rendered paste body<-- SIP read from 192.168.50.66:5060:
INVITE sip:827@192.168.28.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bK0269e1a6854620b3402cafeabbadf861.f27106a3
Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBB501CC2
To: <sip:6499651034@192.168.50.66>
From: <sip:6421434103@192.168.50.68>;tag=537FE150-236D
Date: Wed, 14 Nov 2007 20:21:18 GMT
Call-ID: FE24C1E5-922511DC-8488FB35-7AF324D3@192.168.50.68
CSeq: 101 INVITE
Max-Forwards: 68
Expires: 180
ecord-Route: <sip:192.168.50.66:5060;maddr=192.168.50.66>
Contact: <sip:6421434103@192.168.50.68:5060>
Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,COMET,REFER,SUBSCRIBE,NOTIFY,INFO,UPDATE,REGISTER
Supported: 100rel,timer,replaces
Timestamp: 1195071678
User-Agent: Cisco-SIPGateway/IOS-12.x
Min-SE: 1800
Content-Type: application/sdp
Content-Length: 339
Cisco-Guid: 4263661893-2451902940-3118596111-2412718364
Remote-Party-ID: <sip:6421434103@192.168.50.68>;party=calling;screen=yes;privacy=off
Allow-Events: telephone-event
v=0
o=CiscoSystemsSIP-GW-UserAgent 2733 5209 IN IP4 192.168.50.68
s=SIP Call
c=IN IP4 192.168.50.68
t=0 0
m=audio 16552 RTP/AVP 18 8 0 125 101
c=IN IP4 192.168.50.68
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:125 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (22 headers 14 lines) ---
Using INVITE request as basis request - FE24C1E5-922511DC-8488FB35-7AF324D3@192.168.50.68
Sending to 192.168.50.66 : 5060 (non-NAT)
Found peer 'Compass-2'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 125
Found RTP audio format 101
Peer audio RTP is at port 192.168.50.68:16552
Peer video RTP is at port 192.168.50.68:65535
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format X-CCD
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 827 in default (domain 192.168.28.2)
list_route: hop: <sip:192.168.50.66:5060;maddr=192.168.50.66>
list_route: hop: <sip:6421434103@192.168.50.68:5060>
Transmitting (no NAT) to 192.168.50.66:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bK0269e1a6854620b3402cafeabbadf861.f27106a3;received=192.168.50.66
Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBB501CC2
From: <sip:6421434103@192.168.50.68>;tag=537FE150-236D
To: <sip:6499651034@192.168.50.66>
Call-ID: FE24C1E5-922511DC-8488FB35-7AF324D3@192.168.50.68
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:827@192.168.28.2>
Content-Length: 0
---
-- Executing Macro("SIP/6499651037-08289930", "check-state|927") in new stack
-- Executing RealTime("SIP/6499651037-08289930", "example|exten|927|var_") in new stack
-- Executing NoOp("SIP/6499651037-08289930", "1") in new stack
-- Executing GotoIf("SIP/6499651037-08289930", "1?20") in new stack
-- Goto (macro-check-state,s,20)
-- Executing Set("SIP/6499651037-08289930", "XTEN=827") in new stack
-- Executing NoOp("SIP/6499651037-08289930", "827 is available.") in new stack
-- Executing Macro("SIP/6499651037-08289930", "dialuser|SIP/haydnflower") in new stack
-- Executing Set("SIP/6499651037-08289930", "USER=haydnflower") in new stack
-- Executing NoOp("SIP/6499651037-08289930", "Called dial user") in new stack
-- Executing Dial("SIP/6499651037-08289930", "SIP/haydnflower|15|rtM(csr_log|6421434103)") in new stack
We're at 192.168.1.149 port 15056
Video is at 192.168.1.149 port 16682
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.1.197:5060:
INVITE sip:haydnflower@192.168.1.197 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.149:5060;branch=z9hG4bK014a553b;rport
From: "6421434103" <sip:6421434103@192.168.1.149>;tag=as1737a9d7
To: <sip:haydnflower@192.168.1.197>
Contact: <sip:6421434103@192.168.1.149>
Call-ID: 212521710dca39a36e7991110f2592fb@192.168.1.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Nov 2007 20:19:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 28173 28173 IN IP4 192.168.1.149
s=session
c=IN IP4 192.168.1.149
t=0 0
m=audio 15056 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
------------------------------------
--- (22 headers 14 lines) ---
Using INVITE request as basis request - 4CDB3464-922611DC-84DAFB35-7AF324D3@192.168.50.68
Sending to 192.168.50.66 : 5060 (non-NAT)
Found peer 'Compass-2'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 125
Found RTP audio format 101
Peer audio RTP is at port 192.168.50.68:17512
Peer video RTP is at port 192.168.50.68:65535
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format X-CCD
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 099651031 in default (domain 192.168.28.2)
list_route: hop: <sip:192.168.50.66:5060;maddr=192.168.50.66>
list_route: hop: <sip:6421434103@192.168.50.68:5060>
Transmitting (no NAT) to 192.168.50.66:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bKf5ea0796f67b4aa6ab5727bfdb58df1c.e27dab60;received=192.168.50.66
Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBB545F9
From: <sip:6421434103@192.168.50.68>;tag=5381E528-220
To: <sip:6499651027@192.168.50.66>
Call-ID: 4CDB3464-922611DC-84DAFB35-7AF324D3@192.168.50.68
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:099651031@192.168.28.2>
Content-Length: 0
---
-- Executing Goto("SIP/6499651037-08266118", "main|s|1") in new stack
-- Goto (main,s,1)g
-- Executing Ringing("SIP/6499651037-08266118", "") in new stack
Transmitting (no NAT) to 192.168.50.66:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bKf5ea0796f67b4aa6ab5727bfdb58df1c.e27dab60;received=192.168.50.66
Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBB545F9
From: <sip:6421434103@192.168.50.68>;tag=5381E528-220
To: <sip:6499651027@192.168.50.66>;tag=as4720c8cb
Call-ID: 4CDB3464-922611DC-84DAFB35-7AF324D3@192.168.50.68
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:099651031@192.168.28.2>
Content-Length: 0