<------------->
Reliably Transmitting (no NAT) to 81.136.145.230:14241:
OPTIONS sip:901@81.136.145.230:14241;transport=udp;rinstance=d79365cdba114e76 SIP/2.0
Via: SIP/2.0/UDP 217.172.136.164:5060;branch=z9hG4bK79fd82ad
Max-Forwards: 70
From: "Unknown" <sip:Unknown@217.172.136.164>;tag=as01747e06
To: <sip:901@81.136.145.230:14241;transport=udp;rinstance=d79365cdba114e76>
Contact: <sip:Unknown@217.172.136.164:5060>
Call-ID: 2170989c0eb0ed136c59f63929f7473b@217.172.136.164:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Wed, 04 Jan 2012 15:43:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:81.136.145.230:14241 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.172.136.164:5060;branch=z9hG4bK79fd82ad
Contact: <sip:192.168.100.27:47750>
To: <sip:901@81.136.145.230:14241;transport=udp;rinstance=d79365cdba114e76>;tag=b2637e65
From: "Unknown"<sip:Unknown@217.172.136.164>;tag=as01747e06
Call-ID: 2170989c0eb0ed136c59f63929f7473b@217.172.136.164:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '2170989c0eb0ed136c59f63929f7473b@217.172.136.164:5060' Method: OPTIONS

<--- SIP read from UDP:77.107.134.244:5060 --->
INVITE sip:448435571092@217.172.136.164:5060 SIP/2.0
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK508597ad;rport
Max-Forwards: 70
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>
Contact: <sip:08435571092@77.107.134.244>
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 102 INVITE
User-Agent: VoiceHost PBX v3
Date: Wed, 04 Jan 2012 15:42:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 755820825 755820825 IN IP4 77.107.134.244
s=VoiceHost PBX v3
c=IN IP4 77.107.134.244
b=CT:384
t=0 0
m=audio 15828 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 19572 RTP/AVP 34 98 99
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (14 headers 19 lines) ---
Sending to 77.107.134.244:5060 (no NAT)
Using INVITE request as basis request - 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
No matching peer for '08435571092' from '77.107.134.244:5060'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
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 telephone-event for ID 101
Found RTP video format 34
Found RTP video format 98
Found RTP video format 99
Found video description format H263 for ID 34
Found video description format h263-1998 for ID 98
Found video description format H264 for ID 99
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 77.107.134.244:15828
Looking for 448435571092 in from-sip-external (domain 217.172.136.164:5060)
list_route: hop: <sip:08435571092@77.107.134.244>

<--- Transmitting (no NAT) to 77.107.134.244:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK508597ad;received=77.107.134.244;rport=5060
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:448435571092@217.172.136.164:5060>
Content-Length: 0


<------------>
    -- Executing [448435571092@from-sip-external:1] NoOp("SIP/sip.voicehost.co.uk-000000c0", "Received incoming SIP connection from unknown peer to 448435571092") in new stack
    -- Executing [448435571092@from-sip-external:2] Set("SIP/sip.voicehost.co.uk-000000c0", "DID=448435571092") in new stack
    -- Executing [448435571092@from-sip-external:3] Goto("SIP/sip.voicehost.co.uk-000000c0", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/sip.voicehost.co.uk-000000c0", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/sip.voicehost.co.uk-000000c0", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-01-04 15:43:48.123 GMT.
    -- Executing [s@from-sip-external:6] Answer("SIP/sip.voicehost.co.uk-000000c0", "") in new stack
Audio is at 5060
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 (no NAT) to 77.107.134.244:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK508597ad;received=77.107.134.244;rport=5060
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>;tag=as78b25074
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:448435571092@217.172.136.164:5060>
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 1390126491 1390126491 IN IP4 217.172.136.164
s=Asterisk PBX 1.8.7.1
c=IN IP4 217.172.136.164
t=0 0
m=audio 16344 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=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34 98 99

<------------>

<--- SIP read from UDP:77.107.134.244:5060 --->
ACK sip:448435571092@217.172.136.164:5060 SIP/2.0
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK06cf0955;rport
Max-Forwards: 70
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>;tag=as78b25074
Contact: <sip:08435571092@77.107.134.244>
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 102 ACK
User-Agent: VoiceHost PBX v3
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Executing [s@from-sip-external:7] Wait("SIP/sip.voicehost.co.uk-000000c0", "2") in new stack
[2012-01-04 15:43:33] NOTICE[11730]: channel.c:4147 __ast_read: Dropping incompatible voice frame on SIP/sip.voicehost.co.uk-000000c0 of format ulaw since our native format has changed to 0x8 (alaw)
    -- Executing [s@from-sip-external:8] Playback("SIP/sip.voicehost.co.uk-000000c0", "ss-noservice") in new stack
    -- <SIP/sip.voicehost.co.uk-000000c0> Playing 'ss-noservice.alaw' (language 'en')

<--- SIP read from UDP:77.107.134.244:5060 --->
BYE sip:448435571092@217.172.136.164:5060 SIP/2.0
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK3c29d733;rport
Max-Forwards: 70
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>;tag=as78b25074
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 103 BYE
User-Agent: VoiceHost PBX v3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

------------->
--- (11 headers 0 lines) ---
Sending to 77.107.134.244:5060 (no NAT)
Scheduling destruction of SIP dialog '0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 77.107.134.244:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.107.134.244:5060;branch=z9hG4bK3c29d733;received=77.107.134.244;rport=5060
From: "08435571092" <sip:08435571092@sip.voicehost.co.uk>;tag=as775870da
To: <sip:448435571092@217.172.136.164:5060>;tag=as78b25074
Call-ID: 0166daa26a8b9b307fd13ca60917bb41@sip.voicehost.co.uk
CSeq: 103 BYE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-sip-external, s, 8) exited non-zero on 'SIP/sip.voicehost.co.uk-000000c0'
    -- Executing [h@from-sip-external:1] Hangup("SIP/sip.voicehost.co.uk-000000c0", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/sip.voicehost.co.uk-000000c0'
ardenham-pbx*CLI>
Disconnected from Asterisk server