aries*CLI> sip debug
SIP Debugging enabled
We're at 72.232.182.13 port 11864
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 72.40.10.150:5060:
INVITE sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK305ac009;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as71ad534b
To: <sip:200@72.40.10.150:5060>
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 7039e04d1cd91a3476ced2fb7337714c@72.232.182.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Jul 2006 23:27:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 3534 3534 IN IP4 72.232.182.13
s=session
c=IN IP4 72.232.182.13
t=0 0
m=audio 11864 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
aries*CLI>
<-- SIP read from 72.40.10.150:5060:
SIP/2.0 404 Not Found
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as71ad534b
Call-ID: 7039e04d1cd91a3476ced2fb7337714c@72.232.182.13
CSeq: 102 INVITE
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK305ac009
Server: Linksys/PAP2-3.1.3(LS)
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 72.40.10.150:5060:
ACK sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK305ac009;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as71ad534b
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 7039e04d1cd91a3476ced2fb7337714c@72.232.182.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '7039e04d1cd91a3476ced2fb7337714c@72.232.182.13'
aries*CLI>