rendered paste body--- (11 headers 6 lines)---
Using INVITE request as basis request - 02ba94b86f90fc665f9cd2af1db116ae-44be4581@204.124.238.211
Sending to 207.138.151.33 : 5060 (non-NAT)
Found peer 'NCITE'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 207.138.151.46:10672
Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for s in testing (domain +12037743993)
Reliably Transmitting (no NAT) to 207.138.151.33:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bKa09c9b4842748ef2b03861611290b1f7-0;received=207.138.151.33
From: <sip:12036790137@204.124.238.211:5060>;tag=674a05f923fa80c58011898d98b948a8
To: <sip:+12037743993;rn=+12035174198;npdi=yes@204.124.238.211:5060>;tag=as69f062e5
Call-ID: 02ba94b86f90fc665f9cd2af1db116ae-44be4581@204.124.238.211
CSeq: 64868 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:+12037743993@204.124.238.211>
Content-Length: 0