<-- SIP read from 67.55.159.160:5060:
INVITE sip:s@192.168.1.3:5060 SIP/2.0
Record-Route: <sip:67.55.159.160;ftag=as4bf0df65;lr=on>
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKf414.fb7d8e76.0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK15e54d2d;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>
Contact: <sip:8183550079@67.55.159.156:5060>
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 16
Date: Thu, 21 Dec 2006 19:54:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 11182 11182 IN IP4 67.55.159.156
s=session
c=IN IP4 67.55.159.156
t=0 0
m=audio 19194 RTP/AVP 0 8 3 18 111 5 10 7 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (15 headers 19 lines)---
Using INVITE request as basis request - 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
Sending to 67.55.159.160 : 5060 (NAT)
Found no matching peer or user for '67.55.159.160:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 7
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 67.55.159.156:19194
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G729
Found description format G726-32
Found description format DVI4
Found description format L16
Found description format LPC
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x5fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|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 s in from-sip-external (domain 192.168.1.3)
list_route: hop: <sip:67.55.159.160;ftag=as4bf0df65;lr=on>
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKf414.fb7d8e76.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK15e54d2d;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 102 INVITE
User-Agent: Linksys/SPA2102-3.3.5(a)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.1.3>
Content-Length: 0


---
    -- Executing Ringing("SIP/67.55.159.156-09014f90", "") in new stack
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKf414.fb7d8e76.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK15e54d2d;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>;tag=as13dc0009
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 102 INVITE
User-Agent: Linksys/SPA2102-3.3.5(a)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.1.3>
Content-Length: 0


---
    -- Executing GotoIf("SIP/67.55.159.156-09014f90", "0?from-trunk||1") in new stack
    -- Executing Set("SIP/67.55.159.156-09014f90", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2006-12-21 19:55:05 UTC.
    -- Executing Answer("SIP/67.55.159.156-09014f90", "") in new stack
We're at 192.168.1.3 port 15120
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bKf414.fb7d8e76.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK15e54d2d;rport=5060
Record-Route: <sip:67.55.159.160;ftag=as4bf0df65;lr=on>
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>;tag=as13dc0009
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 102 INVITE
User-Agent: Linksys/SPA2102-3.3.5(a)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.1.3>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 2741 2741 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 15120 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/67.55.159.156-09014f90", "2") in new stack
asterisk1*CLI> 222
<-- SIP read from 67.55.159.160:5060:
ACK sip:s@192.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 67.55.159.160;branch=0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK1d2738f9;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>;tag=as13dc0009
Contact: <sip:8183550079@67.55.159.156:5060>
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0


--- (11 headers 0 lines)---
    -- Executing Playback("SIP/67.55.159.156-09014f90", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
asterisk1*CLI> 222
<-- SIP read from 67.55.159.160:5060:
BYE sip:s@192.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bK0514.8f02ab93.0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK3d2610ae;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>;tag=as13dc0009
Contact: <sip:8183550079@67.55.159.156:5060>
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0


--- (11 headers 0 lines)---
Sending to 67.55.159.160 : 5060 (NAT)
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bK0514.8f02ab93.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK3d2610ae;rport=5060
From: "8183550079" <sip:8183550079@67.55.159.156>;tag=as4bf0df65
To: <sip:6418272550@sipnumber.net>;tag=as13dc0009
Call-ID: 2291852b5446757d2ac4e8df7c7680f2@67.55.159.156
CSeq: 103 BYE
User-Agent: Linksys/SPA2102-3.3.5(a)
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.1.3>
Content-Length: 0


---
  == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/67.55.159.156-09014f90'
    -- Executing NoOp("SIP/67.55.159.156-09014f90", "Hangup") in new stack
    -- Executing Set("SIP/67.55.159.156-09014f90", "DID=s") in new stack
    -- Executing Goto("SIP/67.55.159.156-09014f90", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing Ringing("SIP/67.55.159.156-09014f90", "") in new stack
    -- Executing GotoIf("SIP/67.55.159.156-09014f90", "0?from-trunk|s|1") in new stack
    -- Executing Set("SIP/67.55.159.156-09014f90", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2006-12-21 19:55:09 UTC.
    -- Executing Answer("SIP/67.55.159.156-09014f90", "") in new stack
  == Spawn extension (from-sip-external, s, 4) exited non-zero on 'SIP/67.55.159.156-09014f90'
Destroying call '2291852b5446757d2ac4e8df7c7680f2@67.55.159.156'