asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> sip set debug on SIP Debugging enabled asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5065 ---> INVITE sip:407971296525@87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK4d6d3d47 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Date: Mon, 12 Jan 2009 18:04:48 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: "1000" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 20241 17945 IN IP4 92.63.131.41 s=SIP Call c=IN IP4 92.63.131.41 t=0 0 m=audio 31228 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- == Using SIP RTP CoS mark 5 Sending to 92.63.131.41 : 5065 (no NAT) Using INVITE request as basis request - 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Found peer '1000' for '1000' from 92.63.131.41:5065 asterisk*CLI> <--- Reliably Transmitting (NAT) to 92.63.131.41:5065 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK4d6d3d47;received=92.63.131.41 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as4ab3a6a3 Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 CSeq: 101 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="683ff1e3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5065 ---> ACK sip:407971296525@87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4d6d3d47 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as4ab3a6a3 Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Date: Mon, 12 Jan 2009 18:04:48 GMT CSeq: 101 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP://92.63.131.41:5065 ---> INVITE sip:407971296525@87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Date: Mon, 12 Jan 2009 18:04:49 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Authorization: Digest username="1000",realm="asterisk",uri="sip:87.237.62.235",response="eed1272900a83b0462dd689261f576dc",nonce="683ff1e3",algorithm=MD5 Expires: 180 Content-Type: application/sdp Content-Length: 247 Remote-Party-ID: "1000" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 20241 17945 IN IP4 92.63.131.41 s=SIP Call c=IN IP4 92.63.131.41 t=0 0 m=audio 31228 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- Sending to 92.63.131.41 : 5065 (NAT) Using INVITE request as basis request - 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Found peer '1000' for '1000' from 92.63.131.41:5065 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 92.63.131.41:31228 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 92.63.131.41:31228 Looking for 407971296525 in ricks_room (domain 87.237.62.235) list_route: hop: <--- Transmitting (NAT) to 92.63.131.41:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2;received=92.63.131.41 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [407971296525@ricks_room:1] MixMonitor("SIP/1000-c8005508", "07971296525-20090112_180543.wav") in new stack -- Executing [407971296525@ricks_room:2] Set("SIP/1000-c8005508", "CALLERID(number)=08449109020") in new stack -- Executing [407971296525@ricks_room:3] Set("SIP/1000-c8005508", "CDR(accountcode)=rick_out") in new stack -- Executing [407971296525@ricks_room:4] Dial("SIP/1000-c8005508", "SIP/magrathea-outbound/07971296525") in new stack == Using SIP RTP CoS mark 5 Audio is at 87.237.62.235 port 13842 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.166.5.135:5060: INVITE sip:07971296525@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport Max-Forwards: 70 From: "R Hamnett" ;tag=as21efe42f To: Contact: Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Remote-Party-ID: "R Hamnett" ;privacy=off;screen=no Date: Mon, 12 Jan 2009 18:05:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 329 v=0 o=root 1867352154 1867352154 IN IP4 87.237.62.235 s=Asterisk PBX SVN-branch-1.6.1-r167727 c=IN IP4 87.237.62.235 t=0 0 m=audio 13842 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- -- Called magrathea-outbound/07971296525 == Begin MixMonitor Recording SIP/1000-c8005508 asterisk*CLI> <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport=5060 From: "R Hamnett" ;tag=as21efe42f To: Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net CSeq: 102 INVITE Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Retransmitting #4 (NAT) to 92.63.131.41:5065: NOTIFY sip:1000@92.63.131.41:5065 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK7baf39ba;rport Max-Forwards: 70 From: "asterisk" ;tag=as07ebed2c To: Contact: Call-ID: 29f940c4693186d439237fa16312c30f@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- Retransmitting #4 (NAT) to 92.63.131.41:5060: NOTIFY sip:1002@92.63.131.41:5060 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK4521f9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as77a15329 To: Contact: Call-ID: 20a42d2335a7e3446c54ec7b12589f01@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5060 ---> REGISTER sip:87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK2cb2120d From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235 Call-ID: 003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208 Date: Mon, 12 Jan 2009 18:05:00 GMT CSeq: 6288 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 120 <-------------> --- (11 headers 0 lines) --- Sending to 92.63.131.41 : 5060 (no NAT) <--- Transmitting (NAT) to 92.63.131.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK2cb2120d;received=92.63.131.41 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235;tag=as31621916 Call-ID: 003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208 CSeq: 6288 REGISTER Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40ea93ae" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5060 ---> REGISTER sip:87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK08059cf4 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235 Call-ID: 003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208 Date: Mon, 12 Jan 2009 18:05:00 GMT CSeq: 6289 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="1002",realm="asterisk",uri="sip:87.237.62.235",response="8af440bf8a531d9bb4e53491a976a10a",nonce="40ea93ae",algorithm=MD5 Content-Length: 0 Expires: 120 <-------------> --- (12 headers 0 lines) --- Sending to 92.63.131.41 : 5060 (NAT) <--- Transmitting (NAT) to 92.63.131.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK08059cf4;received=92.63.131.41 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235;tag=as31621916 Call-ID: 003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208 CSeq: 6289 REGISTER Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 12 Jan 2009 18:05:47 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '003094c4-1cb90002-23a9e014-5c332cab@192.168.1.208' in 32000 ms (Method: REGISTER) Retransmitting #5 (NAT) to 92.63.131.41:5065: NOTIFY sip:1000@92.63.131.41:5065 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK7baf39ba;rport Max-Forwards: 70 From: "asterisk" ;tag=as07ebed2c To: Contact: Call-ID: 29f940c4693186d439237fa16312c30f@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- Retransmitting #5 (NAT) to 92.63.131.41:5060: NOTIFY sip:1002@92.63.131.41:5060 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK4521f9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as77a15329 To: Contact: Call-ID: 20a42d2335a7e3446c54ec7b12589f01@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5060 ---> REGISTER sip:87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK1d8ba4b1 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235 Call-ID: 003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208 Date: Mon, 12 Jan 2009 18:05:02 GMT CSeq: 6288 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 120 <-------------> --- (11 headers 0 lines) --- Sending to 92.63.131.41 : 5060 (no NAT) <--- Transmitting (NAT) to 92.63.131.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK1d8ba4b1;received=92.63.131.41 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235;tag=as06538cae Call-ID: 003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208 CSeq: 6288 REGISTER Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3eb0e65a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5060 ---> REGISTER sip:87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK745a68f2 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235 Call-ID: 003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208 Date: Mon, 12 Jan 2009 18:05:02 GMT CSeq: 6289 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="1002",realm="asterisk",uri="sip:87.237.62.235",response="af693ab55afa801530b4b404c1e30b87",nonce="3eb0e65a",algorithm=MD5 Content-Length: 0 Expires: 120 <-------------> --- (12 headers 0 lines) --- Sending to 92.63.131.41 : 5060 (NAT) <--- Transmitting (NAT) to 92.63.131.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.41:5060;branch=z9hG4bK745a68f2;received=92.63.131.41 From: sip:1002@87.237.62.235 To: sip:1002@87.237.62.235;tag=as06538cae Call-ID: 003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208 CSeq: 6289 REGISTER Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 12 Jan 2009 18:05:49 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '003094c4-1cb90003-1e30e8e9-3ba8a699@192.168.1.208' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport=5060 From: "R Hamnett" ;tag=as21efe42f To: ;tag=7660BE84-DB5 Date: Mon, 12 Jan 2009 18:04:34 gmt Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: , Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 194 v=0 o=CiscoSystemsSIP-GW-UserAgent 3082 3771 IN IP4 87.238.72.149 s=SIP Call c=IN IP4 87.238.72.149 t=0 0 m=audio 19070 RTP/AVP 0 c=IN IP4 87.238.72.149 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (15 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 87.238.72.149:19070 Found audio description format PCMU for ID 0 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 87.238.72.149:19070 -- SIP/magrathea-outbound-00a8fb58 is making progress passing it to SIP/1000-c8005508 Audio is at 87.237.62.235 port 15346 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 92.63.131.41:5065 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2;received=92.63.131.41 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as1ef0084b Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 872508525 872508525 IN IP4 87.237.62.235 s=Asterisk PBX SVN-branch-1.6.1-r167727 c=IN IP4 87.237.62.235 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 <------------> asterisk*CLI> <--- SIP read from UDP://92.63.131.41:5065 ---> CANCEL sip:407971296525@87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Date: Mon, 12 Jan 2009 18:04:58 GMT CSeq: 102 CANCEL User-Agent: CSCO/7 Content-Length: 0 Authorization: Digest username="1000",realm="asterisk",uri="sip:87.237.62.235",response="fddef43a2fd5a516bd6ed451feb37b41",nonce="683ff1e3",algorithm=MD5 <-------------> --- (10 headers 0 lines) --- Sending to 92.63.131.41 : 5065 (NAT) <--- Reliably Transmitting (NAT) to 92.63.131.41:5065 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2;received=92.63.131.41 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as1ef0084b Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (NAT) to 92.63.131.41:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.41:5065;branch=z9hG4bK27b913b2;received=92.63.131.41 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as1ef0084b Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 CSeq: 102 CANCEL Server: Asterisk PBX SVN-branch-1.6.1-r167727 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '75bf501b06017670286e29a627f741cf@gk.magrathea.net' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 213.166.5.135:5060: CANCEL sip:07971296525@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport Max-Forwards: 70 From: "R Hamnett" ;tag=as21efe42f To: Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net CSeq: 102 CANCEL User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Remote-Party-ID: "R Hamnett" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '75bf501b06017670286e29a627f741cf@gk.magrathea.net' in 6400 ms (Method: INVITE) asterisk*CLI> <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 200 cancelling Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport=5060 From: "R Hamnett" ;tag=as21efe42f To: ;tag=061301cb7eebb8209e58790d11b8846b-568d Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net CSeq: 102 CANCEL Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP://213.166.5.135:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport=5060 From: "R Hamnett" ;tag=as21efe42f To: ;tag=7660BE84-DB5 Date: Mon, 12 Jan 2009 18:04:44 gmt Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 213.166.5.135:5060: ACK sip:07971296525@gk.magrathea.net SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:5060;branch=z9hG4bK1ad1fac1;rport Max-Forwards: 70 From: "R Hamnett" ;tag=as21efe42f To: ;tag=7660BE84-DB5 Contact: Call-ID: 75bf501b06017670286e29a627f741cf@gk.magrathea.net CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Remote-Party-ID: "R Hamnett" ;privacy=off;screen=no Content-Length: 0 --- Really destroying SIP dialog '75bf501b06017670286e29a627f741cf@gk.magrathea.net' Method: INVITE == Spawn extension (ricks_room, 407971296525, 4) exited non-zero on 'SIP/1000-c8005508' == End MixMonitor Recording SIP/1000-c8005508 <--- SIP read from UDP://92.63.131.41:5065 ---> ACK sip:407971296525@87.237.62.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK27b913b2 From: "1000" ;tag=000a8a4906e3002b62673394-6d473a9f To: ;tag=as1ef0084b Call-ID: 000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41 Date: Mon, 12 Jan 2009 18:04:58 GMT CSeq: 102 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '000a8a49-06e30022-5ba22aa8-5f040ec5@92.63.131.41' Method: ACK Retransmitting #6 (NAT) to 92.63.131.41:5065: NOTIFY sip:1000@92.63.131.41:5065 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK7baf39ba;rport Max-Forwards: 70 From: "asterisk" ;tag=as07ebed2c To: Contact: Call-ID: 29f940c4693186d439237fa16312c30f@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- Retransmitting #6 (NAT) to 92.63.131.41:5060: NOTIFY sip:1002@92.63.131.41:5060 SIP/2.0 Via: SIP/2.0/UDP 87.237.62.235:0;branch=z9hG4bK4521f9a1;rport Max-Forwards: 70 From: "asterisk" ;tag=as77a15329 To: Contact: Call-ID: 20a42d2335a7e3446c54ec7b12589f01@87.237.62.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.6.1-r167727 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@87.237.62.235 Voice-Message: 0/1 (0/0) --- asterisk*CLI> sip set debug off SIP Debugging Disabled