Mine

Connected to Asterisk 1.2.9.1 currently running on aries (pid = 29933)
aries*CLI> sip debug
SIP Debugging enabled
aries*CLI>
<-- SIP read from 66.250.69.13:5060:
INVITE sip:8006170141@host41.com SIP/2.0
Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05
From: "4072930335" <sip:4072930335@66.250.69.13>;tag=as691eedde
To: <sip:8006170141@host41.com>
Contact: <sip:4072930335@66.250.69.13>
Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13
CSeq: 102 INVITE
User-Agent: asterlink/1.2
Date: Sun, 09 Jul 2006 01:48:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 10036 10036 IN IP4 66.250.69.13
s=session
c=IN IP4 66.250.69.13
t=0 0
m=audio 19082 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 20 lines)---
Using INVITE request as basis request - 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13
Sending to 66.250.69.13 : 5060 (non-NAT)
Found peer 'asterlink-switch-03'
Found RTP audio format 10
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 66.250.69.13:19082
Found description format L16
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8006170141 in incoming (domain host41.com)
list_route: hop: <sip:4072930335@66.250.69.13>
Transmitting (no NAT) to 66.250.69.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05;received=66.250.69.13
From: "4072930335" <sip:4072930335@66.250.69.13>;tag=as691eedde
To: <sip:8006170141@host41.com>
Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Length: 0


---
Transmitting (no NAT) to 66.250.69.13:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05;received=66.250.69.13
From: "4072930335" <sip:4072930335@66.250.69.13>;tag=as691eedde
To: <sip:8006170141@host41.com>;tag=as2a234c1b
Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Length: 0


---
Jul  8 21:50:03 WARNING[29936]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81577a8', 10 retries!
aries*CLI>
<-- SIP read from 66.250.69.13:5060:
ACK sip:8006170141@host41.com SIP/2.0
Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05
From: "4072930335" <sip:4072930335@66.250.69.13>;tag=as691eedde
To: <sip:8006170141@host41.com>;tag=as2a234c1b
Contact: <sip:4072930335@66.250.69.13>
Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13
CSeq: 102 ACK
User-Agent: asterlink/1.2
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '17c29dd64e0b98cb5faf961243c1b481@66.250.69.13'