Reliably Transmitting (no NAT) to 192.168.101.19:2048:
OPTIONS sip:snom@192.168.101.19:2048 SIP/2.0
v: SIP/2.0/UDP 192.168.101.250:5060;branch=z9hG4bK7d5d9e26;rport
Max-Forwards: 70
f: "asterisk" <sip:asterisk@192.168.101.250>;tag=as41f1a133
t: <sip:snom@192.168.101.19:2048>
m: <sip:asterisk@192.168.101.250>
i: 4ca4e07e7f3a40ab0e4240be7524c488@192.168.101.250
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.0-rc6
Date: Tue, 24 Nov 2009 00:22:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces, timer
l: 0


---
phone*CLI>
<--- SIP read from UDP:192.168.101.19:2048 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.250:5060;branch=z9hG4bK7d5d9e26;rport=5060
From: "asterisk" <sip:asterisk@192.168.101.250>;tag=as41f1a133
To: <sip:snom@192.168.101.19:2048>
Call-ID: 4ca4e07e7f3a40ab0e4240be7524c488@192.168.101.250
CSeq: 102 OPTIONS
Contact: <sip:16@192.168.101.19:2048>;reg-id=1
User-Agent: snom360/7.3.27
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '4ca4e07e7f3a40ab0e4240be7524c488@192.168.101.250' Method: OPTIONS