asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> Reliably Transmitting (no NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.141:5060;branch=z9hG4bK727316dc Max-Forwards: 70 From: "asterisk" ;tag=as5db3b55b To: Contact: Call-ID: 0fb3878a45c484c2763ae4c642dba79c@10.0.0.141:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date: Thu, 06 Jan 2011 20:19:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '0fb3878a45c484c2763ae4c642dba79c@10.0.0.141:5060' Method: OPTIONS <--- SIP read from TCP:10.0.0.25:5065 ---> REFER sip:2710@10.0.0.141:5060;transport=TCP SIP/2.0 FROM: ;epid=DCDF2BE031;tag=21c28aa0b9 TO: ;tag=as5ac0bd70 CSEQ: 1 REFER CALL-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.0.0.25:5065;branch=z9hG4bK315eaa8a CONTACT: ;automata CONTENT-LENGTH: 0 REFER-TO: REFERRED-BY: USER-AGENT: RTCC/3.0.0.0 <-------------> --- (12 headers 0 lines) --- Call 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 got a SIP call transfer from callee: (REFER)! SIP transfer to extension 2711@internal by 8888@sv-exchange1.fbc.local:5065 <--- Transmitting (no NAT) to 10.0.0.25:5065 ---> SIP/2.0 202 Accepted Via: SIP/2.0/TCP 10.0.0.25:5065;branch=z9hG4bK315eaa8a;received=10.0.0.25 From: ;epid=DCDF2BE031;tag=21c28aa0b9 To: ;tag=as5ac0bd70 Call-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 CSeq: 1 REFER Server: Asterisk PBX 1.8.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.25:5065 Reliably Transmitting (no NAT) to 10.0.0.25:5065: NOTIFY sip:sv-exchange1.fbc.local:5065;transport=Tcp;maddr=10.0.0.25 SIP/2.0 Via: SIP/2.0/TCP 10.0.0.141:5060;branch=z9hG4bK56d6d4e0 Max-Forwards: 70 From: "Chris Baker" ;tag=as5ac0bd70 To: ;tag=21c28aa0b9 Contact: Call-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.8.1.1 Event: refer;id=1 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.25:5065 Reliably Transmitting (no NAT) to 10.0.0.25:5065: NOTIFY sip:sv-exchange1.fbc.local:5065;transport=Tcp;maddr=10.0.0.25 SIP/2.0 Via: SIP/2.0/TCP 10.0.0.141:5060;branch=z9hG4bK307b38ef Max-Forwards: 70 From: "Chris Baker" ;tag=as5ac0bd70 To: ;tag=21c28aa0b9 Contact: Call-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 1.8.1.1 Event: refer;id=1 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- Scheduling destruction of SIP dialog '1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060' in 32000 ms (Method: REFER) == Spawn extension (internal, 2711, 1) exited non-zero on 'SIP/2710-000001e6' Scheduling destruction of SIP dialog 'a8c743cc-b120cda7-57e26522@10.0.1.114' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.1.114:5060 Reliably Transmitting (no NAT) to 10.0.1.114:5060: BYE sip:2710@10.0.1.114 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.141:5060;branch=z9hG4bK2bb62072 Max-Forwards: 70 From: ;tag=as07426c5a To: "Chris Baker" ;tag=4BF9B4A0-8852BFFB Call-ID: a8c743cc-b120cda7-57e26522@10.0.1.114 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.1.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.0.1.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.141:5060;branch=z9hG4bK2bb62072 From: ;tag=as07426c5a To: "Chris Baker" ;tag=4BF9B4A0-8852BFFB CSeq: 102 BYE Call-ID: a8c743cc-b120cda7-57e26522@10.0.1.114 Contact: User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.1.0054 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'a8c743cc-b120cda7-57e26522@10.0.1.114' Method: ACK <--- SIP read from TCP:10.0.0.25:5065 ---> SIP/2.0 200 OK FROM: "Chris Baker";tag=as5ac0bd70 TO: ;tag=21c28aa0b9;epid=DCDF2BE031 CSEQ: 103 NOTIFY CALL-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 VIA: SIP/2.0/TCP 10.0.0.141:5060;branch=z9hG4bK56d6d4e0 CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TCP:10.0.0.25:5065 ---> SIP/2.0 200 OK FROM: "Chris Baker";tag=as5ac0bd70 TO: ;tag=21c28aa0b9;epid=DCDF2BE031 CSEQ: 104 NOTIFY CALL-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 VIA: SIP/2.0/TCP 10.0.0.141:5060;branch=z9hG4bK307b38ef CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TCP:10.0.0.25:5065 ---> BYE sip:2710@10.0.0.141:5060;transport=TCP SIP/2.0 FROM: ;epid=DCDF2BE031;tag=21c28aa0b9 TO: ;tag=as5ac0bd70 CSEQ: 2 BYE CALL-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.0.0.25:5065;branch=z9hG4bK46eba27c CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- Sending to 10.0.0.25:5065 (no NAT) Scheduling destruction of SIP dialog '1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.0.0.25:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.0.25:5065;branch=z9hG4bK46eba27c;received=10.0.0.25 From: ;epid=DCDF2BE031;tag=21c28aa0b9 To: ;tag=as5ac0bd70 Call-ID: 1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060 CSeq: 2 BYE Server: Asterisk PBX 1.8.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '1d7c2f7c08a94bba4861457736b99b95@10.0.0.141:5060' Method: BYE Reliably Transmitting (no NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.141:5060;branch=z9hG4bK22fb7ca9 Max-Forwards: 70 From: "asterisk" ;tag=as7baa68dc To: Contact: Call-ID: 48dbab447fe5c1060db9eda07bc3e130@10.0.0.141:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date: Thu, 06 Jan 2011 20:19:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> sip set debug off SIP Debugging Disabled asterisk*CLI>