sip debug peer 600 SIP Debugging Enabled for IP: 10.1.1.114:5060 The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> INVITE sip:7167069363@10.1.1.202:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB From: "Chris_Martin" ;tag=6F8F873F-735336 To: CSeq: 1 INVITE Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 257 v=0 o=- 1175878434 1175878434 IN IP4 10.1.1.114 s=Polycom IP Phone c=IN IP4 10.1.1.114 t=0 0 m=audio 2244 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Sending to 10.1.1.114 : 5060 (no NAT) Using INVITE request as basis request - 6cae04b2-13d7de4d-e790056c@10.1.1.114 <--- Reliably Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB;received=10.1.1.114 From: "Chris_Martin" ;tag=6F8F873F-735336 To: ;tag=as41904945 Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32e9ec6c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' in 32000 ms (Method: INVITE) Found user '600' TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> ACK sip:7167069363@10.1.1.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK28daca88886B56AB From: "Chris_Martin" ;tag=6F8F873F-735336 To: ;tag=as41904945 CSeq: 1 ACK Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.1.1.114:5060 ---> INVITE sip:7167069363@10.1.1.202:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890 From: "Chris_Martin" ;tag=6F8F873F-735336 To: CSeq: 2 INVITE Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="600", realm="asterisk", nonce="32e9ec6c", uri="sip:7167069363@10.1.1.202:5060;user=phone", response="08e1192f46bd636a36abf9e7ffed1c91", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 257 v=0 o=- 1175878434 1175878434 IN IP4 10.1.1.114 s=Polycom IP Phone c=IN IP4 10.1.1.114 t=0 0 m=audio 2244 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 10.1.1.114 : 5060 (no NAT) Using INVITE request as basis request - 6cae04b2-13d7de4d-e790056c@10.1.1.114 Found user '600' Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.1.1.114:2244 Found description format G722 for ID 9 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=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 10.1.1.114:2244 Looking for 7167069363 in default (domain 10.1.1.202) <--- Reliably Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890;received=10.1.1.114 From: "Chris_Martin" ;tag=6F8F873F-735336 To: ;tag=as41904945 Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' in 32000 ms (Method: INVITE) TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> ACK sip:7167069363@10.1.1.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK853d060151BB7890 From: "Chris_Martin" ;tag=6F8F873F-735336 To: ;tag=as41904945 CSeq: 2 ACK Call-ID: 6cae04b2-13d7de4d-e790056c@10.1.1.114 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Proxy-Authorization: Digest username="600", realm="asterisk", nonce="32e9ec6c", uri="sip:7167069363@10.1.1.202:5060;user=phone", response="08e1192f46bd636a36abf9e7ffed1c91", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> REGISTER sip:10.1.1.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129 From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: CSeq: 23 REGISTER Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Authorization: Digest username="600", realm="asterisk", nonce="1a751fb6", uri="sip:10.1.1.202:5060", response="d7a23b2cef79adc7e1a7bc0666bdf8e1", algorithm=MD5 Max-Forwards: 70 Expires: 30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.1.1.114 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129;received=10.1.1.114 From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 CSeq: 23 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> TWAsterisk*CLI> <--- Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bKf1ffbafeA5B83129;received=10.1.1.114 From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: ;tag=as01a16496 Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 CSeq: 23 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 06 Apr 2007 17:08:07 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '19723fe6-fc84eef1-9931e0c0@10.1.1.114' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '1652f5625ef14199368414bb37a0a960@10.1.1.202' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.1.1.114:5060: NOTIFY sip:600@10.1.1.114 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK3c19d2cf;rport From: "asterisk" ;tag=as4799aba1 To: Contact: Call-ID: 1652f5625ef14199368414bb37a0a960@10.1.1.202 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:asterisk@10.1.1.202 Voice-Message: 6/0 (0/0) --- TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK3c19d2cf;rport From: "asterisk" ;tag=as4799aba1 To: ;tag=E25D798-87EFEC7B CSeq: 102 NOTIFY Call-ID: 1652f5625ef14199368414bb37a0a960@10.1.1.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1652f5625ef14199368414bb37a0a960@10.1.1.202' Method: NOTIFY Really destroying SIP dialog '211e5122-efa4f3fd-d4d0c35c@10.1.1.114' Method: ACK Really destroying SIP dialog '6cae04b2-13d7de4d-e790056c@10.1.1.114' Method: ACK TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> REGISTER sip:10.1.1.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: CSeq: 24 REGISTER Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Authorization: Digest username="600", realm="asterisk", nonce="1a751fb6", uri="sip:10.1.1.202:5060", response="d7a23b2cef79adc7e1a7bc0666bdf8e1", algorithm=MD5 Max-Forwards: 70 Expires: 30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.1.1.114 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D;received=10.1.1.114 From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 CSeq: 24 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> TWAsterisk*CLI> <--- Transmitting (no NAT) to 10.1.1.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.114;branch=z9hG4bK61e29c42891D5C9D;received=10.1.1.114 From: "Chris_Martin" ;tag=59F81183-5EC2D3AA To: ;tag=as01a16496 Call-ID: 19723fe6-fc84eef1-9931e0c0@10.1.1.114 CSeq: 24 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 06 Apr 2007 17:08:37 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '19723fe6-fc84eef1-9931e0c0@10.1.1.114' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '32d41e722bb90fb06914018a56ab468b@10.1.1.202' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.1.1.114:5060: NOTIFY sip:600@10.1.1.114 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK7d3cb859;rport From: "asterisk" ;tag=as143bf4c7 To: Contact: Call-ID: 32d41e722bb90fb06914018a56ab468b@10.1.1.202 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:asterisk@10.1.1.202 Voice-Message: 6/0 (0/0) --- TWAsterisk*CLI> <--- SIP read from 10.1.1.114:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.202:5060;branch=z9hG4bK7d3cb859;rport From: "asterisk" ;tag=as143bf4c7 To: ;tag=9497CB7C-1E36420F CSeq: 102 NOTIFY Call-ID: 32d41e722bb90fb06914018a56ab468b@10.1.1.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.2.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '32d41e722bb90fb06914018a56ab468b@10.1.1.202' Method: NOTIFY TWAsterisk*CLI>