CentPBX*CLI>
<--- SIP read from 10.1.75.102:56654 --->



<------------->
--- (0 headers 1 lines) ---
CentPBX*CLI>
<--- SIP read from 66.23.129.253:5060 --->
INVITE sip:9166751450@10.1.2.162 SIP/2.0
Record-Route: <sip:19166751450@66.23.129.253:5060;nat=yes;ftag=d417e4e051ff5fbf45d15c29effd61d3;lr=on>
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9b3.2f02bea3.0
Via: SIP/2.0/UDP 69.30.45.229:5061;branch=z9hG4bKd2b341c3e47bb7049ec5036d61bc5280;rport=5061
Max-Forwards: 16
From: <sip:9164584780@69.30.45.229>;tag=d417e4e051ff5fbf45d15c29effd61d3
To: <sip:19166751450@66.23.129.253>
Call-ID: SJOMGC0120080205020123006231@209.244.63.52
CSeq: 200 INVITE
Contact: Anonymous <sip:69.30.45.229:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 962295537-2430306225-1784524049-2448181782
h323-conf-id: 962295537-2430306225-1784524049-2448181782
Content-Length: 167
Content-Type: application/sdp

v=0
o=Sippy 178465676 0 IN IP4 69.30.45.229
s=-
t=0 0
m=audio 61084 RTP/AVP 0 18 101
c=IN IP4 209.247.21.225
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (16 headers 8 lines) ---
Sending to 66.23.129.253 : 5060 (no NAT)
Using INVITE request as basis request - SJOMGC0120080205020123006231@209.244.63.52
Found peer 'nexvortex.com'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 209.247.21.225:61084
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.247.21.225:61084
Looking for 9166751450 in from-pstn (domain 10.1.2.162)

<--- Reliably Transmitting (no NAT) to 66.23.129.253:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9b3.2f02bea3.0;received=66.23.129.253
Via: SIP/2.0/UDP 69.30.45.229:5061;branch=z9hG4bKd2b341c3e47bb7049ec5036d61bc5280;rport=5061
From: <sip:9164584780@69.30.45.229>;tag=d417e4e051ff5fbf45d15c29effd61d3
To: <sip:19166751450@66.23.129.253>;tag=as604fa185
Call-ID: SJOMGC0120080205020123006231@209.244.63.52
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'SJOMGC0120080205020123006231@209.244.63.52' in 32000 ms (Method: INVITE)
CentPBX*CLI>
<--- SIP read from 66.23.129.253:5060 --->
ACK sip:9166751450@10.1.2.162 SIP/2.0
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9b3.2f02bea3.0
From: <sip:9164584780@69.30.45.229>;tag=d417e4e051ff5fbf45d15c29effd61d3
Call-ID: SJOMGC0120080205020123006231@209.244.63.52
To: <sip:19166751450@66.23.129.253>;tag=as604fa185
CSeq: 200 ACK
User-Agent: Sip EXpress router(0.9.6 (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'SJOMGC0120080205020123006231@209.244.63.52' Method: ACK
Really destroying SIP dialog '06592bd31c7aa1c02772ea1501945cd2@127.0.0.1' Method: REGISTER
CentPBX*CLI>