<------------->
--- (14 headers 19 lines) ---
Sending to 67.228.46.178 : 5060 (NAT)
Using INVITE request as basis request - 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
Found no matching peer or user for '67.228.46.178:5060'
Found RTP audio format 111
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 67.228.46.178:3474
Found audio description format G726-32 for ID 111
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc0f (g723|gsm|ulaw|alaw|g726|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.228.46.178:3474
Looking for brian_tring in default (domain obnauticus.kicks-ass.org)
list_route: hop: <sip:18000111@67.228.46.178>

<--- Transmitting (NAT) to 67.228.46.178:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;received=67.228.46.178;rport=5060
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:brian_tring@67.160.183.109>
Content-Length: 0


<------------>
    -- Executing [brian_tring@default:1] Set("SIP/67.228.46.178-081c57f0", "CALLERID(name)=TringMe") in new stack
    -- Executing [brian_tring@default:2] Dial("SIP/67.228.46.178-081c57f0", "sip/*9|15") in new stack
Audio is at 67.160.183.109 port 10726
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 76.105.250.53:5060:
INVITE sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec SIP/2.0
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport
From: "TringMe" <sip:18000111@67.160.183.109>;tag=as07c6cd9e
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>
Contact: <sip:18000111@67.160.183.109>
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Jan 2008 06:59:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 2516 2516 IN IP4 67.160.183.109
s=session
c=IN IP4 67.160.183.109
t=0 0
m=audio 10726 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called *9
asterisk*CLI>
<--- SIP read from 76.105.250.53:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport=5060
Contact: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>;tag=5f323a2f
From: "TringMe"<sip:18000111@67.160.183.109>;tag=as07c6cd9e
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/*9-082073c8 is ringing

<--- Transmitting (NAT) to 67.228.46.178:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;received=67.228.46.178;rport=5060
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>;tag=as6aa36c15
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:brian_tring@67.160.183.109>
Content-Length: 0


<------------>
asterisk*CLI>
<--- SIP read from 67.228.46.178:5060 --->
CANCEL sip:brian_tring@obnauticus.kicks-ass.org SIP/2.0
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;rport
Max-Forwards: 70
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 CANCEL
User-Agent: "X-Lite release 1011s stamp 41150"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 67.228.46.178 : 5060 (NAT)

<--- Reliably Transmitting (NAT) to 67.228.46.178:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;received=67.228.46.178;rport=5060
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>;tag=as6aa36c15
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 67.228.46.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;received=67.228.46.178;rport=5060
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>;tag=as6aa36c15
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:brian_tring@67.160.183.109>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c795e0b79e33ff549c72be61d407495@67.160.183.109' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 76.105.250.53:5060:
CANCEL sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec SIP/2.0
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport
From: "TringMe" <sip:18000111@67.160.183.109>;tag=as07c6cd9e
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '3c795e0b79e33ff549c72be61d407495@67.160.183.109' in 32000 ms (Method: INVITE)
  == Spawn extension (default, brian_tring, 2) exited non-zero on 'SIP/67.228.46.178-081c57f0'
    -- Executing [h@default:1] Hangup("SIP/67.228.46.178-081c57f0", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/67.228.46.178-081c57f0'
asterisk*CLI>
<--- SIP read from 67.228.46.178:5060 --->
ACK sip:brian_tring@obnauticus.kicks-ass.org SIP/2.0
Via: SIP/2.0/UDP 67.228.46.178:5060;branch=z9hG4bK247e7618;rport
Max-Forwards: 70
From: "TringMe" <sip:18000111@67.228.46.178>;tag=as47854d70
To: <sip:brian_tring@obnauticus.kicks-ass.org>;tag=as6aa36c15
Contact: <sip:18000111@67.228.46.178>
Call-ID: 1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178
CSeq: 102 ACK
User-Agent: "X-Lite release 1011s stamp 41150"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1c3b262d0f10fc1f7bde87eb5508dc4c@67.228.46.178' Method: ACK
asterisk*CLI>
<--- SIP read from 76.105.250.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport=5060
Contact: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>;tag=5f323a2f
From: "TringMe"<sip:18000111@67.160.183.109>;tag=as07c6cd9e
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 CANCEL
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from 76.105.250.53:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport=5060
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>;tag=5f323a2f
From: "TringMe"<sip:18000111@67.160.183.109>;tag=as07c6cd9e
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 76.105.250.53:5060:
ACK sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec SIP/2.0
Via: SIP/2.0/UDP 67.160.183.109:5060;branch=z9hG4bK10244f61;rport
From: "TringMe" <sip:18000111@67.160.183.109>;tag=as07c6cd9e
To: <sip:*9@76.105.250.53:5060;rinstance=bd3b5d83eb5dfbec>;tag=5f323a2f
Contact: <sip:18000111@67.160.183.109>
Call-ID: 3c795e0b79e33ff549c72be61d407495@67.160.183.109
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Really destroying SIP dialog '3c795e0b79e33ff549c72be61d407495@67.160.183.109' Method: INVITE