SIP Debugging Enabled for IP: 72.55.143.XXX:5060 -- Executing [005642325405@test:1] Dial("SIP/2563105-0819cf80", "sip/5642325405@nyphone|45") in new stack Audio is at 164.77.171.XXX port 16548 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 72.55.143.XXX:5060: INVITE sip:5642325405@72.55.143.XXX SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport From: "2563105" ;tag=as726ac50a To: Contact: Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 20 Jul 2007 03:38:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 2475 2475 IN IP4 164.77.171.XXX s=session c=IN IP4 164.77.171.XXX t=0 0 m=audio 16548 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 5642325405@nyphone vaca*CLI> <--- SIP read from 72.55.143.XXX:5060 ---> SIP/2.0 407 Proxy Authentication Required CSeq: 102 INVITE Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport From: "2563105" ;tag=as726ac50a Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX To: Contact: Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="118490324119231120007472128429" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 72.55.143.XXX:5060: ACK sip:5642325405@72.55.143.XXX SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport From: "2563105" ;tag=as726ac50a To: Contact: Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 164.77.171.XXX port 16548 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 72.55.143.XXX:5060: INVITE sip:5642325405@72.55.143.XXX SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport From: "2563105" ;tag=as726ac50a To: Contact: Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="test770", realm="VoipSwitch", algorithm=MD5, uri="sip:5642325405@72.55.143.XXX", nonce="118490324119231120007472128429", response="413be923621811a639c3b0e83d3a2e74", opaque="" Date: Fri, 20 Jul 2007 03:38:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 2475 2476 IN IP4 164.77.171.XXX s=session c=IN IP4 164.77.171.XXX t=0 0 m=audio 16548 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- vaca*CLI> <--- SIP read from 72.55.143.XXX:5060 ---> SIP/2.0 200 OK CSeq: 103 INVITE Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport From: "2563105" ;tag=as726ac50a Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX To: ;tag=1907470723212675853288937 Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX s=VoipSIP i=Audio Session c=IN IP4 72.55.143.XXX t=0 0 m=audio 8936 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (9 headers 10 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 72.55.143.XXX:8936 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.55.143.XXX:8936 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 72.55.143.XXX, port 5060 Transmitting (no NAT) to 72.55.143.XXX:5060: ACK sip:72.55.143.XXX:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK22458f4a;rport From: "2563105" ;tag=as726ac50a To: ;tag=1907470723212675853288937 Contact: Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/nyphone-081a7768 left from hold -- SIP/nyphone-081a7768 answered SIP/2563105-0819cf80 -- Packet2Packet bridging SIP/2563105-0819cf80 and SIP/nyphone-081a7768 vaca*CLI> <--- SIP read from 72.55.143.XXX:5060 ---> SIP/2.0 183 Session Progress CSeq: 103 INVITE Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport From: "2563105" ;tag=as726ac50a Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX To: ;tag=1907470723212675853288937 Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX s=VoipSIP =Audio Session c=IN IP4 72.55.143.XXX t=0 0 m=audio 8936 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (9 headers 10 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 72.55.143.XXX:8936 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.55.143.XXX:8936 -- Packet2Packet bridging SIP/2563105-0819cf80 and SIP/nyphone-081a7768 Scheduling destruction of SIP dialog '5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 72.55.143.XXX, port 5060 Reliably Transmitting (no NAT) to 72.55.143.XXX:5060: BYE sip:72.55.143.XXX:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK5fedb647;rport From: "2563105" ;tag=as726ac50a To: ;tag=1907470723212675853288937 Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="test770", realm="VoipSwitch", algorithm=MD5, uri="sip:72.55.143.XXX:5060", nonce="118490324119231120007472128429", response="84f5246532adf64b1b47849b6fa03648", opaque="" Content-Length: 0 --- == Spawn extension (test, 005642325405, 1) exited non-zero on 'SIP/2563105-0819cf80' vaca*CLI> <--- SIP read from 72.55.143.XXX:5060 ---> SIP/2.0 200 OK CSeq: 104 BYE Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK5fedb647;rport From: "2563105" ;tag=as726ac50a Call-ID: 5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX To: ;tag=1907470723212675853288937 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '5b7470690d6cd476346bc1113f609a4b@164.77.171.XXX' Method: INVITE vaca*CLI>