920d299d7 (45) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK6ed383f0;rport (63) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 249 (19) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Type: application/sdp (29) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Supported: replaces (19) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Contact: (30) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: (0) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=MxSIP 0 825697346 IN IP4 10.0.0.220 (37) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.220 (19) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 30 18:43:10] --- (13 headers 12 lines) --- [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 200 to standard invite [Jan 30 18:43:10] Found RTP audio format 0 [Jan 30 18:43:10] Found RTP audio format 8 [Jan 30 18:43:10] Found RTP audio format 101 [Jan 30 18:43:10] Peer audio RTP is at port 10.0.0.220:3000 [Jan 30 18:43:10] Found description format PCMU for ID 0 [Jan 30 18:43:10] Found description format PCMA for ID 8 [Jan 30 18:43:10] Found description format telephone-event for ID 101 [Jan 30 18:43:10] Got unsupported a:fmtp in SDP offer [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel SIP/1110-081d5ee8 [Jan 30 18:43:10] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jan 30 18:43:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:43:10] Peer audio RTP is at port 10.0.0.220:3000 [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:5174 process_sdp: We have an owner, now see if we need to change this call [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:43:10] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:7922 build_route: build_route: Contact hop: [Jan 30 18:43:10] list_route: hop: [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:5604 reqprep: Strict routing enforced for session 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:10] set_destination: Parsing for address/port to send to [Jan 30 18:43:10] set_destination: set destination to 10.0.0.220, port 5060 [Jan 30 18:43:10] Transmitting (NAT) to 83.105.95.134:44868: ACK sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK3e8669c1;rport From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Contact: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:10] DEBUG[10976]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081d5ee8 [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:10] -- SIP/1110-081d5ee8 answered SIP/1001-081ca4b0 [Jan 30 18:43:10] DEBUG[10976]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081ca4b0 [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 30 18:43:10] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:10] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:3445 sip_answer: SIP answering channel: SIP/1001-081ca4b0 [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:6381 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:6149 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 30 18:43:10] Audio is at 84.234.24.35 port 15272 [Jan 30 18:43:10] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:10] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:10] Adding codec 0x2 (gsm) to SDP [Jan 30 18:43:10] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 30 18:43:10] <--- Reliably Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKd67c2856e3768bc6;received=83.105.95.134 From: "James B Warrington" ;tag=2c52c682e2e2191d To: ;tag=as0c33fcab Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12980 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 10976 10976 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 15272 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 30 18:43:10] DEBUG[10976]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #141 [Jan 30 18:43:10] <--- SIP read from 83.105.95.134:43652 ---> ACK sip:1110@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK975fa2e23e50ab21 From: "James B Warrington" ;tag=2c52c682e2e2191d To: ;tag=as0c33fcab Contact: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@84.234.24.35", nonce="66c47a43", response="d07c5f2d9dcfef95f58d9381a88812de" Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 12980 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1110@84.234.24.35 SIP/2.0 (33) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK975fa2e23e50ab21 (63) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=2c52c682e2e2191d (81) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=as0c33fcab (52) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@84.234.24.35", nonce="66c47a43", response="d07c5f2d9dcfef95f58d9381a88812de" (168) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 61b3f03b07b4c951@10.0.0.213 (36) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 12980 ACK (15) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 0 (21) [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:43:10] --- (12 headers 0 lines) --- [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #141 [Jan 30 18:43:10] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '61b3f03b07b4c951@10.0.0.213' of Response 12980: Match Not Found [Jan 30 18:43:10] DEBUG[10976]: rtp.c:1149 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:45592 [Jan 30 18:43:10] DEBUG[10976]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 30 18:43:10] DEBUG[10976]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 18:43:10] DEBUG[10976]: rtp.c:1149 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:42808 [Jan 30 18:43:10] DEBUG[10976]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 30 18:43:10] DEBUG[10976]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 18:43:11] DEBUG[10976]: rtp.c:871 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 30 18:43:12] <--- SIP read from 90.195.58.193:50626 ---> <-------------> [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:12] --- (0 headers 1 lines) --- [Jan 30 18:43:12] <--- SIP read from 83.105.95.134:44868 ---> REGISTER sip:www.jb-consultancy.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK39c732a9c Max-Forwards: 70 Content-Length: 0 To: Kevin Drinkwater From: Kevin Drinkwater ;tag=5788fcb6a98facb Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595184 REGISTER Contact: Kevin Drinkwater Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Authorization:Digest response="420dd821f1027a6d67c529c7ac2c802b",username="1110",realm="asterisk",nonce="026f65ec",algorithm=MD5,uri="sip:www.jb-consultancy.com:5060" User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:www.jb-consultancy.com:5060 SIP/2.0 (48) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK39c732a9c (51) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: Kevin Drinkwater (59) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=5788fcb6a98facb (81) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 (52) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1123595184 REGISTER (25) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: Kevin Drinkwater (47) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Authorization:Digest response="420dd821f1027a6d67c529c7ac2c802b",username="1110",realm="asterisk",nonce="026f65ec",algorithm=MD5,uri="sip:www.jb-consultancy.com:5060" (166) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: (0) [Jan 30 18:43:12] --- (13 headers 0 lines) --- [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 - REGISTER (No RTP) [Jan 30 18:43:12] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 30 18:43:12] Using latest REGISTER request as basis request [Jan 30 18:43:12] Sending to 10.0.0.220 : 5060 (no NAT) [Jan 30 18:43:12] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK39c732a9c;received=83.105.95.134 From: Kevin Drinkwater ;tag=5788fcb6a98facb To: Kevin Drinkwater Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595184 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:12] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK39c732a9c;received=83.105.95.134 From: Kevin Drinkwater ;tag=5788fcb6a98facb To: Kevin Drinkwater ;tag=as51e006a1 Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595184 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0865e814" Content-Length: 0 <------------> [Jan 30 18:43:12] Scheduling destruction of SIP dialog 'b2de0b34830f475ebd5bcb2745fec653@10.0.0.220' in 32000 ms (Method: REGISTER) [Jan 30 18:43:13] <--- SIP read from 83.105.95.134:44868 ---> REGISTER sip:www.jb-consultancy.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK4af62a084 Max-Forwards: 70 Content-Length: 0 To: Kevin Drinkwater From: Kevin Drinkwater ;tag=5788fcb6a98facb Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595185 REGISTER Contact: Kevin Drinkwater Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Authorization:Digest response="7d8b4e0d343e584e7abc3fbeef74b887",username="1110",realm="asterisk",nonce="0865e814",algorithm=MD5,uri="sip:www.jb-consultancy.com:5060" User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:www.jb-consultancy.com:5060 SIP/2.0 (48) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK4af62a084 (51) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: Kevin Drinkwater (59) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=5788fcb6a98facb (81) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 (52) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1123595185 REGISTER (25) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: Kevin Drinkwater (47) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Authorization:Digest response="7d8b4e0d343e584e7abc3fbeef74b887",username="1110",realm="asterisk",nonce="0865e814",algorithm=MD5,uri="sip:www.jb-consultancy.com:5060" (166) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: (0) [Jan 30 18:43:13] --- (13 headers 0 lines) --- [Jan 30 18:43:13] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 30 18:43:13] Using latest REGISTER request as basis request [Jan 30 18:43:13] Sending to 83.105.95.134 : 44868 (NAT) [Jan 30 18:43:13] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK4af62a084;received=83.105.95.134 From: Kevin Drinkwater ;tag=5788fcb6a98facb To: Kevin Drinkwater Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595185 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:13] -- Saved useragent "Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26" for peer 1110 [Jan 30 18:43:13] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK4af62a084;received=83.105.95.134 From: Kevin Drinkwater ;tag=5788fcb6a98facb To: Kevin Drinkwater ;tag=as51e006a1 Call-ID: b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 CSeq: 1123595185 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Tue, 30 Jan 2007 18:43:13 GMT Content-Length: 0 <------------> [Jan 30 18:43:13] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:43:13] Scheduling destruction of SIP dialog 'b2de0b34830f475ebd5bcb2745fec653@10.0.0.220' in 32000 ms (Method: REGISTER) [Jan 30 18:43:13] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:13] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:13] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:13] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:13] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:14] <--- SIP read from 83.105.95.134:33742 ---> <-------------> [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:14] --- (0 headers 1 lines) --- [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1110@10.0.0.220 SIP/2.0 (35) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7cafb929;rport (63) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as1d3bc9c5 (59) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (25) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 052e5f2714f666d93e04aec179e19752@84.234.24.35 (54) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:14 GMT (35) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:14] Reliably Transmitting (NAT) to 83.105.95.134:44868: OPTIONS sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7cafb929;rport From: "asterisk" ;tag=as1d3bc9c5 To: Contact: Call-ID: 052e5f2714f666d93e04aec179e19752@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #147 [Jan 30 18:43:14] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 052e5f2714f666d93e04aec179e19752@84.234.24.35 CSeq: 102 OPTIONS From: "asterisk" ;tag=as1d3bc9c5 To: ;tag=a0975c08b72357f Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7cafb929;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 052e5f2714f666d93e04aec179e19752@84.234.24.35 (54) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 OPTIONS (17) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "asterisk" ;tag=as1d3bc9c5 (59) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=a0975c08b72357f (45) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7cafb929;rport (63) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: (30) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces (19) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:14] --- (11 headers 0 lines) --- [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #147 [Jan 30 18:43:14] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '052e5f2714f666d93e04aec179e19752@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:14] Really destroying SIP dialog '052e5f2714f666d93e04aec179e19752@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1202@10.0.0.100:29848 SIP/2.0 (41) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK21640db9;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as0cf7d347 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (31) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 71f6b0ba070cc2de344f6f482d67c401@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 80.177.210.179:29848: OPTIONS sip:1202@10.0.0.100:29848 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK21640db9;rport From: "asterisk" ;tag=as0cf7d347 To: Contact: Call-ID: 71f6b0ba070cc2de344f6f482d67c401@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #150 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1300@192.168.2.3:25756;rinstance=cc4dbab69ab7c5cb SIP/2.0 (69) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK66b3b693;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as68d1f605 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 01c8846432b1e59d68e45a9d1510ebda@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 83.192.156.15:25756: OPTIONS sip:1300@192.168.2.3:25756;rinstance=cc4dbab69ab7c5cb SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK66b3b693;rport From: "asterisk" ;tag=as68d1f605 To: Contact: Call-ID: 01c8846432b1e59d68e45a9d1510ebda@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #152 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:pvt1_1201@10.27.10.99:40618 SIP/2.0 (47) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2eb7972a;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as7c64a26d (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 5a79c6cc487659186ab1497302bc51ac@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 82.33.171.11:40618: OPTIONS sip:pvt1_1201@10.27.10.99:40618 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2eb7972a;rport From: "asterisk" ;tag=as7c64a26d To: Contact: Call-ID: 5a79c6cc487659186ab1497302bc51ac@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #154 [Jan 30 18:43:15] <--- SIP read from 82.33.171.11:40618 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2eb7972a;rport From: "asterisk" ;tag=as7c64a26d To: ;tag=d8fe382fcfbe6361 Call-ID: 5a79c6cc487659186ab1497302bc51ac@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2eb7972a;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as7c64a26d (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=d8fe382fcfbe6361 (58) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 5a79c6cc487659186ab1497302bc51ac@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (42) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:15] --- (11 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #154 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '5a79c6cc487659186ab1497302bc51ac@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '5a79c6cc487659186ab1497302bc51ac@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] <--- SIP read from 80.177.210.179:29848 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK21640db9;rport From: "asterisk" ;tag=as0cf7d347 To: ;tag=ab99e8c0265f0d0a Call-ID: 71f6b0ba070cc2de344f6f482d67c401@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK21640db9;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as0cf7d347 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=ab99e8c0265f0d0a (52) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 71f6b0ba070cc2de344f6f482d67c401@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:15] --- (11 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #150 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '71f6b0ba070cc2de344f6f482d67c401@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '71f6b0ba070cc2de344f6f482d67c401@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1201@10.27.10.99:56962 SIP/2.0 (42) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1d1ee05f;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as1774e87a (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 51c6dc13589e0fb353c4d64018b0f986@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 82.33.171.11:56962: OPTIONS sip:1201@10.27.10.99:56962 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1d1ee05f;rport From: "asterisk" ;tag=as1774e87a To: Contact: Call-ID: 51c6dc13589e0fb353c4d64018b0f986@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #158 [Jan 30 18:43:15] <--- SIP read from 82.33.171.11:56962 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1d1ee05f;rport From: "asterisk" ;tag=as1774e87a To: ;tag=8a9d2e1ceb62f870 Call-ID: 51c6dc13589e0fb353c4d64018b0f986@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1d1ee05f;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as1774e87a (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=8a9d2e1ceb62f870 (53) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 51c6dc13589e0fb353c4d64018b0f986@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (37) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:15] --- (11 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #158 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '51c6dc13589e0fb353c4d64018b0f986@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '51c6dc13589e0fb353c4d64018b0f986@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] <--- SIP read from 83.192.156.15:25756 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK66b3b693;rport=5060 Contact: To: ;tag=7a183f25 From: "asterisk";tag=as68d1f605 Call-ID: 01c8846432b1e59d68e45a9d1510ebda@84.234.24.35 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK66b3b693;rport=5060 (68) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Contact: (32) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=7a183f25 (72) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: From: "asterisk";tag=as68d1f605 (58) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 01c8846432b1e59d68e45a9d1510ebda@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Accept: application/sdp (23) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Accept-Language: en (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:43:15] --- (12 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #152 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '01c8846432b1e59d68e45a9d1510ebda@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '01c8846432b1e59d68e45a9d1510ebda@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1200@10.0.1.100:46310 SIP/2.0 (41) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7768ed2a;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as78737933 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (31) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 80.176.225.214:46310: OPTIONS sip:1200@10.0.1.100:46310 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7768ed2a;rport From: "asterisk" ;tag=as78737933 To: Contact: Call-ID: 365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #162 [Jan 30 18:43:15] <--- SIP read from 80.176.225.214:46310 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7768ed2a;rport From: "asterisk" ;tag=as78737933 To: ;tag=cc9d6aef8b2f615b Call-ID: 365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7768ed2a;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as78737933 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=cc9d6aef8b2f615b (52) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:15] --- (11 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #162 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '365584a73ebb6cbb6f0cb65a0c48801f@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1100@90.195.58.193:50626;rinstance=602a4fcf08e4447f SIP/2.0 (71) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5cf7f7f1;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as0ef196c6 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (61) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 90.195.58.193:50626: OPTIONS sip:1100@90.195.58.193:50626;rinstance=602a4fcf08e4447f SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5cf7f7f1;rport From: "asterisk" ;tag=as0ef196c6 To: Contact: Call-ID: 022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #165 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1003@83.105.95.134:33742;rinstance=dc1e019750a9b869 SIP/2.0 (71) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK33454742;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as087fffd9 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (61) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7bf2aae27ce1b20549996d4f326b6354@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 83.105.95.134:33742: OPTIONS sip:1003@83.105.95.134:33742;rinstance=dc1e019750a9b869 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK33454742;rport From: "asterisk" ;tag=as087fffd9 To: Contact: Call-ID: 7bf2aae27ce1b20549996d4f326b6354@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #167 [Jan 30 18:43:15] <--- SIP read from 90.195.58.193:50626 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5cf7f7f1;rport=5060 Contact: To: ;tag=7e5aaa28 From: "asterisk";tag=as0ef196c6 Call-ID: 022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5cf7f7f1;rport=5060 (68) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Contact: (32) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=7e5aaa28 (74) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: From: "asterisk";tag=as0ef196c6 (58) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Accept: application/sdp (23) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Accept-Language: en (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:43:15] --- (12 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #165 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '022dee0c4f2cdec470b7e95e7dafbca5@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1001@10.0.0.213:4068 SIP/2.0 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK22800023;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as6d2a40ec (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (30) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 83.105.95.134:43652: OPTIONS sip:1001@10.0.0.213:4068 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK22800023;rport From: "asterisk" ;tag=as6d2a40ec To: Contact: Call-ID: 652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #170 [Jan 30 18:43:15] <--- SIP read from 83.105.95.134:33742 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK33454742;rport=5060 Contact: To: ;tag=be118029 From: "asterisk";tag=as087fffd9 Call-ID: 7bf2aae27ce1b20549996d4f326b6354@84.234.24.35 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK33454742;rport=5060 (68) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Contact: (31) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=be118029 (74) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: From: "asterisk";tag=as087fffd9 (58) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7bf2aae27ce1b20549996d4f326b6354@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Accept: application/sdp (23) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Accept-Language: en (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:43:15] --- (12 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #167 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7bf2aae27ce1b20549996d4f326b6354@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '7bf2aae27ce1b20549996d4f326b6354@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] <--- SIP read from 83.105.95.134:43652 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK22800023;rport From: "asterisk" ;tag=as6d2a40ec To: ;tag=as0c33fcab Call-ID: 652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK22800023;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as6d2a40ec (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=as0c33fcab (45) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:15] --- (11 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #170 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '652671430f2a7f1d1ff3073a1c57a9a7@84.234.24.35' Method: OPTIONS [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: OPTIONS sip:1000@192.168.1.211:5060 SIP/2.0 (43) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK792b4d47;rport (63) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as5d5585f3 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (33) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 3237c24870b6e17f1f965079370b38f3@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:15 GMT (35) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:15] Reliably Transmitting (NAT) to 89.213.46.56:65406: OPTIONS sip:1000@192.168.1.211:5060 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK792b4d47;rport From: "asterisk" ;tag=as5d5585f3 To: Contact: Call-ID: 3237c24870b6e17f1f965079370b38f3@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #174 [Jan 30 18:43:15] <--- SIP read from 89.213.46.56:65406 ---> SIP/2.0 200 OK To: ;tag=e04e0aec15aec628i0 From: "asterisk" ;tag=as5d5585f3 Call-ID: 3237c24870b6e17f1f965079370b38f3@84.234.24.35 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK792b4d47 Server: Linksys/SPA941-5.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: To: ;tag=e04e0aec15aec628i0 (56) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as5d5585f3 (59) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Call-ID: 3237c24870b6e17f1f965079370b38f3@84.234.24.35 (54) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK792b4d47 (57) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Server: Linksys/SPA941-5.1.5 (28) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Content-Length: 0 (17) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces (19) [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:15] --- (10 headers 0 lines) --- [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #174 [Jan 30 18:43:15] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '3237c24870b6e17f1f965079370b38f3@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:15] Really destroying SIP dialog '3237c24870b6e17f1f965079370b38f3@84.234.24.35' Method: OPTIONS [Jan 30 18:43:16] DEBUG[10976]: rtp.c:871 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 30 18:43:16] <--- SIP read from 83.105.95.134:44868 ---> INVITE sip:1001@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKaa3a2cf1f Max-Forwards: 70 Content-Length: 246 To: "James B Warrington" ;tag=as6dfc924b From: ;tag=52cb89920d299d7 Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800973 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 825697347 IN IP4 10.0.0.220 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1001@84.234.24.35 SIP/2.0 (36) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKaa3a2cf1f (51) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 246 (19) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: "James B Warrington" ;tag=as6dfc924b (63) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: ;tag=52cb89920d299d7 (47) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1132800973 INVITE (23) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: timer (16) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Contact: (30) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Supported: replaces (19) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 15: (0) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=MxSIP 0 825697347 IN IP4 10.0.0.220 (37) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 30 18:43:16] --- (15 headers 12 lines) --- [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: timer" [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Jan 30 18:43:16] Sending to 83.105.95.134 : 44868 (NAT) [Jan 30 18:43:16] Found RTP audio format 0 [Jan 30 18:43:16] Found RTP audio format 8 [Jan 30 18:43:16] Found RTP audio format 101 [Jan 30 18:43:16] Peer audio RTP is at port 0.0.0.0:3000 [Jan 30 18:43:16] Found description format PCMU for ID 0 [Jan 30 18:43:16] Found description format PCMA for ID 8 [Jan 30 18:43:16] Found description format telephone-event for ID 101 [Jan 30 18:43:16] Got unsupported a:fmtp in SDP offer [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel SIP/1110-081d5ee8 [Jan 30 18:43:16] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jan 30 18:43:16] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:43:16] Peer audio RTP is at port 0.0.0.0:3000 [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:5174 process_sdp: We have an owner, now see if we need to change this call [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:13379 handle_request_invite: Got a SIP re-invite for call 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:13474 handle_request_invite: SIP/1110-081d5ee8: This call is UP.... [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:6381 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:6149 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 30 18:43:16] Audio is at 84.234.24.35 port 15346 [Jan 30 18:43:16] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:16] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:16] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jan 30 18:43:16] <--- Reliably Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKaa3a2cf1f;received=83.105.95.134 From: ;tag=52cb89920d299d7 To: "James B Warrington" ;tag=as6dfc924b Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800973 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 10976 10977 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 15346 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #177 [Jan 30 18:43:16] -- Started music on hold, class 'default', on SIP/1001-081ca4b0 [Jan 30 18:43:16] DEBUG[10976]: channel.c:1991 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 30 18:43:16] DEBUG[10976]: channel.c:2325 __ast_read: Generator got voice, switching to phase locked mode [Jan 30 18:43:16] DEBUG[10976]: channel.c:1991 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 30 18:43:16] DEBUG[10976]: channel.c:2838 set_format: Set channel SIP/1001-081ca4b0 to write format slin [Jan 30 18:43:16] DEBUG[10976]: res_musiconhold.c:253 ast_moh_files_next: SIP/1001-081ca4b0 Opened file 1 '/var/lib/asterisk/moh/fpm-sunshine' [Jan 30 18:43:16] <--- SIP read from 83.105.95.134:44868 ---> ACK sip:1001@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK606d83c9e Max-Forwards: 70 Content-Length: 0 To: "James B Warrington" ;tag=as6dfc924b From: ;tag=52cb89920d299d7 Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800973 ACK Contact: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1001@84.234.24.35 SIP/2.0 (33) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK606d83c9e (51) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: "James B Warrington" ;tag=as6dfc924b (63) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: ;tag=52cb89920d299d7 (47) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1132800973 ACK (20) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: (30) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:16] --- (10 headers 0 lines) --- [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #177 [Jan 30 18:43:16] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' of Response 1132800973: Match Not Found [Jan 30 18:43:18] <--- SIP read from 89.213.46.56:65406 ---> NOTIFY sip:www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a8f244ae From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11170 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:www.jb-consultancy.com SIP/2.0 (41) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a8f244ae (59) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "Ext 1000" ;tag=9404e41c296a3eb8o0 (73) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 75a4866c-d81e80a8@192.168.1.211 (40) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 11170 NOTIFY (18) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Event: keep-alive (17) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Linksys/SPA941-5.1.5 (32) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (17) [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:18] --- (10 headers 0 lines) --- [Jan 30 18:43:18] <--- Transmitting (no NAT) to 89.213.46.56:65406 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a8f244ae;received=89.213.46.56 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: ;tag=as49d8c28c Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11170 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:43:18] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 [Jan 30 18:43:19] <--- SIP read from 83.192.156.15:25756 ---> <-------------> [Jan 30 18:43:19] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:19] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:19] --- (0 headers 1 lines) --- [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Jan 30 18:43:20] Scheduling destruction of SIP dialog '03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35' in 7040 ms (Method: NOTIFY) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:1110@10.0.0.220 SIP/2.0 (34) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7d9a5133;rport (63) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "asterisk" ;tag=as283fca39 (59) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (25) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (36) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35 (54) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 NOTIFY (16) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Event: message-summary (22) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Type: application/simple-message-summary (48) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 87 (18) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: Messages-Waiting: no (20) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: Message-Account: sip:500@84.234.24.35 (37) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: Voice-Message: 0/0 (0/0) (24) [Jan 30 18:43:20] Reliably Transmitting (NAT) to 83.105.95.134:44868: NOTIFY sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7d9a5133;rport From: "asterisk" ;tag=as283fca39 To: Contact: Call-ID: 03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 87 Messages-Waiting: no Message-Account: sip:500@84.234.24.35 Voice-Message: 0/0 (0/0) --- [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #179 [Jan 30 18:43:20] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35 CSeq: 102 NOTIFY From: "asterisk" ;tag=as283fca39 To: ;tag=15213f601dd9883 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7d9a5133;rport Content-Length: 0 Contact: Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35 (54) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 NOTIFY (16) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "asterisk" ;tag=as283fca39 (59) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=15213f601dd9883 (45) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7d9a5133;rport (63) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (30) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: replaces (19) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:20] --- (10 headers 0 lines) --- [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #179 [Jan 30 18:43:20] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:20] Really destroying SIP dialog '03bc38a53a11bd4c78cd2a0867d9e062@84.234.24.35' Method: NOTIFY [Jan 30 18:43:20] RTCP SR transmission error, rtcp halted [Jan 30 18:43:20] NOTICE[10962]: sched.c:283 ast_sched_del: Attempted to delete nonexistent schedule entry 143! [Jan 30 18:43:21] <--- SIP read from 83.105.95.134:44868 ---> INVITE sip:1003@www.jb-consultancy.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK89c1b926b Max-Forwards: 70 Content-Length: 274 To: 1003 From: Kevin Drinkwater ;tag=cb4785690eaec3d Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635678 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Kevin Drinkwater Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 975047167 IN IP4 10.0.0.220 s=SIP Call c=IN IP4 10.0.0.220 t=0 0 m=audio 3002 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1003@www.jb-consultancy.com:5060 SIP/2.0 (51) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK89c1b926b (51) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 274 (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: 1003 (47) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=cb4785690eaec3d (81) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (52) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1777635678 INVITE (23) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: timer (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Contact: Kevin Drinkwater (47) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Supported: replaces (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 15: (0) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=MxSIP 0 975047167 IN IP4 10.0.0.220 (37) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.220 (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 3002 RTP/AVP 0 8 18 101 (31) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:30 (10) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:on - - - - (24) [Jan 30 18:43:21] --- (15 headers 13 lines) --- [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 - INVITE (With RTP) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: timer" [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Jan 30 18:43:21] Sending to 10.0.0.220 : 5060 (no NAT) [Jan 30 18:43:21] Using INVITE request as basis request - 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:21] <--- Reliably Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK89c1b926b;received=83.105.95.134 From: Kevin Drinkwater ;tag=cb4785690eaec3d To: 1003 ;tag=as51eff4ae Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635678 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e22794f" Content-Length: 0 <------------> [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #180 [Jan 30 18:43:21] Scheduling destruction of SIP dialog '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' in 32000 ms (Method: INVITE) [Jan 30 18:43:21] Found user '1110' [Jan 30 18:43:21] <--- SIP read from 83.105.95.134:44868 ---> ACK sip:1003@www.jb-consultancy.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK89c1b926b Max-Forwards: 70 Content-Length: 0 To: 1003 ;tag=as51eff4ae From: Kevin Drinkwater ;tag=cb4785690eaec3d Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635678 ACK User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1003@www.jb-consultancy.com:5060 SIP/2.0 (48) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK89c1b926b (51) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: 1003 ;tag=as51eff4ae (62) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=cb4785690eaec3d (81) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (52) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1777635678 ACK (20) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: (0) [Jan 30 18:43:21] --- (9 headers 0 lines) --- [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #180 [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' of Response 1777635678: Match Not Found [Jan 30 18:43:21] <--- SIP read from 83.105.95.134:44868 ---> INVITE sip:1003@www.jb-consultancy.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40 Max-Forwards: 70 Content-Length: 274 To: 1003 From: Kevin Drinkwater ;tag=cb4785690eaec3d Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Kevin Drinkwater Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="aa990e519a87ebabfb26b10e26dd5627",username="1110",realm="asterisk",nonce="1e22794f",algorithm=MD5,uri="sip:1003@www.jb-consultancy.com:5060" User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 975047167 IN IP4 10.0.0.220 s=SIP Call c=IN IP4 10.0.0.220 t=0 0 m=audio 3002 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1003@www.jb-consultancy.com:5060 SIP/2.0 (51) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40 (51) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 274 (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: 1003 (47) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=cb4785690eaec3d (81) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (52) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1777635679 INVITE (23) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: timer (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Contact: Kevin Drinkwater (47) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Supported: replaces (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: Proxy-Authorization:Digest response="aa990e519a87ebabfb26b10e26dd5627",username="1110",realm="asterisk",nonce="1e22794f",algorithm=MD5,uri="sip:1003@www.jb-consultancy.com:5060" (177) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 15: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 16: (0) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=MxSIP 0 975047167 IN IP4 10.0.0.220 (37) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.220 (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 3002 RTP/AVP 0 8 18 101 (31) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:30 (10) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:on - - - - (24) [Jan 30 18:43:21] --- (16 headers 13 lines) --- [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:21] Sending to 83.105.95.134 : 44868 (NAT) [Jan 30 18:43:21] Using INVITE request as basis request - 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:21] Found user '1110' [Jan 30 18:43:21] Found RTP audio format 0 [Jan 30 18:43:21] Found RTP audio format 8 [Jan 30 18:43:21] Found RTP audio format 18 [Jan 30 18:43:21] Found RTP audio format 101 [Jan 30 18:43:21] Peer audio RTP is at port 10.0.0.220:3002 [Jan 30 18:43:21] Found description format PCMU for ID 0 [Jan 30 18:43:21] Found description format PCMA for ID 8 [Jan 30 18:43:21] Found description format G729 for ID 18 [Jan 30 18:43:21] Found description format telephone-event for ID 101 [Jan 30 18:43:21] Got unsupported a:fmtp in SDP offer [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel [Jan 30 18:43:21] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jan 30 18:43:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:43:21] Peer audio RTP is at port 10.0.0.220:3002 [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:13326 handle_request_invite: Checking SIP call limits for device 1110 [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3066 update_call_counter: Call from peer '1110' is 2 out of 200 [Jan 30 18:43:21] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:43:21] Looking for 1003 in int_kevd (domain www.jb-consultancy.com) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:7922 build_route: build_route: Contact hop: Kevin Drinkwater [Jan 30 18:43:21] list_route: hop: [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:13401 handle_request_invite: SIP/1110-081c70f8: New call is still down.... Trying... [Jan 30 18:43:21] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40;received=83.105.95.134 From: Kevin Drinkwater ;tag=cb4785690eaec3d To: 1003 Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:21] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081c70f8 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:21] DEBUG[10984]: pbx.c:1770 pbx_extension_helper: Launching 'Set' [Jan 30 18:43:21] -- Executing [1003@int_kevd:1] Set("SIP/1110-081c70f8", "OUTTRUNK=KEVD_OUT") in new stack [Jan 30 18:43:21] DEBUG[10984]: pbx.c:1770 pbx_extension_helper: Launching 'NoOp' [Jan 30 18:43:21] -- Executing [1003@int_kevd:2] NoOp("SIP/1110-081c70f8", "") in new stack [Jan 30 18:43:21] DEBUG[10984]: pbx.c:1770 pbx_extension_helper: Launching 'Dial' [Jan 30 18:43:21] -- Executing [1003@int_kevd:3] Dial("SIP/1110-081c70f8", "SIP/1003|30|tT") in new stack [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:15191 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3792 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_kevd-1003-3. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_kevd-1003-2. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable OUTTRUNK. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_kevd-1003-1. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 30 18:43:21] DEBUG[10984]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:2828 sip_call: Outgoing Call for 1003 [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:3066 update_call_counter: Call to peer '1003' is 1 out of 200 [Jan 30 18:43:21] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:2842 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:6149 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 30 18:43:21] Audio is at 84.234.24.35 port 16376 [Jan 30 18:43:21] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:21] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:21] Adding codec 0x2 (gsm) to SDP [Jan 30 18:43:21] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1003@83.105.95.134:33742;rinstance=dc1e019750a9b869 SIP/2.0 (70) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport (63) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 2: From: "Kevin Drinkwater" ;tag=as3bb42809 (63) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 3: To: (61) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 4: Contact: (32) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 (54) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:43:21 GMT (35) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 13: Content-Length: 287 (19) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4547 parse_request: Header 14: (0) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: o=root 10984 10984 IN IP4 84.234.24.35 (38) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: s=session (9) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: m=audio 16376 RTP/AVP 0 8 3 101 (31) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:21] Reliably Transmitting (NAT) to 83.105.95.134:33742: INVITE sip:1003@83.105.95.134:33742;rinstance=dc1e019750a9b869 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport From: "Kevin Drinkwater" ;tag=as3bb42809 To: Contact: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:43:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 10984 10984 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 16376 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 30 18:43:21] DEBUG[10984]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #182 [Jan 30 18:43:21] -- Called 1003 [Jan 30 18:43:21] <--- SIP read from 83.105.95.134:33742 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport=5060 Contact: To: ;tag=6a07df26 From: "Kevin Drinkwater";tag=as3bb42809 Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport=5060 (68) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Contact: (66) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=6a07df26 (74) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: From: "Kevin Drinkwater";tag=as3bb42809 (62) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 (54) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: (0) [Jan 30 18:43:21] --- (9 headers 0 lines) --- [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2120 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #182 - INVITE (got response) [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35' Request 102: Found [Jan 30 18:43:21] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 180 to standard invite [Jan 30 18:43:21] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081dab20 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 30 18:43:21] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:21] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:21] -- SIP/1003-081dab20 is ringing [Jan 30 18:43:21] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40;received=83.105.95.134 From: Kevin Drinkwater ;tag=cb4785690eaec3d To: 1003 ;tag=as32c92e30 Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:26] DEBUG[10984]: rtp.c:866 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 83.105.95.134:61081 [Jan 30 18:43:26] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 132 bytes [Jan 30 18:43:26] DEBUG[10984]: rtp.c:1149 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:61080 [Jan 30 18:43:26] DEBUG[10984]: chan_sip.c:6381 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 30 18:43:26] DEBUG[10984]: chan_sip.c:6149 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jan 30 18:43:26] DEBUG[10984]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 30 18:43:26] Audio is at 84.234.24.35 port 10870 [Jan 30 18:43:26] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:26] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:26] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:26] DEBUG[10984]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:26] DEBUG[10984]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jan 30 18:43:26] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40;received=83.105.95.134 From: Kevin Drinkwater ;tag=cb4785690eaec3d To: 1003 ;tag=as32c92e30 Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 10984 10984 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 10870 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 30 18:43:26] DEBUG[10984]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 30 18:43:26] DEBUG[10984]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 18:43:27] <--- SIP read from 83.105.95.134:33742 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport=5060 Contact: To: ;tag=6a07df26 From: "Kevin Drinkwater";tag=as3bb42809 Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 183 v=0 o=- 1 2 IN IP4 10.0.0.250 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.250 t=0 0 m=audio 53080 RTP/AVP 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK499242d6;rport=5060 (68) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Contact: (66) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=6a07df26 (74) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: From: "Kevin Drinkwater";tag=as3bb42809 (62) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 (54) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 183 (19) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=- 1 2 IN IP4 10.0.0.250 (25) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.250 (19) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 53080 RTP/AVP 0 8 3 101 (31) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:27] --- (11 headers 9 lines) --- [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 200 to standard invite [Jan 30 18:43:27] Found RTP audio format 0 [Jan 30 18:43:27] Found RTP audio format 8 [Jan 30 18:43:27] Found RTP audio format 3 [Jan 30 18:43:27] Found RTP audio format 101 [Jan 30 18:43:27] Peer audio RTP is at port 10.0.0.250:53080 [Jan 30 18:43:27] Got unsupported a:fmtp in SDP offer [Jan 30 18:43:27] Found description format telephone-event for ID 101 [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel SIP/1003-081dab20 [Jan 30 18:43:27] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 30 18:43:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:43:27] Peer audio RTP is at port 10.0.0.250:53080 [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:5174 process_sdp: We have an owner, now see if we need to change this call [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:43:27] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 2 (In use) [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:7922 build_route: build_route: Contact hop: [Jan 30 18:43:27] list_route: hop: [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:5604 reqprep: Strict routing enforced for session 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 [Jan 30 18:43:27] set_destination: Parsing for address/port to send to [Jan 30 18:43:27] set_destination: set destination to 83.105.95.134, port 33742 [Jan 30 18:43:27] Transmitting (NAT) to 83.105.95.134:33742: ACK sip:1003@83.105.95.134:33742;rinstance=dc1e019750a9b869 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4c03ddc9;rport From: "Kevin Drinkwater" ;tag=as3bb42809 To: ;tag=6a07df26 Contact: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:43:27] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081dab20 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 2 (In use) [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:27] -- SIP/1003-081dab20 answered SIP/1110-081c70f8 [Jan 30 18:43:27] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081c70f8 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:27] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:27] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:3445 sip_answer: SIP answering channel: SIP/1110-081c70f8 [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:6381 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:6149 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 30 18:43:27] Audio is at 84.234.24.35 port 10870 [Jan 30 18:43:27] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:43:27] Adding codec 0x8 (alaw) to SDP [Jan 30 18:43:27] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jan 30 18:43:27] <--- Reliably Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKee5b41c40;received=83.105.95.134 From: Kevin Drinkwater ;tag=cb4785690eaec3d To: 1003 ;tag=as32c92e30 Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 10984 10985 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 10870 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 30 18:43:27] DEBUG[10984]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #186 [Jan 30 18:43:27] DEBUG[10984]: rtp.c:1149 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:61080 [Jan 30 18:43:27] DEBUG[10984]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 18:43:27] <--- SIP read from 83.105.95.134:44868 ---> ACK sip:1003@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK70ccaadf6 Max-Forwards: 70 Content-Length: 0 To: 1003 ;tag=as32c92e30 From: Kevin Drinkwater ;tag=cb4785690eaec3d Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 1777635679 ACK Contact: Kevin Drinkwater Proxy-Authorization:Digest response="0c29e68087a5f73755f91e1bb8d2d461",username="1110",realm="asterisk",nonce="1e22794f",algorithm=MD5,uri="sip:1003@84.234.24.35" User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1003@84.234.24.35 SIP/2.0 (33) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bK70ccaadf6 (51) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: 1003 ;tag=as32c92e30 (62) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: Kevin Drinkwater ;tag=cb4785690eaec3d (81) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (52) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1777635679 ACK (20) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: Kevin Drinkwater (47) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Proxy-Authorization:Digest response="0c29e68087a5f73755f91e1bb8d2d461",username="1110",realm="asterisk",nonce="1e22794f",algorithm=MD5,uri="sip:1003@84.234.24.35" (162) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:27] --- (11 headers 0 lines) --- [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #186 [Jan 30 18:43:27] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' of Response 1777635679: Match Not Found [Jan 30 18:43:27] DEBUG[10984]: rtp.c:1149 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:42810 [Jan 30 18:43:27] DEBUG[10984]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 30 18:43:27] DEBUG[10984]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 18:43:29] <--- SIP read from 83.105.95.134:44868 ---> REFER sip:1001@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKfa8c6ece9 Max-Forwards: 70 Content-Length: 0 To: "James B Warrington" ;tag=as6dfc924b From: ;tag=52cb89920d299d7 Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800974 REFER Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Refer-To: sip:1003@www.jb-consultancy.com:5060?Replaces=317e388f9c1f17511419b520b5a07ec6%4010.0.0.220%3bto-tag%3das32c92e30%3bfrom-tag%3dcb4785690eaec3d Referred-By: Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REFER sip:1001@84.234.24.35 SIP/2.0 (35) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKfa8c6ece9 (51) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: "James B Warrington" ;tag=as6dfc924b (63) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: ;tag=52cb89920d299d7 (47) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1132800974 REFER (22) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: timer (16) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Contact: (30) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Refer-To: sip:1003@www.jb-consultancy.com:5060?Replaces=317e388f9c1f17511419b520b5a07ec6%4010.0.0.220%3bto-tag%3das32c92e30%3bfrom-tag%3dcb4785690eaec3d (152) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Referred-By: (34) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: Supported: replaces (19) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 15: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 16: (0) [Jan 30 18:43:29] --- (16 headers 0 lines) --- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received REFER (9) - Command in SIP REFER [Jan 30 18:43:29] Call 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 got a SIP call transfer from caller: (REFER)! [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:8654 get_refer_info: Attended transfer: Will use Replace-Call-ID : 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (No check of from/to tags) [Jan 30 18:43:29] SIP transfer to extension 1003@int_kevd by 1110@10.0.0.220 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:13879 handle_request_refer: SIP attended transfer: Transferer channel SIP/1110-081d5ee8, transferee channel SIP/1001-081ca4b0 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:13895 handle_request_refer: Got SIP transfer, applying to bridged peer 'SIP/1001-081ca4b0' [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:8508 get_sip_pvt_byid_locked: Looking for callid 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (fromtag cb4785690eaec3d totag as32c92e30) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:8532 get_sip_pvt_byid_locked: Matched INCOMING call - their tag is cb4785690eaec3d Our tag is as32c92e30 [Jan 30 18:43:29] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKfa8c6ece9;received=83.105.95.134 From: ;tag=52cb89920d299d7 To: "James B Warrington" ;tag=as6dfc924b Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800974 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:13650 local_attended_transfer: SIP attended transfer: trying to bridge SIP/1110-081c70f8 and SIP/1001-081ca4b0 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12663 attempt_transfer: Sip transfer:-------------------- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12665 attempt_transfer: -- Transferer to PBX channel: SIP/1110-081d5ee8 State Up [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12669 attempt_transfer: -- Transferer to PBX second channel (target): SIP/1110-081c70f8 State Up [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12673 attempt_transfer: -- Bridged call to transferee: SIP/1001-081ca4b0 State Up [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12677 attempt_transfer: -- Bridged call to transfer target: SIP/1003-081dab20 State Up [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12680 attempt_transfer: -- END Sip transfer:-------------------- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12688 attempt_transfer: SIP transfer: Four channels to handle [Jan 30 18:43:29] -- Stopped music on hold on SIP/1001-081ca4b0 [Jan 30 18:43:29] DEBUG[10962]: channel.c:2838 set_format: Set channel SIP/1001-081ca4b0 to write format ulaw [Jan 30 18:43:29] DEBUG[10962]: channel.c:1991 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12719 attempt_transfer: SIP transfer: trying to masquerade SIP/1001-081ca4b0 into SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10962]: channel.c:3227 ast_channel_masquerade: Planning to masquerade channel SIP/1001-081ca4b0 into the structure of SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10962]: channel.c:3241 ast_channel_masquerade: Done planning to masquerade channel SIP/1001-081ca4b0 into the structure of SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:12724 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:5604 reqprep: Strict routing enforced for session 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:29] set_destination: Parsing for address/port to send to [Jan 30 18:43:29] set_destination: set destination to 10.0.0.220, port 5060 [Jan 30 18:43:29] Reliably Transmitting (NAT) to 83.105.95.134:44868: NOTIFY sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK543e79a3;rport From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Contact: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=1132800974 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #187 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:13678 local_attended_transfer: SIP attended transfer: Unlocking channel SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10984]: channel.c:3358 ast_do_masquerade: Actually Masquerading SIP/1001-081ca4b0(6) into the structure of SIP/1110-081c70f8(6) [Jan 30 18:43:29] DEBUG[10984]: channel.c:3370 ast_do_masquerade: Got clone lock for masquerade on 'SIP/1001-081ca4b0' at 0x81b50c0 [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:3570 sip_fixup: SIP Fixup: New owner for dialogue 317e388f9c1f17511419b520b5a07ec6@10.0.0.220: SIP/1001-081ca4b0 (Old parent: SIP/1001-081ca4b0) [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:3296 sip_hangup: Hangup call SIP/1001-081ca4b0, SIP callid 317e388f9c1f17511419b520b5a07ec6@10.0.0.220) [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:3304 sip_hangup: update_call_counter(1110) - decrement call limit counter on hangup [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:3040 update_call_counter: Call from peer '1110' removed from call limit 200 [Jan 30 18:43:29] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] Scheduling destruction of SIP dialog '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' in 32000 ms (Method: ACK) [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:5604 reqprep: Strict routing enforced for session 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 [Jan 30 18:43:29] set_destination: Parsing for address/port to send to [Jan 30 18:43:29] set_destination: set destination to 10.0.0.220, port 5060 [Jan 30 18:43:29] Reliably Transmitting (NAT) to 83.105.95.134:44868: BYE sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0dc0ef01;rport From: 1003 ;tag=as32c92e30 To: Kevin Drinkwater ;tag=cb4785690eaec3d Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #189 [Jan 30 18:43:29] DEBUG[10984]: channel.c:3565 ast_do_masquerade: Putting channel SIP/1001-081ca4b0 in 4/4 formats [Jan 30 18:43:29] DEBUG[10984]: chan_sip.c:3570 sip_fixup: SIP Fixup: New owner for dialogue 61b3f03b07b4c951@10.0.0.213: SIP/1001-081ca4b0 (Old parent: SIP/1110-081c70f8) [Jan 30 18:43:29] DEBUG[10984]: channel.c:3601 ast_do_masquerade: Released clone lock on 'SIP/1110-081c70f8' [Jan 30 18:43:29] DEBUG[10984]: channel.c:3611 ast_do_masquerade: Done Masquerading SIP/1001-081ca4b0 (6) [Jan 30 18:43:29] DEBUG[10976]: channel.c:3793 ast_generic_bridge: Didn't get a frame from channel: SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10976]: channel.c:4111 ast_channel_bridge: Bridge stops bridging channels SIP/1110-081c70f8 and SIP/1110-081d5ee8 [Jan 30 18:43:29] DEBUG[10976]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/1110-081d5ee8' [Jan 30 18:43:29] DEBUG[10976]: chan_sip.c:3281 sip_hangup: SIP Transfer: Not hanging up right now... Rescheduling hangup for 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35. [Jan 30 18:43:29] Scheduling destruction of SIP dialog '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' in 10240 ms (Method: REFER) [Jan 30 18:43:29] DEBUG[10976]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081d5ee8 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] DEBUG[10976]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 30 18:43:29] DEBUG[10976]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 30 18:43:29] DEBUG[10976]: pbx.c:2368 __ast_pbx_run: Spawn extension (int_jbc,1110,3) exited non-zero on 'SIP/1110-081c70f8' [Jan 30 18:43:29] == Spawn extension (int_jbc, 1110, 3) exited non-zero on 'SIP/1110-081c70f8' [Jan 30 18:43:29] DEBUG[10976]: channel.c:1692 ast_hangup: Hanging up zombie 'SIP/1110-081c70f8' [Jan 30 18:43:29] DEBUG[10976]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081c70f8 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:43:29] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:43:29] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:43:29] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 103 NOTIFY From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK543e79a3;rport Content-Length: 0 Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 103 NOTIFY (16) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=52cb89920d299d7 (45) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK543e79a3;rport (63) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Supported: replaces (19) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: (0) [Jan 30 18:43:29] --- (9 headers 0 lines) --- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #187 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' of Request 103: Match Not Found [Jan 30 18:43:29] SIP Response message for INCOMING dialog NOTIFY arrived [Jan 30 18:43:29] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 CSeq: 102 BYE From: 1003 ;tag=as32c92e30 To: Kevin Drinkwater ;tag=cb4785690eaec3d Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0dc0ef01;rport Content-Length: 0 Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 317e388f9c1f17511419b520b5a07ec6@10.0.0.220 (52) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 BYE (13) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: 1003 ;tag=as32c92e30 (64) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: Kevin Drinkwater ;tag=cb4785690eaec3d (79) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0dc0ef01;rport (63) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Supported: replaces (19) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: (0) [Jan 30 18:43:29] --- (9 headers 0 lines) --- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #189 [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' of Request 102: Match Not Found [Jan 30 18:43:29] SIP Response message for INCOMING dialog BYE arrived [Jan 30 18:43:29] Really destroying SIP dialog '317e388f9c1f17511419b520b5a07ec6@10.0.0.220' Method: ACK [Jan 30 18:43:29] <--- SIP read from 83.105.95.134:44868 ---> BYE sip:1001@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKf003ed5c4 Max-Forwards: 70 Content-Length: 0 To: "James B Warrington" ;tag=as6dfc924b From: ;tag=52cb89920d299d7 Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800975 BYE Supported: timer Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: BYE sip:1001@84.234.24.35 SIP/2.0 (33) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKf003ed5c4 (51) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Content-Length: 0 (17) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: "James B Warrington" ;tag=as6dfc924b (63) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: ;tag=52cb89920d299d7 (47) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 1132800975 BYE (20) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: timer (16) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces (19) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:29] --- (11 headers 0 lines) --- [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received BYE (8) - Command in SIP BYE [Jan 30 18:43:29] Sending to 83.105.95.134 : 44868 (NAT) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:29] Scheduling destruction of SIP dialog '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' in 10240 ms (Method: BYE) [Jan 30 18:43:29] DEBUG[10962]: chan_sip.c:14119 handle_request_bye: Received bye, no owner, selfdestruct soon. [Jan 30 18:43:29] <--- Transmitting (NAT) to 83.105.95.134:44868 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.220;branch=z9hG4bKf003ed5c4;received=83.105.95.134 From: ;tag=52cb89920d299d7 To: "James B Warrington" ;tag=as6dfc924b Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 1132800975 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:29] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:32] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:33] <--- SIP read from 89.213.46.56:65406 ---> NOTIFY sip:www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-4339ea14 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11171 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:www.jb-consultancy.com SIP/2.0 (41) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-4339ea14 (59) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "Ext 1000" ;tag=9404e41c296a3eb8o0 (73) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 75a4866c-d81e80a8@192.168.1.211 (40) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 11171 NOTIFY (18) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Event: keep-alive (17) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Linksys/SPA941-5.1.5 (32) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (17) [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:33] --- (10 headers 0 lines) --- [Jan 30 18:43:33] <--- Transmitting (no NAT) to 89.213.46.56:65406 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-4339ea14;received=89.213.46.56 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: ;tag=as362d4062 Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11171 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:43:33] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 *CLI> [Jan 30 18:43:36] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes show hints -= Registered Asterisk Dial Plan Hints =- 1301@internal_all : SIP/1301 State:Unavailable Watchers 0 1300@internal_all : SIP/1300 State:Idle Watchers 0 1202@internal_all : SIP/1202 State:Idle Watchers 0 1201@internal_all : SIP/1201 State:Idle Watchers 0 1200@internal_all : SIP/1200 State:Idle Watchers 0 1110@internal_all : SIP/1110 State:InUse Watchers 0 1100@internal_all : SIP/1100 State:Idle Watchers 0 1003@internal_all : SIP/1003 State:InUse Watchers 0 1002@internal_all : SIP/1002 State:Unavailable Watchers 0 1001@internal_all : SIP/1001 State:InUse Watchers 0 1000@internal_all : SIP/1000 State:Idle Watchers 0 ---------------- - 11 hints registered The 'show hints' command is deprecated and will be removed in a future release. Please use 'core show hints' instead. *CLI> [Jan 30 18:43:39] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:2015 __sip_autodestruct: Finally hanging up channel after transfer: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:5604 reqprep: Strict routing enforced for session 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 [Jan 30 18:43:39] set_destination: Parsing for address/port to send to [Jan 30 18:43:39] set_destination: set destination to 10.0.0.220, port 5060 [Jan 30 18:43:39] Reliably Transmitting (NAT) to 83.105.95.134:44868: BYE sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2cfa6d8b;rport From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #192 [Jan 30 18:43:39] Scheduling destruction of SIP dialog '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' in 10240 ms (Method: BYE) [Jan 30 18:43:39] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 481 Call Does Not Exist Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 CSeq: 104 BYE From: "James B Warrington" ;tag=as6dfc924b To: ;tag=52cb89920d299d7 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2cfa6d8b;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 481 Call Does Not Exist (31) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 2d32246a479d8ce64b5d33af06d733a0@84.234.24.35 (54) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 104 BYE (13) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as6dfc924b (65) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=52cb89920d299d7 (45) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2cfa6d8b;rport (63) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: (0) [Jan 30 18:43:39] --- (8 headers 0 lines) --- [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #192 [Jan 30 18:43:39] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:43:39] SIP Response message for INCOMING dialog BYE arrived [Jan 30 18:43:39] Really destroying SIP dialog '2d32246a479d8ce64b5d33af06d733a0@84.234.24.35' Method: BYE [Jan 30 18:43:42] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:42] <--- SIP read from 90.195.58.193:50626 ---> <-------------> [Jan 30 18:43:42] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:42] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:42] --- (0 headers 1 lines) --- [Jan 30 18:43:44] <--- SIP read from 83.105.95.134:33742 ---> <-------------> [Jan 30 18:43:44] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:44] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:44] --- (0 headers 1 lines) --- [Jan 30 18:43:45] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:45] DEBUG[10962]: chan_sip.c:2008 __sip_autodestruct: Auto destroying SIP dialog 'b2de0b34830f475ebd5bcb2745fec653@10.0.0.220' [Jan 30 18:43:45] DEBUG[10962]: chan_sip.c:3098 sip_destroy: Destroying SIP dialog b2de0b34830f475ebd5bcb2745fec653@10.0.0.220 [Jan 30 18:43:45] Really destroying SIP dialog 'b2de0b34830f475ebd5bcb2745fec653@10.0.0.220' Method: REGISTER [Jan 30 18:43:48] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:48] <--- SIP read from 89.213.46.56:65406 ---> NOTIFY sip:www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a4ed19c7 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11172 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:www.jb-consultancy.com SIP/2.0 (41) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a4ed19c7 (59) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "Ext 1000" ;tag=9404e41c296a3eb8o0 (73) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 75a4866c-d81e80a8@192.168.1.211 (40) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 11172 NOTIFY (18) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Event: keep-alive (17) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Linksys/SPA941-5.1.5 (32) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (17) [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:48] --- (10 headers 0 lines) --- [Jan 30 18:43:48] <--- Transmitting (no NAT) to 89.213.46.56:65406 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-a4ed19c7;received=89.213.46.56 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: ;tag=as7b1e9b6a Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11172 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:43:48] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 [Jan 30 18:43:49] <--- SIP read from 83.192.156.15:25756 ---> <-------------> [Jan 30 18:43:49] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: (0) [Jan 30 18:43:49] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: (0) [Jan 30 18:43:49] --- (0 headers 1 lines) --- [Jan 30 18:43:51] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 30 18:43:51] DEBUG[10984]: rtp.c:871 ast_rtcp_read: Got RTCP report of 160 bytes [Jan 30 18:43:52] <--- SIP read from 83.105.95.134:33742 ---> BYE sip:1110@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:25742;branch=z9hG4bK-d87543-c273447b38234533-1--d87543-;rport Max-Forwards: 70 Contact: To: "Kevin Drinkwater";tag=as3bb42809 From: ;tag=6a07df26 Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 2 BYE User-Agent: X-Lite release 1006e stamp 34025 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: BYE sip:1110@84.234.24.35 SIP/2.0 (33) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.250:25742;branch=z9hG4bK-d87543-c273447b38234533-1--d87543-;rport (89) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: Max-Forwards: 70 (16) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: Contact: (66) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: "Kevin Drinkwater";tag=as3bb42809 (60) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: From: ;tag=6a07df26 (76) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 (54) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 2 BYE (11) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Reason: SIP;description="User Hung Up" (38) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:52] --- (11 headers 0 lines) --- [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received BYE (8) - Command in SIP BYE [Jan 30 18:43:52] Sending to 83.105.95.134 : 33742 (NAT) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:14115 handle_request_bye: Received bye, issuing owner hangup [Jan 30 18:43:52] <--- Transmitting (NAT) to 83.105.95.134:33742 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.250:25742;branch=z9hG4bK-d87543-c273447b38234533-1--d87543-;received=83.105.95.134;rport=33742 From: ;tag=6a07df26 To: "Kevin Drinkwater";tag=as3bb42809 Call-ID: 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:43:52] DEBUG[10984]: channel.c:3793 ast_generic_bridge: Didn't get a frame from channel: SIP/1003-081dab20 [Jan 30 18:43:52] DEBUG[10984]: channel.c:4111 ast_channel_bridge: Bridge stops bridging channels SIP/1001-081ca4b0 and SIP/1003-081dab20 [Jan 30 18:43:52] DEBUG[10984]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/1003-081dab20' [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3296 sip_hangup: Hangup call SIP/1003-081dab20, SIP callid 7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35) [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3304 sip_hangup: update_call_counter(1003) - decrement call limit counter on hangup [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3040 update_call_counter: Call to peer '1003' removed from call limit 200 [Jan 30 18:43:52] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:52] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081dab20 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1003 [Jan 30 18:43:52] DEBUG[10984]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 30 18:43:52] DEBUG[10984]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 30 18:43:52] DEBUG[10984]: pbx.c:2368 __ast_pbx_run: Spawn extension (int_kevd,1003,3) exited non-zero on 'SIP/1001-081ca4b0' [Jan 30 18:43:52] == Spawn extension (int_kevd, 1003, 3) exited non-zero on 'SIP/1001-081ca4b0' [Jan 30 18:43:52] DEBUG[10984]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/1001-081ca4b0' [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3296 sip_hangup: Hangup call SIP/1001-081ca4b0, SIP callid 61b3f03b07b4c951@10.0.0.213) [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3304 sip_hangup: update_call_counter(1001) - decrement call limit counter on hangup [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:3040 update_call_counter: Call from peer '1001' removed from call limit 200 [Jan 30 18:43:52] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 1 (Not in use) [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:52] Scheduling destruction of SIP dialog '61b3f03b07b4c951@10.0.0.213' in 32000 ms (Method: ACK) [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:5604 reqprep: Strict routing enforced for session 61b3f03b07b4c951@10.0.0.213 [Jan 30 18:43:52] set_destination: Parsing for address/port to send to [Jan 30 18:43:52] set_destination: set destination to 10.0.0.213, port 4068 [Jan 30 18:43:52] Reliably Transmitting (NAT) to 83.105.95.134:43652: BYE sip:1001@10.0.0.213:4068 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK20c4a62b;rport From: ;tag=as0c33fcab To: "James B Warrington" ;tag=2c52c682e2e2191d Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:43:52] DEBUG[10984]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #195 [Jan 30 18:43:52] DEBUG[10984]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081ca4b0 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 1 (Not in use) [Jan 30 18:43:52] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:43:52] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:43:52] <--- SIP read from 83.105.95.134:43652 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK20c4a62b;rport From: ;tag=as0c33fcab To: "James B Warrington" ;tag=2c52c682e2e2191d Call-ID: 61b3f03b07b4c951@10.0.0.213 CSeq: 102 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK20c4a62b;rport (63) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as0c33fcab (54) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: "James B Warrington" ;tag=2c52c682e2e2191d (79) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 61b3f03b07b4c951@10.0.0.213 (36) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 102 BYE (13) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (35) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Supported: replaces, timer (26) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (17) [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:43:52] --- (11 headers 0 lines) --- [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #195 [Jan 30 18:43:52] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '61b3f03b07b4c951@10.0.0.213' of Request 102: Match Not Found [Jan 30 18:43:52] SIP Response message for INCOMING dialog BYE arrived [Jan 30 18:43:52] Really destroying SIP dialog '7563f05b3efd5daa7168c0fe19d3d41d@84.234.24.35' Method: BYE [Jan 30 18:43:52] Really destroying SIP dialog '61b3f03b07b4c951@10.0.0.213' Method: ACK *CLI> show hints -= Registered Asterisk Dial Plan Hints =- 1301@internal_all : SIP/1301 State:Unavailable Watchers 0 1300@internal_all : SIP/1300 State:Idle Watchers 0 1202@internal_all : SIP/1202 State:Idle Watchers 0 1201@internal_all : SIP/1201 State:Idle Watchers 0 1200@internal_all : SIP/1200 State:Idle Watchers 0 1110@internal_all : SIP/1110 State:InUse Watchers 0 1100@internal_all : SIP/1100 State:Idle Watchers 0 1003@internal_all : SIP/1003 State:Idle Watchers 0 1002@internal_all : SIP/1002 State:Unavailable Watchers 0 1001@internal_all : SIP/1001 State:Idle Watchers 0 1000@internal_all : SIP/1000 State:Idle Watchers 0 ---------------- - 11 hints registered *CLI> [Jan 30 18:43:59] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7170288@talk.orbtalk.co.uk [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 7edd20344f786c154064055c22061aa8@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #196 [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7170288@talk.orbtalk.co.uk [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK14de783c;rport (63) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as5e1697e7 (53) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 7edd20344f786c154064055c22061aa8@84.234.24.35 (54) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7170288", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="57a395aa4749ee1ac227ff42584d8c0d", opaque="" (219) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:59] REGISTER 13 headers, 0 lines [Jan 30 18:43:59] REGISTER attempt 1 to 7170288@talk.orbtalk.co.uk [Jan 30 18:43:59] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK14de783c;rport From: ;tag=as5e1697e7 To: Call-ID: 7edd20344f786c154064055c22061aa8@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7170288", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="57a395aa4749ee1ac227ff42584d8c0d", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #197 [Jan 30 18:43:59] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK14de783c;rport=5060 From: ;tag=as5e1697e7 To: ;tag=13724058500c4c29465e0085181d997d.539e Call-ID: 7edd20344f786c154064055c22061aa8@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK14de783c;rport=5060 (68) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as5e1697e7 (53) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.539e (78) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 7edd20344f786c154064055c22061aa8@84.234.24.35 (54) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:59] --- (10 headers 0 lines) --- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #197 [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7edd20344f786c154064055c22061aa8@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 196 [Jan 30 18:43:59] Scheduling destruction of SIP dialog '7edd20344f786c154064055c22061aa8@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:43:59] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:43:59] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7170298@talk.orbtalk.co.uk [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 1ca036d136e94b54794fce437567dd2a@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #200 [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7170298@talk.orbtalk.co.uk [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK37a299fc;rport (63) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as5016fcc4 (53) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 1ca036d136e94b54794fce437567dd2a@84.234.24.35 (54) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7170298", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="95d7063cebb271af59dd10d481fb5238", opaque="" (219) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:43:59] REGISTER 13 headers, 0 lines [Jan 30 18:43:59] REGISTER attempt 1 to 7170298@talk.orbtalk.co.uk [Jan 30 18:43:59] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK37a299fc;rport From: ;tag=as5016fcc4 To: Call-ID: 1ca036d136e94b54794fce437567dd2a@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7170298", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="95d7063cebb271af59dd10d481fb5238", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #201 [Jan 30 18:43:59] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK37a299fc;rport=5060 From: ;tag=as5016fcc4 To: ;tag=13724058500c4c29465e0085181d997d.cd24 Call-ID: 1ca036d136e94b54794fce437567dd2a@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2151 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK37a299fc;rport=5060 (68) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as5016fcc4 (53) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.cd24 (78) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 1ca036d136e94b54794fce437567dd2a@84.234.24.35 (54) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2151 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:43:59] --- (10 headers 0 lines) --- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #201 [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '1ca036d136e94b54794fce437567dd2a@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 200 [Jan 30 18:43:59] Scheduling destruction of SIP dialog '1ca036d136e94b54794fce437567dd2a@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:43:59] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:43:59] <--- SIP read from 83.105.95.134:43652 ---> INVITE sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK2a654c6255879bf8 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: Contact: Supported: replaces, timer Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41448 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8000 8000 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@www.jb-consultancy.com SIP/2.0 (46) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK2a654c6255879bf8 (63) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=f95dfaf37b93c84e (81) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Supported: replaces, timer (26) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 665d922ff2e21edf@10.0.0.213 (36) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 41448 INVITE (18) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 331 (19) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: (0) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=1001 8000 8000 IN IP4 10.0.0.213 (34) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 30 18:43:59] --- (13 headers 16 lines) --- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to Off [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 665d922ff2e21edf@10.0.0.213 - INVITE (With RTP) [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1678 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -replaces- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: replaces [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1686 parse_sip_options: Found SIP option: -timer- [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1692 parse_sip_options: Matched SIP option: timer [Jan 30 18:43:59] Sending to 10.0.0.213 : 4068 (no NAT) [Jan 30 18:43:59] Using INVITE request as basis request - 665d922ff2e21edf@10.0.0.213 [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:43:59] <--- Reliably Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK2a654c6255879bf8;received=83.105.95.134 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as7dad237a Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41448 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43fac959" Content-Length: 0 <------------> [Jan 30 18:43:59] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #204 [Jan 30 18:43:59] Scheduling destruction of SIP dialog '665d922ff2e21edf@10.0.0.213' in 32000 ms (Method: INVITE) [Jan 30 18:43:59] Found user '1001' [Jan 30 18:44:00] <--- SIP read from 83.105.95.134:43652 ---> ACK sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK2a654c6255879bf8 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as7dad237a Contact: Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41448 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1110@www.jb-consultancy.com SIP/2.0 (43) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bK2a654c6255879bf8 (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=f95dfaf37b93c84e (81) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=as7dad237a (52) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 665d922ff2e21edf@10.0.0.213 (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: CSeq: 41448 ACK (15) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Content-Length: 0 (21) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: (0) [Jan 30 18:44:00] --- (11 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #204 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '665d922ff2e21edf@10.0.0.213' of Response 41448: Match Not Found [Jan 30 18:44:00] <--- SIP read from 83.105.95.134:43652 ---> INVITE sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: Contact: Supported: replaces, timer Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="43fac959", response="974db3bd9ae51327c0260419b5537385" Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8000 8001 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@www.jb-consultancy.com SIP/2.0 (46) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=f95dfaf37b93c84e (81) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Supported: replaces, timer (26) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="43fac959", response="974db3bd9ae51327c0260419b5537385" (178) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Call-ID: 665d922ff2e21edf@10.0.0.213 (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: CSeq: 41449 INVITE (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 13: Content-Length: 331 (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 14: (0) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: o=1001 8000 8001 IN IP4 10.0.0.213 (34) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: s=SIP Call (10) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: m=audio 6008 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 30 18:44:00] --- (14 headers 16 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 30 18:44:00] Sending to 83.105.95.134 : 43652 (NAT) [Jan 30 18:44:00] Using INVITE request as basis request - 665d922ff2e21edf@10.0.0.213 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:44:00] Found user '1001' [Jan 30 18:44:00] Found RTP audio format 0 [Jan 30 18:44:00] Found RTP audio format 3 [Jan 30 18:44:00] Found RTP audio format 4 [Jan 30 18:44:00] Found RTP audio format 8 [Jan 30 18:44:00] Found RTP audio format 18 [Jan 30 18:44:00] Found RTP audio format 2 [Jan 30 18:44:00] Found RTP audio format 101 [Jan 30 18:44:00] Peer audio RTP is at port 10.0.0.213:6008 [Jan 30 18:44:00] Found description format PCMU for ID 0 [Jan 30 18:44:00] Found description format GSM for ID 3 [Jan 30 18:44:00] Found description format G723 for ID 4 [Jan 30 18:44:00] Found description format PCMA for ID 8 [Jan 30 18:44:00] Found description format G729 for ID 18 [Jan 30 18:44:00] Found description format G726-32 for ID 2 [Jan 30 18:44:00] Found description format telephone-event for ID 101 [Jan 30 18:44:00] Got unsupported a:fmtp in SDP offer [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:5090 process_sdp: T38 state changed to 0 on channel [Jan 30 18:44:00] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 30 18:44:00] Peer audio RTP is at port 10.0.0.213:6008 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:5167 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:13326 handle_request_invite: Checking SIP call limits for device 1001 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3066 update_call_counter: Call from peer '1001' is 1 out of 200 [Jan 30 18:44:00] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:00] Looking for 1110 in int_jbc (domain www.jb-consultancy.com) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7922 build_route: build_route: Contact hop: [Jan 30 18:44:00] list_route: hop: [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:13401 handle_request_invite: SIP/1001-081c4210: New call is still down.... Trying... [Jan 30 18:44:00] <--- Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888;received=83.105.95.134 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:44:00] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081c4210 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:00] DEBUG[10986]: pbx.c:1770 pbx_extension_helper: Launching 'Set' [Jan 30 18:44:00] -- Executing [1110@int_jbc:1] Set("SIP/1001-081c4210", "OUTTRUNK=JBC_OUT") in new stack [Jan 30 18:44:00] DEBUG[10986]: pbx.c:1770 pbx_extension_helper: Launching 'SIPAddHeader' [Jan 30 18:44:00] -- Executing [1110@int_jbc:2] SIPAddHeader("SIP/1001-081c4210", ""Alert-Info: "") in new stack [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:16882 sip_addheader: SIP Header added ""Alert-Info: "" as _SIPADDHEADER01 [Jan 30 18:44:00] DEBUG[10986]: pbx.c:1770 pbx_extension_helper: Launching 'Dial' [Jan 30 18:44:00] -- Executing [1110@int_jbc:3] Dial("SIP/1001-081c4210", "SIP/1110|30|tT") in new stack [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:15191 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:2573 do_setnat: Setting NAT on RTP to On [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3787 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3788 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3789 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3790 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3792 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3813 sip_new: This channel will not be able to handle video. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-3. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3281 ast_channel_inherit_variables: Copying soft-transferable variable SIPADDHEADER01. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-2. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable OUTTRUNK. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1110-1. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 30 18:44:00] DEBUG[10986]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:2828 sip_call: Outgoing Call for 1110 [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:3066 update_call_counter: Call to peer '1110' is 2 out of 200 [Jan 30 18:44:00] DEBUG[10986]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 7 (Ring+Inuse) [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:2842 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:6786 transmit_invite: Adding SIP Header "Alert-Info" with content :: [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:6149 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:6150 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 30 18:44:00] Audio is at 84.234.24.35 port 28316 [Jan 30 18:44:00] Adding codec 0x4 (ulaw) to SDP [Jan 30 18:44:00] Adding codec 0x8 (alaw) to SDP [Jan 30 18:44:00] Adding codec 0x2 (gsm) to SDP [Jan 30 18:44:00] Adding non-codec 0x1 (telephone-event) to SDP [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:6281 add_sdp: -- Done with adding codecs to SDP [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:6326 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 0: INVITE sip:1110@10.0.0.220 SIP/2.0 (34) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport (63) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=as363e406c (65) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 3: To: (25) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 4: Contact: (32) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 5: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 9: Date: Tue, 30 Jan 2007 18:44:00 GMT (35) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 11: Supported: replaces (19) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 12: Alert-Info: (26) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 13: Content-Type: application/sdp (29) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 14: Content-Length: 287 (19) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4547 parse_request: Header 15: (0) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: v=0 (3) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: o=root 10986 10986 IN IP4 84.234.24.35 (38) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: s=session (9) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: t=0 0 (5) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: m=audio 28316 RTP/AVP 0 8 3 101 (31) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=ptime:20 (10) [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:4579 parse_request: Line: a=sendrecv (10) [Jan 30 18:44:00] Reliably Transmitting (NAT) to 83.105.95.134:44868: INVITE sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport From: "James B Warrington" ;tag=as363e406c To: Contact: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jan 2007 18:44:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Alert-Info: Content-Type: application/sdp Content-Length: 287 v=0 o=root 10986 10986 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 28316 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 30 18:44:00] DEBUG[10986]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #206 [Jan 30 18:44:00] -- Called 1110 [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7170297@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #208 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7170297@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK152a930d;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as123aa148 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7170297", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="eca89fad5fb73bef5387582ee8848cce", opaque="" (219) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 7170297@talk.orbtalk.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK152a930d;rport From: ;tag=as123aa148 To: Call-ID: 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7170297", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="eca89fad5fb73bef5387582ee8848cce", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #209 [Jan 30 18:44:00] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK152a930d;rport=5060 From: ;tag=as123aa148 To: ;tag=13724058500c4c29465e0085181d997d.85ad Call-ID: 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2151 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK152a930d;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as123aa148 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.85ad (78) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2151 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #209 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 208 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '6a2e4faa1571ee4d7e4635d06cad04b9@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 8687732@sipgate.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 72cc7485690ac780330647ca3966cd55@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] -- parse_srv: SRV mapped to host sipgate.co.uk, port 5060 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for sipgate.co.uk id #212 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 8687732@sipgate.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:sipgate.co.uk SIP/2.0 (34) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK564538aa;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as08135ebd (48) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (31) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 72cc7485690ac780330647ca3966cd55@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="8687732", realm="sipgate.co.uk", algorithm=MD5, uri="sip:sipgate.co.uk", nonce="45bf92b15c818183c33068c9cb4bab66d8855d47", response="8f2fbf7a73cdc746413b163b723cc43f", opaque="" (209) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 8687732@sipgate.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.10.79.23:5060: REGISTER sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK564538aa;rport From: ;tag=as08135ebd To: Call-ID: 72cc7485690ac780330647ca3966cd55@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="8687732", realm="sipgate.co.uk", algorithm=MD5, uri="sip:sipgate.co.uk", nonce="45bf92b15c818183c33068c9cb4bab66d8855d47", response="8f2fbf7a73cdc746413b163b723cc43f", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #213 [Jan 30 18:44:00] <--- SIP read from 217.10.79.23:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK564538aa;rport=5060 From: ;tag=as08135ebd To: ;tag=968ada9c3ffba887db8ff01d701887bd.aec4 Call-ID: 72cc7485690ac780330647ca3966cd55@84.234.24.35 CSeq: 104 REGISTER Date: Tue, 30 Jan 2007 18:43:58 GMT Contact: ;expires=120 Content-Length: 0 <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK564538aa;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as08135ebd (48) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=968ada9c3ffba887db8ff01d701887bd.aec4 (73) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 72cc7485690ac780330647ca3966cd55@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Date: Tue, 30 Jan 2007 18:43:58 GMT (35) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: (0) [Jan 30 18:44:00] --- (9 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #213 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '72cc7485690ac780330647ca3966cd55@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 212 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '72cc7485690ac780330647ca3966cd55@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7169833@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #216 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7169833@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5db44007;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as29de3ccb (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7169833", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="aa381073e339347325f398b73df91069", opaque="" (219) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 7169833@talk.orbtalk.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5db44007;rport From: ;tag=as29de3ccb To: Call-ID: 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7169833", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="aa381073e339347325f398b73df91069", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #217 [Jan 30 18:44:00] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5db44007;rport=5060 From: ;tag=as29de3ccb To: ;tag=13724058500c4c29465e0085181d997d.e363 Call-ID: 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5db44007;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as29de3ccb (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.e363 (78) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 5575c9cd0c5f152105aee3647b16c08f@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #217 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '5575c9cd0c5f152105aee3647b16c08f@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 216 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '5575c9cd0c5f152105aee3647b16c08f@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:00] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 100 Trying Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as363e406c To: ;tag=f866b25df2771ce Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as363e406c (65) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=f866b25df2771ce (45) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: (0) [Jan 30 18:44:00] --- (8 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2120 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #206 - INVITE (got response) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' Request 102: Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 100 to standard invite [Jan 30 18:44:00] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 180 Ringing Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as363e406c To: ;tag=f866b25df2771ce Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport Content-Length: 0 Allow-Events: talk, hold, conference Contact: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as363e406c (65) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=f866b25df2771ce (45) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Allow-Events: talk, hold, conference (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Contact: (30) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2129 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' Request 102: Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 180 to standard invite [Jan 30 18:44:00] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081d5ee8 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 7 (Ring+Inuse) [Jan 30 18:44:00] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:00] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:00] -- SIP/1110-081d5ee8 is ringing [Jan 30 18:44:00] <--- Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888;received=83.105.95.134 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as66750ca3 Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7169832@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #220 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7169832@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK50e9e369;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as53515c9b (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7169832", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="0dacc4857d474d293e64427f77cbd758", opaque="" (219) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 7169832@talk.orbtalk.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK50e9e369;rport From: ;tag=as53515c9b To: Call-ID: 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7169832", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="0dacc4857d474d293e64427f77cbd758", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #221 [Jan 30 18:44:00] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK50e9e369;rport=5060 From: ;tag=as53515c9b To: ;tag=13724058500c4c29465e0085181d997d.540d Call-ID: 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2149 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK50e9e369;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as53515c9b (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.540d (78) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 016ec07d16e0aeca1526e49a01d3e954@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2149 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #221 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '016ec07d16e0aeca1526e49a01d3e954@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 220 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '016ec07d16e0aeca1526e49a01d3e954@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7169915@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #224 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7169915@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0121b5d9;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as3f8d82a9 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7169915", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="b95b00c67dcca80182a0a1a277150cec", opaque="" (219) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 7169915@talk.orbtalk.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0121b5d9;rport From: ;tag=as3f8d82a9 To: Call-ID: 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7169915", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="b95b00c67dcca80182a0a1a277150cec", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #225 [Jan 30 18:44:00] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0121b5d9;rport=5060 From: ;tag=as3f8d82a9 To: ;tag=13724058500c4c29465e0085181d997d.edfb Call-ID: 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0121b5d9;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as3f8d82a9 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.edfb (78) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 68c690c351d1584304d7a00d3e3e6d62@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2148 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #225 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '68c690c351d1584304d7a00d3e3e6d62@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 224 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '68c690c351d1584304d7a00d3e3e6d62@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:7086 sip_reregister: -- Re-registration for 7167332@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4284 sip_alloc: Allocating new SIP dialog for 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 - REGISTER (No RTP) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7245 transmit_register: Scheduled a registration timeout for talk.orbtalk.co.uk id #228 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:7292 transmit_register: >>> Re-using Auth data for 7167332@talk.orbtalk.co.uk [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 (39) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1049d182;rport (63) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as62b1d420 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (36) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Authorization: Digest username="7167332", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="949f3820d9b4d2d806469690364a3872", opaque="" (219) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Expires: 120 (12) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Contact: (32) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Event: registration (19) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: Content-Length: 0 (17) [Jan 30 18:44:00] REGISTER 13 headers, 0 lines [Jan 30 18:44:00] REGISTER attempt 1 to 7167332@talk.orbtalk.co.uk [Jan 30 18:44:00] Reliably Transmitting (no NAT) to 217.14.132.178:5060: REGISTER sip:talk.orbtalk.co.uk SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1049d182;rport From: ;tag=as62b1d420 To: Call-ID: 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7167332", realm="talk.orbtalk.co.uk", algorithm=MD5, uri="sip:talk.orbtalk.co.uk", nonce="45bf8f45b3b57bda944641f6e0af458bb3c6e59b", response="949f3820d9b4d2d806469690364a3872", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #229 [Jan 30 18:44:00] <--- SIP read from 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1049d182;rport=5060 From: ;tag=as62b1d420 To: ;tag=13724058500c4c29465e0085181d997d.11f0 Call-ID: 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 CSeq: 104 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.2 (i386/linux)) Content-Length: 0 Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2145 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" <-------------> [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK1049d182;rport=5060 (68) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: ;tag=as62b1d420 (53) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=13724058500c4c29465e0085181d997d.11f0 (78) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 49c55cb9562471da5bd3553f56ea15ce@84.234.24.35 (54) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 104 REGISTER (18) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Contact: ;expires=120 (44) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Server: Sip EXpress router (0.9.2 (i386/linux)) (47) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Content-Length: 0 (17) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Warning: 392 217.14.132.178:5060 "Noisy feedback tells: pid=2145 req_src_ip=84.234.24.35 req_src_port=5060 in_uri=sip:talk.orbtalk.co.uk out_uri=sip:talk.orbtalk.co.uk via_cnt==1" (180) [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:00] --- (10 headers 0 lines) --- [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #229 [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '49c55cb9562471da5bd3553f56ea15ce@84.234.24.35' of Request 104: Match Not Found [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11982 handle_response_register: Registration successful [Jan 30 18:44:00] DEBUG[10962]: chan_sip.c:11985 handle_response_register: Cancelling timeout 228 [Jan 30 18:44:00] Scheduling destruction of SIP dialog '49c55cb9562471da5bd3553f56ea15ce@84.234.24.35' in 32000 ms (Method: REGISTER) [Jan 30 18:44:00] NOTICE[10962]: chan_sip.c:12037 handle_response_register: Outbound Registration: Expiry for talk.orbtalk.co.uk is 120 sec (Scheduling reregistration in 105 s) [Jan 30 18:44:01] <--- SIP read from 83.105.95.134:43652 ---> CANCEL sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 CANCEL User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: CANCEL sip:1110@www.jb-consultancy.com SIP/2.0 (46) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 (63) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=f95dfaf37b93c84e (81) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (37) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 665d922ff2e21edf@10.0.0.213 (36) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 41449 CANCEL (18) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (21) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:01] --- (10 headers 0 lines) --- [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL [Jan 30 18:44:01] Sending to 83.105.95.134 : 43652 (NAT) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:1631 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 665d922ff2e21edf@10.0.0.213 [Jan 30 18:44:01] <--- Reliably Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888;received=83.105.95.134 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as66750ca3 Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #232 [Jan 30 18:44:01] <--- Transmitting (NAT) to 83.105.95.134:43652 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888;received=83.105.95.134 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as66750ca3 Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 30 18:44:01] DEBUG[10986]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 30 18:44:01] DEBUG[10986]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/1110-081d5ee8' [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3296 sip_hangup: Hangup call SIP/1110-081d5ee8, SIP callid 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35) [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3304 sip_hangup: update_call_counter(1110) - decrement call limit counter on hangup [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3040 update_call_counter: Call to peer '1110' removed from call limit 200 [Jan 30 18:44:01] DEBUG[10986]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3317 sip_hangup: Hanging up channel in state Ringing (not UP) [Jan 30 18:44:01] Scheduling destruction of SIP dialog '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' in 7040 ms (Method: INVITE) [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:2069 __sip_ack: Acked pending invite 102 [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:44:01] Reliably Transmitting (NAT) to 83.105.95.134:44868: CANCEL sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport From: "James B Warrington" ;tag=as363e406c To: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:1973 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #234 [Jan 30 18:44:01] Scheduling destruction of SIP dialog '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' in 7040 ms (Method: INVITE) [Jan 30 18:44:01] DEBUG[10986]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110-081d5ee8 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 2 (In use) [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:01] DEBUG[10986]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 30 18:44:01] DEBUG[10986]: pbx.c:2368 __ast_pbx_run: Spawn extension (int_jbc,1110,3) exited non-zero on 'SIP/1001-081c4210' [Jan 30 18:44:01] == Spawn extension (int_jbc, 1110, 3) exited non-zero on 'SIP/1001-081c4210' [Jan 30 18:44:01] DEBUG[10986]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/1001-081c4210' [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3296 sip_hangup: Hangup call SIP/1001-081c4210, SIP callid 665d922ff2e21edf@10.0.0.213) [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3304 sip_hangup: update_call_counter(1001) - decrement call limit counter on hangup [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:2994 update_call_counter: Updating call counter for incoming call [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3040 update_call_counter: Call from peer '1001' removed from call limit 200 [Jan 30 18:44:01] DEBUG[10986]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 1 (Not in use) [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:01] DEBUG[10986]: chan_sip.c:3317 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 30 18:44:01] DEBUG[10986]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081c4210 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 1 (Not in use) [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1001 [Jan 30 18:44:01] <--- SIP read from 83.105.95.134:43652 ---> ACK sip:1110@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 From: "James B Warrington" ;tag=f95dfaf37b93c84e To: ;tag=as66750ca3 Contact: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="43fac959", response="8e9236463f44a5ce216adc14db261ce8" Call-ID: 665d922ff2e21edf@10.0.0.213 CSeq: 41449 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: ACK sip:1110@www.jb-consultancy.com SIP/2.0 (43) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:4068;branch=z9hG4bKcdc50a567e095888 (63) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "James B Warrington" ;tag=f95dfaf37b93c84e (81) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: ;tag=as66750ca3 (52) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Contact: (35) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1110@www.jb-consultancy.com", nonce="43fac959", response="8e9236463f44a5ce216adc14db261ce8" (178) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Call-ID: 665d922ff2e21edf@10.0.0.213 (36) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: CSeq: 41449 ACK (15) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 11: Content-Length: 0 (21) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 12: (0) [Jan 30 18:44:01] --- (12 headers 0 lines) --- [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:14538 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #232 [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '665d922ff2e21edf@10.0.0.213' of Response 41449: Match Not Found [Jan 30 18:44:01] Really destroying SIP dialog '665d922ff2e21edf@10.0.0.213' Method: ACK [Jan 30 18:44:01] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 200 OK Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 CANCEL From: "James B Warrington" ;tag=as363e406c To: ;tag=f866b25df2771ce Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport Content-Length: 0 Contact: Supported: replaces User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 (54) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 CANCEL (16) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as363e406c (65) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=f866b25df2771ce (45) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport (63) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Contact: (30) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: Supported: replaces (19) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:01] --- (10 headers 0 lines) --- [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2077 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #234 [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' of Request 102: Match Not Found [Jan 30 18:44:01] <--- SIP read from 83.105.95.134:44868 ---> SIP/2.0 487 Request Terminated Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 INVITE From: "James B Warrington" ;tag=as363e406c To: ;tag=f866b25df2771ce Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 (54) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: CSeq: 102 INVITE (16) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: From: "James B Warrington" ;tag=as363e406c (65) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: To: ;tag=f866b25df2771ce (45) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport (63) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Content-Length: 0 (17) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.1077 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: (0) [Jan 30 18:44:01] --- (8 headers 0 lines) --- [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2087 __sip_ack: Stopping retransmission on '7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35' of Request 102: Match Found [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:11578 handle_response_invite: SIP response 487 to standard invite [Jan 30 18:44:01] Transmitting (NAT) to 83.105.95.134:44868: ACK sip:1110@10.0.0.220 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK4bb373aa;rport From: "James B Warrington" ;tag=as363e406c To: ;tag=f866b25df2771ce Contact: Call-ID: 7d30ab9e6601e3ce04245d1d6c8e7e65@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:2994 update_call_counter: Updating call counter for outgoing call [Jan 30 18:44:01] DEBUG[10962]: chan_sip.c:3040 update_call_counter: Call to peer '1110' removed from call limit 200 [Jan 30 18:44:01] DEBUG[10962]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1110 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:287 do_state_change: Changing state for SIP/1110 - state 1 (Not in use) [Jan 30 18:44:01] DEBUG[10959]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1110 [Jan 30 18:44:01] DEBUG[10959]: chan_sip.c:15133 sip_devicestate: Checking device state for peer 1110 show hints -= Registered Asterisk Dial Plan Hints =- 1301@internal_all : SIP/1301 State:Unavailable Watchers 0 1300@internal_all : SIP/1300 State:Idle Watchers 0 1202@internal_all : SIP/1202 State:Idle Watchers 0 1201@internal_all : SIP/1201 State:Idle Watchers 0 1200@internal_all : SIP/1200 State:Idle Watchers 0 1110@internal_all : SIP/1110 State:Idle Watchers 0 1100@internal_all : SIP/1100 State:Idle Watchers 0 1003@internal_all : SIP/1003 State:Idle Watchers 0 1002@internal_all : SIP/1002 State:Unavailable Watchers 0 1001@internal_all : SIP/1001 State:Idle Watchers 0 1000@internal_all : SIP/1000 State:Idle Watchers 0 ---------------- - 11 hints registered *CLI> [Jan 30 18:44:03] <--- SIP read from 89.213.46.56:65406 ---> NOTIFY sip:www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-88b6fb91 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11173 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 0: NOTIFY sip:www.jb-consultancy.com SIP/2.0 (41) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-88b6fb91 (59) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 2: From: "Ext 1000" ;tag=9404e41c296a3eb8o0 (73) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 3: To: (32) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 4: Call-ID: 75a4866c-d81e80a8@192.168.1.211 (40) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 5: CSeq: 11173 NOTIFY (18) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 7: Event: keep-alive (17) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 8: User-Agent: Linksys/SPA941-5.1.5 (32) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 9: Content-Length: 0 (17) [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:4547 parse_request: Header 10: (0) [Jan 30 18:44:03] --- (10 headers 0 lines) --- [Jan 30 18:44:03] <--- Transmitting (no NAT) to 89.213.46.56:65406 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5060;branch=z9hG4bK-88b6fb91;received=89.213.46.56 From: "Ext 1000" ;tag=9404e41c296a3eb8o0 To: ;tag=as174f4b1f Call-ID: 75a4866c-d81e80a8@192.168.1.211 CSeq: 11173 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 30 18:44:03] DEBUG[10962]: chan_sip.c:14716 sipsock_read: Invalid SIP message - rejected , no callid, len 367 stop now [Jan 30 18:44:05] Beginning asterisk shutdown.... [Jan 30 18:44:05] Executing last minute cleanups [Jan 30 18:44:05] == Destroying musiconhold processes [Jan 30 18:44:05] Asterisk cleanly ending (0). [Jan 30 18:44:05] DEBUG[10954]: asterisk.c:1192 quit_handler: Asterisk ending (0).