Sending to 66.241.96.221 : 5060 (no NAT)
Using INVITE request as basis request - 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
No matching peer for '9493072156' from '66.241.96.221:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.241.96.221:12158
Looking for 7144514219 in from-sip-external (domain 192.168.15.182)
list_route: hop: <sip:9493072156@66.241.96.221:5060>

<--- Transmitting (NAT) to 66.241.96.221:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK059625d0;received=66.241.96.221;rport=5060
From: "9493072156" <sip:9493072156@66.241.96.221:5060>;tag=as62aa6ce6
To: <sip:7144514219@192.168.15.182:5060>
Call-ID: 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7144514219@192.168.15.182>
Content-Length: 0


<------------>
    -- Executing [7144514219@from-sip-external:1] NoOp("SIP/66.241.96.221:5060-00000001", "Received incoming SIP connection from unknown peer to 7144514219") in new stack
    -- Executing [7144514219@from-sip-external:2] Set("SIP/66.241.96.221:5060-00000001", "DID=7144514219") in new stack
    -- Executing [7144514219@from-sip-external:3] Goto("SIP/66.241.96.221:5060-00000001", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/66.241.96.221:5060-00000001", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/66.241.96.221:5060-00000001", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2011-10-16 15:40:47.537 PDT.
    -- Executing [s@from-sip-external:6] Answer("SIP/66.241.96.221:5060-00000001", "") in new stack
Audio is at 192.168.15.182 port 10362
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<--- Reliably Transmitting (NAT) to 66.241.96.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK059625d0;received=66.241.96.221;rport=5060
From: "9493072156" <sip:9493072156@66.241.96.221:5060>;tag=as62aa6ce6
To: <sip:7144514219@192.168.15.182:5060>;tag=as45eddc4e
Call-ID: 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7144514219@192.168.15.182>
Content-Type: application/sdp
Content-Length: 302

v=0
o=root 1200314749 1200314749 IN IP4 192.168.15.182
s=Asterisk PBX 1.6.2.5-0ubuntu1.4
c=IN IP4 192.168.15.182
t=0 0
m=audio 10362 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
pbx*CLI>
<--- SIP read from UDP:66.241.96.221:5060 --->
ACK sip:7144514219@192.168.15.182:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK6822401a;rport
From: "9493072156" <sip:9493072156@66.241.96.221:5060>;tag=as62aa6ce6
To: <sip:7144514219@192.168.15.182:5060>;tag=as45eddc4e
Contact: <sip:9493072156@66.241.96.221:5060>
Call-ID: 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Executing [s@from-sip-external:7] Wait("SIP/66.241.96.221:5060-00000001", "2") in new stack
    -- Executing [s@from-sip-external:8] Playback("SIP/66.241.96.221:5060-00000001", "ss-noservice") in new stack
    -- <SIP/66.241.96.221:5060-00000001> Playing 'ss-noservice.gsm' (language 'en')
Really destroying SIP dialog '0e09c77773efdf676145f65207a277d0@127.0.1.1' Method: REGISTER
    -- Executing [s@from-sip-external:9] PlayTones("SIP/66.241.96.221:5060-00000001", "congestion") in new stack
    -- Executing [s@from-sip-external:10] Congestion("SIP/66.241.96.221:5060-00000001", "5") in new stack
pbx*CLI>
<--- SIP read from UDP:66.241.96.221:5060 --->
BYE sip:7144514219@192.168.15.182:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK1870098f;rport
From: "9493072156" <sip:9493072156@66.241.96.221:5060>;tag=as62aa6ce6
To: <sip:7144514219@192.168.15.182:5060>;tag=as45eddc4e
Call-ID: 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 66.241.96.221 : 5060 (NAT)

<--- Transmitting (NAT) to 66.241.96.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK1870098f;received=66.241.96.221;rport=5060
From: "9493072156" <sip:9493072156@66.241.96.221:5060>;tag=as62aa6ce6
To: <sip:7144514219@192.168.15.182:5060>;tag=as45eddc4e
Call-ID: 0c23a75813bc91e25d99fe080c721d8d@66.241.96.221
CSeq: 103 BYE
Server: FPBX-2.9.0(1.6.2.5)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/66.241.96.221:5060-00000001'
    -- Executing [h@from-sip-external:1] Hangup("SIP/66.241.96.221:5060-00000001", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/66.241.96.221:5060-00000001'
Really destroying SIP dialog '0c23a75813bc91e25d99fe080c721d8d@66.241.96.221' Method: BYE
[Oct 16 15:40:50] NOTICE[2964]: chan_sip.c:11588 sip_reregister:    -- Re-registration for  5sta_1@inbound27.vitelity.net
       > doing dnsmgr_lookup for 'inbound27.vitelity.net'
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 66.241.96.221:5060:
REGISTER sip:inbound27.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.15.182:5060;branch=z9hG4bK0c2b5e2d;rport
Max-Forwards: 70
From: <sip:5sta_1@inbound27.vitelity.net>;tag=as6edfecd8
To: <sip:5sta_1@inbound27.vitelity.net>
Call-ID: 0e09c77773efdf676145f65207a277d0@127.0.1.1
CSeq: 110 REGISTER
User-Agent: FPBX-2.9.0(1.6.2.5)
Authorization: Digest username="5sta_1", realm="asterisk", algorithm=MD5, uri="sip:inbound27.vitelity.net", nonce="52b9148c", response="15a0e1a42ae696df925b8456f628381d"
Expires: 120
Contact: <sip:s@192.168.15.182>
Content-Length: 0