SIP Debugging Enabled for IP: 4.79.212.236
pbx*CLI>
<-- SIP read from 4.79.212.236:5060:
INVITE sip:+18638774181@66.192.107.225:5060 SIP/2.0
Record-Route: <sip:4.79.212.236;lr;ftag=VPSF506071629460>
Record-Route: <sip:4.79.212.228;lr;ftag=VPSF506071629460>
Record-Route: <sip:4.79.212.236;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.228;branch=z9hG4bK7b1a.137cb913.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1149642003164
From: "Cell Phone   FL" <sip:+13522278124@4.68.250.148>;tag=VPSF506071629460
To: <sip:+18638774181@4.79.212.236:5060>
Call-ID: TMPMGC0120060804210907049522@209.244.63.33
CSeq: 1 INVITE
Contact: <sip:+13522278124@4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Type: application/sdp
Content-Length: 119
Remote-Party-ID: "Cell Phone   FL"  <sip:+13522278124@4.68.250.148>;party=calling;screen=yes;privacy=off

v=0
o=- 1154725747 1154725748 IN IP4 209.247.22.161
s=-
c=IN IP4 209.247.22.161
t=0 0
m=audio 60622 RTP/AVP 0 18

--- (17 headers 6 lines)---
Using INVITE request as basis request - TMPMGC0120060804210907049522@209.244.63.33
Sending to 4.79.212.236 : 5060 (non-NAT)
Found peer 'broadband'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 209.247.22.161:60622
Peer video RTP is at port 209.247.22.161:65535
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 +18638774181 in local (domain 66.192.107.225)
Reliably Transmitting (no NAT) to 4.79.212.236:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 4.79.212.236;branch=0;received=4.79.212.236
Via: SIP/2.0/UDP 4.79.212.228;branch=z9hG4bK7b1a.137cb913.0
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1149642003164
From: "Cell Phone   FL" <sip:+13522278124@4.68.250.148>;tag=VPSF506071629460
To: <sip:+18638774181@4.79.212.236:5060>;tag=as243f7717
Call-ID: TMPMGC0120060804210907049522@209.244.63.33
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:+18638774181@66.192.107.225>
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 4.79.212.236:5060:
ACK sip:+18638774181@66.192.107.225:5060 SIP/2.0
Max-Forwards: 16
Record-Route: <sip:4.79.212.236;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 4.79.212.236;branch=0
Via: SIP/2.0/UDP 4.79.212.228;branch=z9hG4bK7b1a.137cb913.0
From: "Cell Phone   FL" <sip:+13522278124@4.68.250.148>;tag=VPSF506071629460
Call-ID: TMPMGC0120060804210907049522@209.244.63.33
To: <sip:+18638774181@4.79.212.236:5060>;tag=as243f7717
CSeq: 1 ACK
User-Agent: Bandwidth.com SDE (v1.0)
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call 'TMPMGC0120060804210907049522@209.244.63.33'