<-------------> --- (12 headers 14 lines) --- Sending to 192.168.1.3 : 5061 (NAT) Using INVITE request as basis request - 25455785-DAD8-A9F9-0743-27C411964846@192.168.1.3 Found user 'benjamoon' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format gsm for ID 3 Found description format iLBC for ID 98 Found description format speex for ID 97 Found description format telephone-event for ID 101 Capabilities: us - 0x20c (ulaw|alaw|speex), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x20c (ulaw|alaw|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.3:8000 Looking for 06735877769 in maison (domain 192.168.1.3) <--- Reliably Transmitting (NAT) to 192.168.1.3:5061 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5061;branch=z9hG4bK68DB27FF3949611223A2BD9EC70E9C4C;received=192.168.1.3;rport=5061 From: Benjamin Frank ;tag=310675945 To: ;tag=as733ae40f Call-ID: 25455785-DAD8-A9F9-0743-27C411964846@192.168.1.3 CSeq: 29519 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '25455785-DAD8-A9F9-0743-27C411964846@192.168.1.3' in 32000 ms (Method: INVITE) kalida*CLI> <--- SIP read from 192.168.1.3:5061 ---> ACK sip:06735877769@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5061;rport;branch=z9hG4bK68DB27FF3949611223A2BD9EC70E9C4C From: Benjamin Frank ;tag=310675945 To: ;tag=as733ae40f Contact: Call-ID: 25455785-DAD8-A9F9-0743-27C411964846@192.168.1.3 CSeq: 29519 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.1.3:5061: OPTIONS sip:benjamoon@192.168.1.3:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK03feeb04;rport From: "asterisk" ;tag=as0ac5214c To: Contact: Call-ID: 584395d541ab4149150e6ee3663b2b98@192.168.1.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 27 Dec 2007 13:58:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- kalida*CLI> <--- SIP read from 192.168.1.3:5061 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK03feeb04;rport From: "asterisk" ;tag=as0ac5214c To: ;tag=13275586 Contact: Call-ID: 584395d541ab4149150e6ee3663b2b98@192.168.1.3 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1105d Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '584395d541ab4149150e6ee3663b2b98@192.168.1.3' Method: OPTIONS