<--- SIP read from UDP:10.0.0.128:5060 ---> REFER sip:201@10.0.0.5:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bKf6a493cb4f5c93d12 Max-Forwards: 70 From: "216" ;tag=cc566ea55c To: ;tag=as79c112b2 Call-ID: cd109d4de3acce4f CSeq: 4912 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="55i",realm="asterisk",nonce="26ac1f9a",uri="sip:201@10.0.0.5:5060",response="3016b872db86be5dfd7d81364f51dcf0",algorithm=MD5 Contact: "216" ;+sip.instance="" Refer-To: "Line 1" Referred-By: Supported: path User-Agent: Aastra 55i/3.2.2.2063 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Call cd109d4de3acce4f got a SIP call transfer from caller: (REFER)! SIP transfer to extension 0015651f831f@local by 55i@10.0.0.5 <--- Transmitting (no NAT) to 10.0.0.128:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bKf6a493cb4f5c93d12;received=10.0.0.128 From: "216" ;tag=cc566ea55c To: ;tag=as79c112b2 Call-ID: cd109d4de3acce4f CSeq: 4912 REFER Server: Asterisk PBX 1.8.11.2-9 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 10.0.0.128:5060 Reliably Transmitting (no NAT) to 10.0.0.128:5060: NOTIFY sip:55i@10.0.0.128:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1abad28c Max-Forwards: 70 From: ;tag=as79c112b2 To: "216" ;tag=cc566ea55c Contact: Call-ID: cd109d4de3acce4f CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.8.11.2-9 Event: refer;id=4912 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 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.128:5060 Audio is at 19144 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.128:5060: INVITE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK6383945f Max-Forwards: 70 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Contact: Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 106 INVITE User-Agent: Asterisk PBX 1.8.11.2-9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "Line 1" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 231 v=0 o=root 1084297811 1084297817 IN IP4 10.0.0.5 s=Asterisk PBX 1.8.11.2-9 c=IN IP4 10.0.0.5 t=0 0 m=audio 19144 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK) Scheduling destruction of SIP dialog 'cd109d4de3acce4f' in 32000 ms (Method: REFER) <--- SIP read from UDP:10.0.0.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1abad28c From: ;tag=as79c112b2 To: "216" ;tag=cc566ea55c Call-ID: cd109d4de3acce4f CSeq: 102 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 55i/3.2.2.2063 Supported: path Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.0.0.128:5060 ---> BYE sip:201@10.0.0.5:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK5cb87f9c522291850 Max-Forwards: 70 From: "216" ;tag=cc566ea55c To: ;tag=as79c112b2 Call-ID: cd109d4de3acce4f CSeq: 4913 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="55i",realm="asterisk",nonce="26ac1f9a",uri="sip:201@10.0.0.5:5060",response="677cc5608b76a4dafa5ef87f04942d3d",algorithm=MD5 Supported: path User-Agent: Aastra 55i/3.2.2.2063 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.0.0.128:5060 (no NAT) Scheduling destruction of SIP dialog 'cd109d4de3acce4f' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.0.0.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK5cb87f9c522291850;received=10.0.0.128 From: "216" ;tag=cc566ea55c To: ;tag=as79c112b2 Call-ID: cd109d4de3acce4f CSeq: 4913 BYE Server: Asterisk PBX 1.8.11.2-9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:10.0.0.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK6383945f From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 106 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "216" ;+sip.instance="" Server: Aastra 55i/3.2.2.2063 Supported: path, replaces Content-Type: application/sdp Content-Length: 229 v=0 o=MxSIP 0 7 IN IP4 10.0.0.128 s=SIP Call c=IN IP4 10.0.0.128 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 12 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.128:5060 Reliably Transmitting (no NAT) to 10.0.0.128:5060: BYE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5e7de6c7 Max-Forwards: 70 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 107 BYE User-Agent: Asterisk PBX 1.8.11.2-9 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 ontent-Length: 0 --- Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK) Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.0.128:3000 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.128:5060 Transmitting (no NAT) to 10.0.0.128:5060: ACK sip:55i@10.0.0.128:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1e809647 Max-Forwards: 70 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Contact: Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.11.2-9 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.128:5060 Reliably Transmitting (no NAT) to 10.0.0.128:5060: BYE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK0ce74ee4 Max-Forwards: 70 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 108 BYE User-Agent: Asterisk PBX 1.8.11.2-9 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 ontent-Length: 0 --- Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK) <--- SIP read from UDP:10.0.0.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5e7de6c7 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 107 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 55i/3.2.2.2063 Supported: path Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.0.0.128:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK0ce74ee4 From: "Line 1" ;tag=as1ed3dea1 To: ;tag=1002760794 Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk CSeq: 108 BYE Server: Aastra 55i/3.2.2.2063 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' Method: ACK [Jul 3 17:45:38] NOTICE[29358]: chan_sip.c:13368 sip_reg_timeout: -- Registration for 'slicksip@10.0.0.250' timed out, trying again (Attempt #3)