polifast2*CLI> <-- SIP read from 201.20.205.12:5060: INVITE sip:824785501@200.170.105.40 SIP/2.0 Via: SIP/2.0/UDP 201.20.205.12;rport;branch=z9hG4bK41SQScyZF8aUQ Max-Forwards: 69 From: "Ekiga Teste" ;tag=rFQgX2D3HpBcm To: Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16 CSeq: 85153128 INVITE Contact: User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 279 Remote-Party-ID: "Ekiga Teste" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 4735461259273311988 7209532826057431998 IN IP4 200.170.105.40 s=FreeSWITCH c=IN IP4 201.20.205.12 t=0 0 a=sendrecv m=audio 16388 RTP/AVP 8 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --- (15 headers 12 lines) --- Using INVITE request as basis request - 3b1949e5-9b9f-122a-e2b2-000475e7ee16 Sending to 201.20.205.12 : 5060 (NAT) Found peer 'fsua' Found RTP audio format 8 Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 13 Peer audio RTP is at port 201.20.205.12:16388 Found description format PCMA Found description format telephone-event Found description format CN Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Looking for 824785501 in from-sip-extern (domain 200.170.105.40) Reliably Transmitting (no NAT) to 201.20.205.12:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 201.20.205.12;branch=z9hG4bK41SQScyZF8aUQ;received=201.20.205.12;rport=5060 From: "Ekiga Teste" ;tag=rFQgX2D3HpBcm To: ;tag=as76d4b9db Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16 CSeq: 85153128 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- polifast2*CLI> <-- SIP read from 201.20.205.12:5060: ACK sip:824785501@200.170.105.40 SIP/2.0 Via: SIP/2.0/UDP 201.20.205.12;rport;branch=z9hG4bK41SQScyZF8aUQ Max-Forwards: 70 From: "Ekiga Teste" ;tag=rFQgX2D3HpBcm To: ;tag=as76d4b9db Call-ID: 3b1949e5-9b9f-122a-e2b2-000475e7ee16 CSeq: 85153128 ACK Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3b1949e5-9b9f-122a-e2b2-000475e7ee16' 12 headers, 0 lines Reliably Transmitting (no NAT) to 201.20.205.12:5060: OPTIONS sip:fsua@201.20.205.12:5060 SIP/2.0 Via: SIP/2.0/UDP 200.170.105.40:5060;branch=z9hG4bK4070ff24;rport From: "Unknown" ;tag=as49e5f6b8 To: Contact: Call-ID: 355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Jun 2007 20:08:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- polifast2*CLI> <-- SIP read from 201.20.205.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.170.105.40:5060;branch=z9hG4bK4070ff24;rport=5060 From: "Unknown" ;tag=as49e5f6b8 To: ;tag=Srg9yXy6eZ1yF Call-ID: 355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40 CSeq: 102 OPTIONS Contact: User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '355daaea70cd1c1e0ccb8a05139a0b39@200.170.105.40' polifast2*CLI> sip no debug SIP Debugging Disabled polifast2*CLI>