[root@pbx asterisk]# asterisk -vvvvvvvvvvvvvvvvvvvvddddddddddddddddddrgc
Parsing /etc/asterisk/asterisk.conf
Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.10 currently running on pbx (pid = 26872)
Verbosity was 3 and is now 20
Core debug was 0 and is now 18
Asterisk Ready.
-- Remote UNIX connection
pbx*CLI>
pbx*CLI> sip debug peer 207.218.174.244
SIP Debugging Enabled for IP: 207.218.174.244:5060
pbx*CLI>
<-- SIP read from 207.218.174.244:5060:
INVITE sip:17135745600;npdi=yes;rn=17135745600@XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1
From: <sip:000@207.218.174.244:5060;GX=GXSUPERDIDGOLD-louuld03bbqn1>;tag=SDs46a201-gK0d5ba019
To: <sip:17135745600@XXX.XXX.XXX.XXX:5060>
Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1
CSeq: 31123 INVITE
Max-Forwards: 69
Allow: OPTIONS,INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay
Contact: <sip:000@207.218.174.244:5060;GX=GXSUPERDIDGOLD-louuld03bbqn1;transport=udp>
Content-Length: 186
Content-Disposition: session; handling=required
Content-Type: application/sdp
P-Asserted-Identity: <sip:000@67.17.63.239:4000>
v=0
o=Sonus_UAC 22906 26156 IN IP4 207.218.174.244
s=SIP Media Capabilities
c=IN IP4 207.218.174.244
t=0 0
m=audio 53174 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:20
--- (14 headers 9 lines)---
Using INVITE request as basis request - SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1
Sending to 207.218.174.244 : 5060 (NAT)
Found peer '207.218.174.244'
Found RTP audio format 0
Peer audio RTP is at port 207.218.174.244:53174
Found description format PCMU
Capabilities: us - 0x51f (g723|gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for s in global (domain 17135745600)
Reliably Transmitting (NAT) to 207.218.174.244:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1;received=207.218.174.244
From: <sip:000@207.218.174.244:5060;GX=GXSUPERDIDGOLD-louuld03bbqn1>;tag=SDs46a201-gK0d5ba019
To: <sip:17135745600@XXX.XXX.XXX.XXX:5060>;tag=as785b49c2
Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1
CSeq: 31123 INVITE
User-Agent: pbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:17135745600@XXX.XXX.XXX.XXX>
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 207.218.174.244:5060:
ACK sip:17135745600;npdi=yes;rn=17135745600@XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 207.218.174.244:5060;branch=z9hG4bK00252m100gm1skc70641.1
CSeq: 31123 ACK
From: <sip:000@207.218.174.244:5060;GX=GXSUPERDIDGOLD-louuld03bbqn1>;tag=SDs46a201-gK0d5ba019
To: <sip:17135745600@XXX.XXX.XXX.XXX:5060>;tag=as785b49c2
Call-ID: SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call 'SDs46a201-b433483c00496a6bbd048565fcfd1e8a-v3000i1'
pbx*CLI> exit
Executing last minute cleanups
Asterisk cleanly ending (0).
[root@pbx asterisk]#