<------------->
--- (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 <sip:benjamoon@192.168.1.3:5061>;tag=310675945
To: <sip:06735877769@192.168.1.3>;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 <sip:benjamoon@192.168.1.3:5061>;tag=310675945
To: <sip:06735877769@192.168.1.3>;tag=as733ae40f
Contact: <sip:benjamoon@192.168.1.3:5061>
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" <sip:asterisk@192.168.1.3>;tag=as0ac5214c
To: <sip:benjamoon@192.168.1.3:5061>
Contact: <sip:asterisk@192.168.1.3>
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" <sip:asterisk@192.168.1.3>;tag=as0ac5214c
To: <sip:benjamoon@192.168.1.3:5061>;tag=13275586
Contact: <sip:benjamoon@192.168.1.3:5061>
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