LINUX3*CLI> LINUX3*CLI> LINUX3*CLI> LINUX3*CLI> sip set debug on SIP Debugging enabled Retransmitting #2 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK04f11b26;rport Max-Forwards: 70 From: "asterisk" ;tag=as2d1eb558 To: Contact: Call-ID: 41c4fc02370edae41fcdcfe42b813833@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7147b2a5;rport Max-Forwards: 70 From: "asterisk" ;tag=as6c35085a To: Contact: Call-ID: 63a614d6709a04371be2b8d04e44e717@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK2586e4f2;rport Max-Forwards: 70 From: "asterisk" ;tag=as057a0502 To: Contact: Call-ID: 1b449a160a2a19154dd1a4021d0521f4@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK04f11b26;rport Max-Forwards: 70 From: "asterisk" ;tag=as2d1eb558 To: Contact: Call-ID: 41c4fc02370edae41fcdcfe42b813833@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7147b2a5;rport Max-Forwards: 70 From: "asterisk" ;tag=as6c35085a To: Contact: Call-ID: 63a614d6709a04371be2b8d04e44e717@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK2586e4f2;rport Max-Forwards: 70 From: "asterisk" ;tag=as057a0502 To: Contact: Call-ID: 1b449a160a2a19154dd1a4021d0521f4@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK04f11b26;rport Max-Forwards: 70 From: "asterisk" ;tag=as2d1eb558 To: Contact: Call-ID: 41c4fc02370edae41fcdcfe42b813833@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '41c4fc02370edae41fcdcfe42b813833@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7147b2a5;rport Max-Forwards: 70 From: "asterisk" ;tag=as6c35085a To: Contact: Call-ID: 63a614d6709a04371be2b8d04e44e717@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '63a614d6709a04371be2b8d04e44e717@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK2586e4f2;rport Max-Forwards: 70 From: "asterisk" ;tag=as057a0502 To: Contact: Call-ID: 1b449a160a2a19154dd1a4021d0521f4@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '1b449a160a2a19154dd1a4021d0521f4@173.162.11.91' Method: OPTIONS Really destroying SIP dialog '3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91' Method: REGISTER Really destroying SIP dialog 'f67039f98092d50187140da40fd79003@173.162.11.91' Method: REGISTER Reliably Transmitting (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK300620f9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae62020 To: Contact: Call-ID: 7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK50c14c02;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e3b0f1b To: Contact: Call-ID: 2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7ce2e9a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as473fe8e9 To: Contact: Call-ID: 6b66ba39699711d66d669ea92057a91b@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK300620f9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae62020 To: Contact: Call-ID: 7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK50c14c02;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e3b0f1b To: Contact: Call-ID: 2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7ce2e9a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as473fe8e9 To: Contact: Call-ID: 6b66ba39699711d66d669ea92057a91b@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK300620f9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae62020 To: Contact: Call-ID: 7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK50c14c02;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e3b0f1b To: Contact: Call-ID: 2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7ce2e9a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as473fe8e9 To: Contact: Call-ID: 6b66ba39699711d66d669ea92057a91b@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK300620f9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae62020 To: Contact: Call-ID: 7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK50c14c02;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e3b0f1b To: Contact: Call-ID: 2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7ce2e9a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as473fe8e9 To: Contact: Call-ID: 6b66ba39699711d66d669ea92057a91b@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK300620f9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6ae62020 To: Contact: Call-ID: 7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '7dcf6fb829a0f8ba3b12ad6655515cc5@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK50c14c02;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e3b0f1b To: Contact: Call-ID: 2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '2cda02fa1791f0a815e1d90f31e7996c@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7ce2e9a6;rport Max-Forwards: 70 From: "asterisk" ;tag=as473fe8e9 To: Contact: Call-ID: 6b66ba39699711d66d669ea92057a91b@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '6b66ba39699711d66d669ea92057a91b@173.162.11.91' Method: OPTIONS Reliably Transmitting (no NAT) to 209.62.1.2:5060: OPTIONS sip:209.62.1.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK27768838;rport Max-Forwards: 70 From: "asterisk" ;tag=as76d8e9be To: Contact: Call-ID: 27f24df62b9f47d12b91114275b43696@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:209.62.1.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK27768838;received=173.162.11.91;rport=5060 From: "asterisk" ;tag=as76d8e9be To: ;tag=as520b091d Call-ID: 27f24df62b9f47d12b91114275b43696@173.162.11.91 CSeq: 102 OPTIONS User-Agent: VoIPMS/SERAST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '27f24df62b9f47d12b91114275b43696@173.162.11.91' Method: OPTIONS Really destroying SIP dialog '819b0edda535ec353b6abd982b3acade@173.162.11.91' Method: ACK <--- SIP read from UDP:173.162.11.83:5060 ---> INVITE sip:12392449062@173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bKd893cbad0 Content-Length: 246 To: sip:12392449062@173.162.11.91 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319996 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: sip:mediatrix1@173.162.11.83 Supported: replaces User-Agent: MxSipApp/5.0.27.210 v=0 o=MxSIP 1671524399255226255 1555567423193077049 IN IP4 173.162.11.83 s=- c=IN IP4 173.162.11.83 t=0 0 a=sendrecv m=audio 5004 RTP/AVP 0 18 4 8 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 <-------------> --- (15 headers 11 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to 173.162.11.83 : 5060 (no NAT) Using INVITE request as basis request - 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 Found peer 'mediatrix1' for 'mediatrix1' from 173.162.11.83:5060 <--- Reliably Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bKd893cbad0;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd To: sip:12392449062@173.162.11.91;tag=as796e9c3e Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319996 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="664e8c2f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91' in 6400 ms (Method: INVITE) <--- SIP read from UDP:173.162.11.83:5060 ---> ACK sip:12392449062@173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bKd893cbad0 Content-Length: 0 To: sip:12392449062@173.162.11.91;tag=as796e9c3e From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319996 ACK User-Agent: MxSipApp/5.0.27.210 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:173.162.11.83:5060 ---> INVITE sip:12392449062@173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK7d71fe963 Content-Length: 246 To: sip:12392449062@173.162.11.91 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319997 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Contact: sip:mediatrix1@173.162.11.83 Content-Type: application/sdp Authorization:Digest response="e991480646186d77cd3ca9b91c72e9ea",username="mediatrix1",realm="asterisk",nonce="664e8c2f",algorithm=MD5,uri="sip:12392449062@173.162.11.91" Supported: replaces User-Agent: MxSipApp/5.0.27.210 v=0 o=MxSIP 1671524399255226255 1555567423193077049 IN IP4 173.162.11.83 s=- c=IN IP4 173.162.11.83 t=0 0 a=sendrecv m=audio 5004 RTP/AVP 0 18 4 8 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 <-------------> --- (16 headers 11 lines) --- Sending to 173.162.11.83 : 5060 (no NAT) Using INVITE request as basis request - 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 Found peer 'mediatrix1' for 'mediatrix1' from 173.162.11.83:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 13 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing) Peer audio RTP is at port 173.162.11.83:5004 Looking for 12392449062 in jamko_phones (domain 173.162.11.91) list_route: hop: <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK7d71fe963;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd To: sip:12392449062@173.162.11.91 Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319997 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> -- Executing [12392449062@jamko_phones:1] Set("SIP/mediatrix1-00000067", "SPYGROUP=10005") in new stack -- Executing [12392449062@jamko_phones:2] Dial("SIP/mediatrix1-00000067", "SIP/12392449062@my_service_provider_4,120") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 173.162.11.91 port 15964 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 67.216.35.162:5060: INVITE sip:12392449062@67.216.35.162 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;rport Max-Forwards: 70 From: "JamKo Fax" ;tag=as5d1e0077 To: Contact: Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.10 Remote-Party-ID: "JamKo Fax" ;privacy=off;screen=no Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 237 v=0 o=root 343635226 343635226 IN IP4 173.162.11.91 s=Asterisk PBX 1.6.2.10 c=IN IP4 173.162.11.91 t=0 0 m=audio 15964 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 12392449062@my_service_provider_4 <--- SIP read from UDP:67.216.35.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;received=173.162.11.91;rport=5060 From: "JamKo Fax" ;tag=as5d1e0077 To: Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE Server: Gafachi UAS v110.09 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Reliably Transmitting (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK5707a94a;rport Max-Forwards: 70 From: "asterisk" ;tag=as490074c7 To: Contact: Call-ID: 20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7d57bbbb;rport Max-Forwards: 70 From: "asterisk" ;tag=as271267f9 To: Contact: Call-ID: 6b47f3963431c5ef0912b57a22d56f13@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK15d5de1e;rport Max-Forwards: 70 From: "asterisk" ;tag=as223dae15 To: Contact: Call-ID: 2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK5707a94a;rport Max-Forwards: 70 From: "asterisk" ;tag=as490074c7 To: Contact: Call-ID: 20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7d57bbbb;rport Max-Forwards: 70 From: "asterisk" ;tag=as271267f9 To: Contact: Call-ID: 6b47f3963431c5ef0912b57a22d56f13@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK15d5de1e;rport Max-Forwards: 70 From: "asterisk" ;tag=as223dae15 To: Contact: Call-ID: 2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK5707a94a;rport Max-Forwards: 70 From: "asterisk" ;tag=as490074c7 To: Contact: Call-ID: 20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7d57bbbb;rport Max-Forwards: 70 From: "asterisk" ;tag=as271267f9 To: Contact: Call-ID: 6b47f3963431c5ef0912b57a22d56f13@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK15d5de1e;rport Max-Forwards: 70 From: "asterisk" ;tag=as223dae15 To: Contact: Call-ID: 2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.35.162:5060: OPTIONS sip:67.216.35.162 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK2a780c7b;rport Max-Forwards: 70 From: "asterisk" ;tag=as381b221b To: Contact: Call-ID: 282b4c2b7896ffc96dbec32940eff150@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:67.216.35.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK2a780c7b From: "asterisk" ;tag=as381b221b To: ;tag=fegss Call-ID: 282b4c2b7896ffc96dbec32940eff150@173.162.11.91 CSeq: 102 OPTIONS Server: Gafachi UAS v110.09 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '282b4c2b7896ffc96dbec32940eff150@173.162.11.91' Method: OPTIONS Retransmitting #3 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK5707a94a;rport Max-Forwards: 70 From: "asterisk" ;tag=as490074c7 To: Contact: Call-ID: 20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7d57bbbb;rport Max-Forwards: 70 From: "asterisk" ;tag=as271267f9 To: Contact: Call-ID: 6b47f3963431c5ef0912b57a22d56f13@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK15d5de1e;rport Max-Forwards: 70 From: "asterisk" ;tag=as223dae15 To: Contact: Call-ID: 2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK5707a94a;rport Max-Forwards: 70 From: "asterisk" ;tag=as490074c7 To: Contact: Call-ID: 20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '20f7cfd263505d8976a1d3f50bf1bab2@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK7d57bbbb;rport Max-Forwards: 70 From: "asterisk" ;tag=as271267f9 To: Contact: Call-ID: 6b47f3963431c5ef0912b57a22d56f13@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '6b47f3963431c5ef0912b57a22d56f13@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK15d5de1e;rport Max-Forwards: 70 From: "asterisk" ;tag=as223dae15 To: Contact: Call-ID: 2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:54:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '2b6c0738760a45ab0509b4cf2cd41949@173.162.11.91' Method: OPTIONS <--- SIP read from UDP:67.216.35.162:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;received=173.162.11.91;rport=5060 From: "JamKo Fax" ;tag=as5d1e0077 To: ;tag=gss6e365d32d624300343ac470738800c92 Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE Server: Gafachi UAS v110.09 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 381044115 381044115 IN IP4 67.216.38.3 s=session c=IN IP4 67.216.38.3 t=0 0 m=audio 46360 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 67.216.38.3:46360 -- SIP/my_service_provider_4-00000068 is making progress passing it to SIP/mediatrix1-00000067 Audio is at 173.162.11.91 port 17310 Adding codec 0x4 (ulaw) to SDP <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK7d71fe963;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd To: sip:12392449062@173.162.11.91;tag=as6e91efe9 Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319997 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=root 209194343 209194343 IN IP4 173.162.11.91 s=Asterisk PBX 1.6.2.10 c=IN IP4 173.162.11.91 t=0 0 m=audio 17310 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:67.216.35.162:5060 ---> INVITE sip:2396346540@173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 67.216.35.162:5060;branch=z9hG4bKd8f71e3f From: ;tag=gss6e365d32d624300343ac470738800c92 To: "JamKo Fax" ;tag=as5d1e0077 Contact: Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE User-Agent: Gafachi UAC v110.09 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 281 v=0 o=root 381044115 381044116 IN IP4 67.216.38.3 s=session c=IN IP4 67.216.38.3 t=0 0 m=image 46362 udptl t38 a=T38FaxVersion:0 a=T38FaxRateManagement:transferredTCFlocalTCF a=T38FaxUdpEC:t38UDPFEC a=T38FaxMaxBufferSize:2000 a=T38MaxDatagram:512 a=T38FaxMaxRate:14400 <-------------> --- (12 headers 12 lines) --- <--- Transmitting (no NAT) to 67.216.35.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.216.35.162:5060;branch=z9hG4bKd8f71e3f;received=67.216.35.162 From: ;tag=gss6e365d32d624300343ac470738800c92 To: "JamKo Fax" ;tag=as5d1e0077 Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Now forwarding SIP/mediatrix1-00000067 to 'Local/2396346540@incoming_calls' (thanks to SIP/my_service_provider_4-00000068) [Aug 11 09:55:07] NOTICE[29924]: chan_local.c:534 local_call: No such extension/context 2396346540@incoming_calls while calling Local channel [Aug 11 09:55:07] NOTICE[29924]: app_dial.c:789 do_forward: Failed to dial on local channel for call forward to '2396346540@incoming_calls' Scheduling destruction of SIP dialog '64325d454b4b0a567e7095b90049cd1c@173.162.11.91' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 67.216.35.162:5060: CANCEL sip:12392449062@67.216.35.162 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;rport Max-Forwards: 70 From: "JamKo Fax" ;tag=as5d1e0077 To: Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.2.10 Remote-Party-ID: "JamKo Fax" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '64325d454b4b0a567e7095b90049cd1c@173.162.11.91' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/mediatrix1-00000067' status is 'CHANUNAVAIL' <--- Reliably Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK7d71fe963;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd To: sip:12392449062@173.162.11.91;tag=as6e91efe9 Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319997 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Content-Length: 0 <------------> <--- SIP read from UDP:67.216.35.162:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 67.216.35.162:5060;branch=z9hG4bKd8f71e3f;received=173.162.11.91 From: ;tag=gss6e365d32d624300343ac470738800c92 To: "JamKo Fax" ;tag=as5d1e0077 Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 INVITE Server: Gafachi UAS v110.09 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 67.216.35.162:5060: ACK sip:12392449062@67.216.35.162 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;rport Max-Forwards: 70 From: "JamKo Fax" ;tag=as5d1e0077 To: ;tag=as5d1e0077 Contact: Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.10 Remote-Party-ID: "JamKo Fax" ;privacy=off;screen=no Content-Length: 0 --- <--- SIP read from UDP:67.216.35.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK57975633;received=173.162.11.91;rport=5060 From: "JamKo Fax" ;tag=as5d1e0077 To: ;tag=as5d1e0077 Call-ID: 64325d454b4b0a567e7095b90049cd1c@173.162.11.91 CSeq: 102 CANCEL Server: Gafachi UAS v110.09 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '64325d454b4b0a567e7095b90049cd1c@173.162.11.91' Method: INVITE <--- SIP read from UDP:173.162.11.83:5060 ---> ACK sip:12392449062@173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK7d71fe963 Content-Length: 0 To: sip:12392449062@173.162.11.91;tag=as6e91efe9 From: sip:mediatrix1@173.162.11.91;tag=d1e1d72de89b9cd Call-ID: 2d0afd74eb0c928e3d8cc76b5463580f@173.162.11.91 CSeq: 1066319997 ACK Authorization:Digest response="e991480646186d77cd3ca9b91c72e9ea",username="mediatrix1",realm="asterisk",nonce="664e8c2f",algorithm=MD5,uri="sip:12392449062@173.162.11.91" User-Agent: MxSipApp/5.0.27.210 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:173.162.11.83:5060 ---> REGISTER sip:173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK17432cf24 Content-Length: 0 To: sip:mediatrix1@173.162.11.91 From: sip:mediatrix1@173.162.11.91;tag=92088682140cebc Call-ID: 3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91 CSeq: 273975632 REGISTER Contact: sip:mediatrix1@173.162.11.83 Authorization:Digest response="ab454897a15820dcf9c90b2f12e2c5c1",username="mediatrix1",realm="asterisk",nonce="4a7a5ac5",algorithm=MD5,uri="sip:173.162.11.91" User-Agent: MxSipApp/5.0.27.210 <-------------> --- (10 headers 0 lines) --- Sending to 173.162.11.83 : 5060 (no NAT) <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK17432cf24;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=92088682140cebc To: sip:mediatrix1@173.162.11.91;tag=as4156f936 Call-ID: 3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91 CSeq: 273975632 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="081cfc37" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:173.162.11.83:5060 ---> REGISTER sip:173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK9c46b2604 Content-Length: 0 To: sip:mediatrix2@173.162.11.91 From: sip:mediatrix2@173.162.11.91;tag=ac95b289ea31aa6 Call-ID: f67039f98092d50187140da40fd79003@173.162.11.91 CSeq: 490847926 REGISTER Contact: sip:mediatrix2@173.162.11.83 Authorization:Digest response="2cc85ec3f955c776463532b744a03f61",username="mediatrix2",realm="asterisk",nonce="7b634543",algorithm=MD5,uri="sip:173.162.11.91" User-Agent: MxSipApp/5.0.27.210 <-------------> --- (10 headers 0 lines) --- Sending to 173.162.11.83 : 5060 (no NAT) <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK9c46b2604;received=173.162.11.83 From: sip:mediatrix2@173.162.11.91;tag=ac95b289ea31aa6 To: sip:mediatrix2@173.162.11.91;tag=as6b4f3afd Call-ID: f67039f98092d50187140da40fd79003@173.162.11.91 CSeq: 490847926 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38815e3f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'f67039f98092d50187140da40fd79003@173.162.11.91' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:173.162.11.83:5060 ---> REGISTER sip:173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bKb31ccf4af Content-Length: 0 To: sip:mediatrix1@173.162.11.91 From: sip:mediatrix1@173.162.11.91;tag=92088682140cebc Call-ID: 3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91 CSeq: 273975633 REGISTER Contact: sip:mediatrix1@173.162.11.83 Authorization:Digest response="35d50e49b8da4138f2b020efdba21c7e",username="mediatrix1",realm="asterisk",nonce="081cfc37",algorithm=MD5,uri="sip:173.162.11.91" User-Agent: MxSipApp/5.0.27.210 <-------------> --- (10 headers 0 lines) --- Sending to 173.162.11.83 : 5060 (no NAT) Reliably Transmitting (no NAT) to 173.162.11.83:5060: OPTIONS sip:mediatrix1@173.162.11.83 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK0d27bbe1;rport Max-Forwards: 70 From: "asterisk" ;tag=as54e555e8 To: Contact: Call-ID: 29a10d6105e2785055d5dfcf486bf54d@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bKb31ccf4af;received=173.162.11.83 From: sip:mediatrix1@173.162.11.91;tag=92088682140cebc To: sip:mediatrix1@173.162.11.91;tag=as4156f936 Call-ID: 3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91 CSeq: 273975633 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 11 Aug 2010 13:55:11 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3b47fa0a2420b0caeaefee1da98a0e56@173.162.11.91' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:173.162.11.83:5060 ---> REGISTER sip:173.162.11.91 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK438395297 Content-Length: 0 To: sip:mediatrix2@173.162.11.91 From: sip:mediatrix2@173.162.11.91;tag=ac95b289ea31aa6 Call-ID: f67039f98092d50187140da40fd79003@173.162.11.91 CSeq: 490847927 REGISTER Contact: sip:mediatrix2@173.162.11.83 Authorization:Digest response="2c6c60f713d72fca20bd8010fcf02352",username="mediatrix2",realm="asterisk",nonce="38815e3f",algorithm=MD5,uri="sip:173.162.11.91" User-Agent: MxSipApp/5.0.27.210 <-------------> --- (10 headers 0 lines) --- Sending to 173.162.11.83 : 5060 (no NAT) Reliably Transmitting (no NAT) to 173.162.11.83:5060: OPTIONS sip:mediatrix2@173.162.11.83 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK018b3a7d;rport Max-Forwards: 70 From: "asterisk" ;tag=as0a0c61ea To: Contact: Call-ID: 55490bdd212d424b5aef28db1976b3e2@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 173.162.11.83:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 173.162.11.83;branch=z9hG4bK438395297;received=173.162.11.83 From: sip:mediatrix2@173.162.11.91;tag=ac95b289ea31aa6 To: sip:mediatrix2@173.162.11.91;tag=as6b4f3afd Call-ID: f67039f98092d50187140da40fd79003@173.162.11.91 CSeq: 490847927 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 11 Aug 2010 13:55:11 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'f67039f98092d50187140da40fd79003@173.162.11.91' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:173.162.11.83:5060 ---> SIP/2.0 405 Method Not Allowed Call-ID: 29a10d6105e2785055d5dfcf486bf54d@173.162.11.91 CSeq: 102 OPTIONS From: "asterisk" ;tag=as54e555e8 To: ;tag=ab3cf2bf3f7f970 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK0d27bbe1;rport Content-Length: 0 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY User-Agent: MxSipApp/5.0.27.210 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '29a10d6105e2785055d5dfcf486bf54d@173.162.11.91' Method: OPTIONS <--- SIP read from UDP:173.162.11.83:5060 ---> SIP/2.0 405 Method Not Allowed Call-ID: 55490bdd212d424b5aef28db1976b3e2@173.162.11.91 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0a0c61ea To: ;tag=9d940b1bf8ba29c Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK018b3a7d;rport Content-Length: 0 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY User-Agent: MxSipApp/5.0.27.210 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '55490bdd212d424b5aef28db1976b3e2@173.162.11.91' Method: OPTIONS Reliably Transmitting (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK74a71c16;rport Max-Forwards: 70 From: "asterisk" ;tag=as78b1acc6 To: Contact: Call-ID: 415ee21372c9e20d76c80c5e6d97024c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK367566ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as62fb76af To: Contact: Call-ID: 7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK69ddde6b;rport Max-Forwards: 70 From: "asterisk" ;tag=as619084b0 To: Contact: Call-ID: 483be64705b50ee0747f6fab22ddba05@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK74a71c16;rport Max-Forwards: 70 From: "asterisk" ;tag=as78b1acc6 To: Contact: Call-ID: 415ee21372c9e20d76c80c5e6d97024c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK367566ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as62fb76af To: Contact: Call-ID: 7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK69ddde6b;rport Max-Forwards: 70 From: "asterisk" ;tag=as619084b0 To: Contact: Call-ID: 483be64705b50ee0747f6fab22ddba05@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK74a71c16;rport Max-Forwards: 70 From: "asterisk" ;tag=as78b1acc6 To: Contact: Call-ID: 415ee21372c9e20d76c80c5e6d97024c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK367566ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as62fb76af To: Contact: Call-ID: 7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #2 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK69ddde6b;rport Max-Forwards: 70 From: "asterisk" ;tag=as619084b0 To: Contact: Call-ID: 483be64705b50ee0747f6fab22ddba05@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK74a71c16;rport Max-Forwards: 70 From: "asterisk" ;tag=as78b1acc6 To: Contact: Call-ID: 415ee21372c9e20d76c80c5e6d97024c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK367566ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as62fb76af To: Contact: Call-ID: 7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK69ddde6b;rport Max-Forwards: 70 From: "asterisk" ;tag=as619084b0 To: Contact: Call-ID: 483be64705b50ee0747f6fab22ddba05@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 67.216.38.3:5060: OPTIONS sip:67.216.38.3 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK74a71c16;rport Max-Forwards: 70 From: "asterisk" ;tag=as78b1acc6 To: Contact: Call-ID: 415ee21372c9e20d76c80c5e6d97024c@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '415ee21372c9e20d76c80c5e6d97024c@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.2:5060: OPTIONS sip:67.216.38.2 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK367566ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as62fb76af To: Contact: Call-ID: 7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '7c681b7d33c4db172ad0bdfe15311afc@173.162.11.91' Method: OPTIONS Retransmitting #4 (no NAT) to 67.216.38.4:5060: OPTIONS sip:67.216.38.4 SIP/2.0 Via: SIP/2.0/UDP 173.162.11.91:5060;branch=z9hG4bK69ddde6b;rport Max-Forwards: 70 From: "asterisk" ;tag=as619084b0 To: Contact: Call-ID: 483be64705b50ee0747f6fab22ddba05@173.162.11.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.10 Date: Wed, 11 Aug 2010 13:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '483be64705b50ee0747f6fab22ddba05@173.162.11.91' Method: OPTIONS LINUX3*CLI> sip set debug off SIP Debugging Disabled LINUX3*CLI> core set debug 0 Core debug is now OFF LINUX3*CLI>