--- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.10.1.23: 5061; transport UDP (non-NAT) Transmitting (no NAT) to 10.10.1.23:5061:UDP with fd 16 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.23:5061;branch=z9hG4bK7e5ec2b9;received=10.10.1.23 From: sip:112@10.10.1.16 To: sip:112@10.10.1.16 Call-ID: 000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23 CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- UDP write: fd 16 -- Contact header: transport -- Contact header: q asterisk1*CLI> Transmitting (no NAT) to 10.10.1.23:5061:UDP with fd 16 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.23:5061;branch=z9hG4bK7e5ec2b9;received=10.10.1.23 From: sip:112@10.10.1.16 To: sip:112@10.10.1.16;tag=as76222803 Call-ID: 000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23 CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Sun, 01 Oct 2006 20:10:35 GMT Content-Length: 0 --- asterisk1*CLI> UDP write: fd 16 Scheduling destruction of call '000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23' in 15000 ms asterisk1*CLI> <-- Sip read from 207.174.202.4:5060:UDP INVITE sip:s@10.10.1.16;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;rport From: "Cell Phone CA" ;tag=as132f5b62 To: Contact: Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4 CSeq: 102 INVITE User-Agent: "TELIAX GEAR v2.1.0x" Max-Forwards: 70 Date: Sun, 01 Oct 2006 20:14:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 315 v=0 o=root 724 724 IN IP4 207.174.202.4 s=session c=IN IP4 207.174.202.4 t=0 0 m=audio 10088 RTP/AVP 0 18 111 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 14 lines)--- Using INVITE request as basis request - 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4 Sending to 207.174.202.4: 5060; transport UDP (NAT) Found peer 'teliax' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 111 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 207.174.202.4:10088 Found description format PCMU Found description format G729 Found description format G726-32 Found description format GSM Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in default (domain 10.10.1.16;transport=UDP) list_route: hop: Transmitting (NAT) to 207.174.202.4:5060:UDP with fd 16 SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;received=207.174.202.4;rport=5060 From: "Cell Phone CA" ;tag=as132f5b62 To: Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- UDP write: fd 16 -- Executing Playback("SIP/mybiz-08d48280", "vm-goodbye") in new stack We're at 10.10.1.16 port 17248 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 207.174.202.4:5060:UDP with fd 16 SIP/2.0 200 OK Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;received=207.174.202.4;rport=5060 From: "Cell Phone CA" ;tag=as132f5b62 To: ;tag=as6d0a49e8 Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 5675 5675 IN IP4 10.10.1.16 s=session c=IN IP4 10.10.1.16 t=0 0 m=audio 17248 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- UDP write: fd 16 -- Playing 'vm-goodbye' (language 'en') asterisk1*CLI> <-- Sip read from 207.174.202.4:5060:UDP ACK sip:s@10.10.1.16;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK11f016d9;rport From: "Cell Phone CA" ;tag=as132f5b62 To: ;tag=as6d0a49e8 Contact: Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4 CSeq: 102 ACK User-Agent: "TELIAX GEAR v2.1.0x" Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- asterisk1*CLI> -- Executing Macro("SIP/mybiz-08d48280", "hangupcall") in new stack -- Executing ResetCDR("SIP/mybiz-08d48280", "w") in new stack asterisk1*CLI> -- Executing NoCDR("SIP/mybiz-08d48280", "") in new stack -- Executing Wait("SIP/mybiz-08d48280", "5") in new stack asterisk1*CLI> <-- Sip read from 10.10.1.3:37810:UDP --- (0 headers 1 lines)--- asterisk1*CLI> 12 headers, 3 lines Reliably Transmitting (no NAT) to 10.10.1.23:5061:UDP with fd 16 NOTIFY sip:112@10.10.1.23:5061 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.16:5060;branch=z9hG4bK5826f50b From: "mybiz" ;tag=as230f95ca To: Contact: Call-ID: 7265553c6e240b0f5b6378b2482ad6d4@10.10.1.16 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@10.10.1.16 Voice-Message: 0/0 (0/0)