Mine

asterisk1*CLI> sip debug ip sip.vtwhite.com
SIP Debugging Enabled for IP: 216.120.251.172
12 headers, 0 lines
Reliably Transmitting (NAT) to 216.120.251.172:5060:
OPTIONS sip:sip.vtwhite.com SIP/2.0
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK29a596e3;rport
From: "Unknown" <sip:Unknown@76.4.108.63>;tag=as76819a66
To: <sip:sip.vtwhite.com>
Contact: <sip:Unknown@76.4.108.63>
Call-ID: 57a5904a515b50d850ce38c2167d30f9@76.4.108.63
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 May 2007 12:09:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI> 
<-- SIP read from 216.120.251.172:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.4.108.63:5060;branch=z9hG4bK29a596e3;rport=10110
From: "Unknown" <sip:Unknown@76.4.108.63>;tag=as76819a66
To: <sip:sip.vtwhite.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.6b34
Call-ID: 57a5904a515b50d850ce38c2167d30f9@76.4.108.63
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding: 
Accept-Language: en
Support: 
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 216.120.251.172:5060 "Noisy feedback tells:  pid=8518 req_src_ip=76.4.108.63 req_src_port=10110 in_uri=sip:sip.vtwhite.com out_uri=sip:sip.vtwhite.com via_cnt==1"


--- (13 headers 0 lines) ---
Destroying call '57a5904a515b50d850ce38c2167d30f9@76.4.108.63'