
<--- SIP read from 213.166.5.130:5060 --->
INVITE sip:01273808802@85.13.195.164 SIP/2.0
Record-Route: <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C>
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=calling;screen=yes;privacy=off
From: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: MSSGW
Allow: INVITE, BYE, CANCEL, ACK
CSeq: 101 INVITE
Max-Forwards: 14
Timestamp: 1197590680
Contact: <sip:07743898503@213.166.5.134:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 418

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6672 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 18 4 3 98 0 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (22 headers 17 lines) ---
  == Using TOS bits 0
  == Using CoS mark 5
Sending to 213.166.5.130 : 5060 (no NAT)
Using INVITE request as basis request - FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
No user '07743898503' in SIP users list
Found peer 'inbound' for '07743898503' from 213.166.5.130:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format GSM for ID 3
Found audio description format G726-32 for ID 98
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
Looking for 01273808802 in main (domain 85.13.195.164)
list_route: hop: <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C>

<--- Transmitting (no NAT) to 213.166.5.130:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0;received=213.166.5.130
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F
Record-Route: <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C>
From: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 101 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01273808802@85.13.195.164>
Content-Length: 0


<------------>
    -- Executing [01273808802@main:1] Answer("SIP/213.166.5.134-b6c2a160", "") in new stack
Audio is at 85.13.195.164 port 18366
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.0;received=213.166.5.130
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8ED47F
Record-Route: <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C>
From: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>;tag=as007348af
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 101 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01273808802@85.13.195.164>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 983608028 983608028 IN IP4 85.13.195.164
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 85.13.195.164
t=0 0
m=audio 18366 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Executing [01273808802@main:2] Dial("SIP/213.166.5.134-b6c2a160", "SIP/01392213713@outbound") in new stack
  == Using TOS bits 0
  == Using CoS mark 5
Audio is at 85.13.195.164 port 18716
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.166.5.135:5060:
INVITE sip:01392213713@gk.magrathea.net SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Date: Fri, 14 Dec 2007 00:04:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1017897569 1017897569 IN IP4 85.13.195.164
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 85.13.195.164
t=0 0
m=audio 18716 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 01392213713@outbound
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=a3e149d53f0faf6bf88ecc254282dfd9.45bb
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="gk.magrathea.net", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e"
Server: Sip EXpress router (0.8.99-dev (i386/linux))
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 213.166.5.135:5060:
ACK sip:01392213713@gk.magrathea.net SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK62d9ec39;rport
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=a3e149d53f0faf6bf88ecc254282dfd9.45bb
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
Audio is at 85.13.195.164 port 18716
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.166.5.135:5060:
INVITE sip:01392213713@gk.magrathea.net SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:01392213713@gk.magrathea.net", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e", response="441b7ae56d981c76963c5c4e93353e48", opaque=""
Date: Fri, 14 Dec 2007 00:04:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1017897569 1017897570 IN IP4 85.13.195.164
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 85.13.195.164
t=0 0
m=audio 18716 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
ACK sip:01273808802@85.13.195.164:5060 SIP/2.0
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKeb4.82a3dde4.2
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8EE51A
From: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>;tag=as007348af
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Max-Forwards: 14
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 103 INVITE
Server: Sip EXpress router (0.8.99-dev (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:*200441392213713@87.238.72.155:5060>
Record-Route: <sip:87.238.72.153;lr=on;ftag=as6c85b7af>,<sip:213.166.5.135;ftag=as6c85b7af;lr=on>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 6149 2047 IN IP4 87.238.72.155
s=SIP Call
c=IN IP4 87.238.72.155
t=0 0
m=audio 18144 RTP/AVP 8 101
c=IN IP4 87.238.72.155
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (15 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 87.238.72.155:18144
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 87.238.72.155:18144
    -- SIP/outbound-08206e60 is making progress passing it to SIP/213.166.5.134-b6c2a160
set_destination: Parsing <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C> for address/port to send to
set_destination: set destination to 213.166.5.130, port 5060
Audio is at 85.13.195.164 port 18366
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.166.5.130:5060:
INVITE sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport
Route: <sip:213.166.5.130;lr=on;ftag=28BEA090-1D6C>
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 983608028 983608029 IN IP4 87.238.72.155
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 87.238.72.155
t=0 0
m=audio 18144 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 INVITE
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
list_route: hop: <sip:07743898503@213.166.5.134:5060>
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5b07a085;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6c3fb7d2;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK18572230;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK67f02392;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK3c5ae694;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK64b2294b;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:*200441392213713@87.238.72.155:5060>
Record-Route: <sip:87.238.72.153;lr=on;ftag=as6c85b7af>,<sip:213.166.5.135;ftag=as6c85b7af;lr=on>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 6149 2047 IN IP4 87.238.72.155
s=SIP Call
c=IN IP4 87.238.72.155
t=0 0
m=audio 18144 RTP/AVP 8 101
c=IN IP4 87.238.72.155
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 87.238.72.155:18144
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 87.238.72.155:18144
--- set_address_from_contact host '87.238.72.155'
list_route: hop: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>
list_route: hop: <sip:87.238.72.153;lr=on;ftag=as6c85b7af>
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Transmitting (no NAT) to 213.166.5.135:5060:
ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK686cf6a3;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
    -- SIP/outbound-08206e60 answered SIP/213.166.5.134-b6c2a160
    -- Native bridging SIP/213.166.5.134-b6c2a160 and SIP/outbound-08206e60
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Audio is at 85.13.195.164 port 18716
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.166.5.135:5060:
INVITE sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 104 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1017897569 1017897571 IN IP4 213.166.5.134
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 104 INVITE
Server: Sip EXpress router (0.8.99-dev (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK5faf1434;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Date: Fri, 14 Dec 2007 00:04:51 gmt
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 104 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:*200441392213713@87.238.72.155:5060>
Record-Route: <sip:87.238.72.153;lr=on;ftag=as6c85b7af>,<sip:213.166.5.135;ftag=as6c85b7af;lr=on>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 6149 2048 IN IP4 87.238.72.155
s=SIP Call
c=IN IP4 87.238.72.155
t=0 0
m=audio 18144 RTP/AVP 8 101
c=IN IP4 87.238.72.155
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 87.238.72.155:18144
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 87.238.72.155:18144
--- set_address_from_contact host '87.238.72.155'
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Transmitting (no NAT) to 213.166.5.135:5060:
ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK33852ac0;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 104 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK155ac28b;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK6334bbd2;rport=5060
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:07743898503@213.166.5.134>;party=called;screen=yes;privacy=off
Contact: <sip:07743898503@213.166.5.134:5060>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4487 6673 IN IP4 213.166.5.134
s=SIP Call
c=IN IP4 213.166.5.134
t=0 0
m=audio 22614 RTP/AVP 8 101
c=IN IP4 213.166.5.134
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.134:22614
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.166.5.134:22614
--- set_address_from_contact host '213.166.5.134'
set_destination: Parsing <sip:07743898503@213.166.5.134:5060> for address/port to send to
set_destination: set destination to 213.166.5.134, port 5060
Transmitting (no NAT) to 213.166.5.134:5060:
ACK sip:07743898503@213.166.5.134:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK241d5a13;rport
Max-Forwards: 70
From: <sip:441273808802@n.e164.org.uk>;tag=as007348af
To: "07743898503" <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
Contact: <sip:01273808802@85.13.195.164>
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
turing*CLI>
<--- SIP read from 213.166.5.130:5060 --->
BYE sip:01273808802@85.13.195.164:5060 SIP/2.0
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbb4.9ba89a77.0
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8F82687
From: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>;tag=as007348af
Date: Fri, 14 Dec 2007 00:04:40 gmt
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 14
Timestamp: 1197590700
CSeq: 102 BYE
Reason: Q.850;cause=127
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 213.166.5.130 : 5060 (no NAT)

<--- Transmitting (no NAT) to 213.166.5.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.166.5.130;branch=z9hG4bKbb4.9ba89a77.0;received=213.166.5.130
Via: SIP/2.0/UDP 213.166.5.134:5060;branch=z9hG4bK9FD8F82687
From: <sip:07743898503@213.166.5.134>;tag=28BEA090-1D6C
To: <sip:441273808802@n.e164.org.uk>;tag=as007348af
Call-ID: FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134
CSeq: 102 BYE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01273808802@85.13.195.164>
Content-Length: 0


<------------>
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Audio is at 85.13.195.164 port 18716
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.166.5.135:5060:
INVITE sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 105 INVITE
User-Agent: Asterisk PBX SVN-trunk-r92855
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1017897569 1017897572 IN IP4 85.13.195.164
s=Asterisk PBX SVN-trunk-r92855
c=IN IP4 85.13.195.164
t=0 0
m=audio 18716 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' in 32000 ms (Method: INVITE)
  == Spawn extension (main, 01273808802, 2) exited non-zero on 'SIP/213.166.5.134-b6c2a160'
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 105 INVITE
Server: Sip EXpress router (0.8.99-dev (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'FFEA6627-A90E11DC-B8CBD4B2-5B8D3061@213.166.5.134' Method: BYE
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK790e4a2c;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Date: Fri, 14 Dec 2007 00:05:00 gmt
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 105 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:*200441392213713@87.238.72.155:5060>
Record-Route: <sip:87.238.72.153;lr=on;ftag=as6c85b7af>,<sip:213.166.5.135;ftag=as6c85b7af;lr=on>
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 6149 2049 IN IP4 87.238.72.155
s=SIP Call
c=IN IP4 87.238.72.155
t=0 0
m=audio 18144 RTP/AVP 8 101
c=IN IP4 87.238.72.155
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 87.238.72.155:18144
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 87.238.72.155:18144
--- set_address_from_contact host '87.238.72.155'
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Transmitting (no NAT) to 213.166.5.135:5060:
ACK sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK20d2886a;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Contact: <sip:07743898503@85.13.195.164>
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 105 ACK
User-Agent: Asterisk PBX SVN-trunk-r92855
Content-Length: 0


---
set_destination: Parsing <sip:213.166.5.135;ftag=as6c85b7af;lr=on> for address/port to send to
set_destination: set destination to 213.166.5.135, port 5060
Reliably Transmitting (no NAT) to 213.166.5.135:5060:
BYE sip:*200441392213713@87.238.72.155:5060 SIP/2.0
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK042f8076;rport
Route: <sip:213.166.5.135;ftag=as6c85b7af;lr=on>,<sip:87.238.72.153;lr=on;ftag=as6c85b7af>
Max-Forwards: 70
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
CSeq: 106 BYE
User-Agent: Asterisk PBX SVN-trunk-r92855
Proxy-Authorization: Digest username="ordersip", realm="gk.magrathea.net", algorithm=MD5, uri="sip:*200441392213713@87.238.72.155:5060", nonce="4761c9d7aef4305d8c15e55557646250a9e49c1e", response="646daec82d933d60dd61366dd0a1d973", opaque=""
Content-Length: 0


---
Scheduling destruction of SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' in 32000 ms (Method: INVITE)
turing*CLI>
<--- SIP read from 213.166.5.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.13.195.164:5060;branch=z9hG4bK042f8076;rport=5060
From: "07743898503" <sip:07743898503@85.13.195.164>;tag=as6c85b7af
To: <sip:01392213713@gk.magrathea.net>;tag=39AE1F04-F6D
Date: Fri, 14 Dec 2007 00:05:00 gmt
Call-ID: 25e865a71ceadeb207ee702953649f68@85.13.195.164
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 106 BYE


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '25e865a71ceadeb207ee702953649f68@85.13.195.164' Method: INVITE
turing*CLI>