<-- SIP read from 4.68.251.13:50953: 
INVITE sip:+18666772071@64.156.11.171:5060 SIP/2.0
Via: SIP/2.0/UDP 4.68.251.13:5060;branch=z9hG4bK506071629581-1130227721058
From: <sip:+15107503817@4.68.251.13>;tag=VPSF506071629581
To: <sip:+18666772071@64.156.11.171:5060>
Call-ID: SEAMGC6920060222012633001139@209.244.63.178
CSeq: 1 INVITE
Contact: <sip:+15107503817@4.68.251.13:5060;transport=udp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 115
Remote-Party-ID:  <sip:+15107503817@209.244.63.178>;party=calling;screen=yes;privacy=off

v=0
o=- 1140571593 1140571594 IN IP4 63.215.27.91
s=-
c=IN IP4 63.215.27.91
t=0 0
m=audio 60218 RTP/AVP 0 18

--- (11 headers 6 lines)---
Using INVITE request as basis request - SEAMGC6920060222012633001139@209.244.63.178
Sending to 4.68.251.13 : 5060 (non-NAT)
Found peer 'level3a'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 63.215.27.91:60218
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 +18666772071 in level3inbound (domain 64.156.11.171)
list_route: hop: <sip:+15107503817@4.68.251.13:5060;transport=udp>
Transmitting (no NAT) to 4.68.251.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 4.68.251.13:5060;branch=z9hG4bK506071629581-1130227721058;received=4.68.251.13
From: <sip:+15107503817@4.68.251.13>;tag=VPSF506071629581
To: <sip:+18666772071@64.156.11.171:5060>
Call-ID: SEAMGC6920060222012633001139@209.244.63.178
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:+18666772071@64.156.11.171>
Content-Length: 0


---
    -- Executing Answer("SIP/4.68.251.13-b7909058", "") in new stack
We're at 64.156.11.171 port 15334
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 4.68.251.13:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.68.251.13:5060;branch=z9hG4bK506071629581-1130227721058;received=4.68.251.13
From: <sip:+15107503817@4.68.251.13>;tag=VPSF506071629581
To: <sip:+18666772071@64.156.11.171:5060>;tag=as5efaa4e7
Call-ID: SEAMGC6920060222012633001139@209.244.63.178
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:+18666772071@64.156.11.171>
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 24449 24449 IN IP4 64.156.11.171
s=session
c=IN IP4 64.156.11.171
t=0 0
m=audio 15334 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---


<-- SIP read from 4.68.251.13:50953: 
ACK sip:+18666772071@64.156.11.171 SIP/2.0
Via: SIP/2.0/UDP 4.68.251.13:5060;branch=z9hG4bK506071629581-1130227721060
From: <sip:+15107503817@4.68.251.13>;tag=VPSF506071629581
To: <sip:+18666772071@64.156.11.171:5060>;tag=as5efaa4e7
Call-ID: SEAMGC6920060222012633001139@209.244.63.178
CSeq: 1 ACK
Contact: <sip:4.68.251.13:5060;transport=udp>
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
ev02d*CLI> 
[root@ev02d asterisk]#