rendered paste body<------------->
--- (11 headers 0 lines) ---
[Ktampa3*CLI>
<--- SIP read from 74.100.136.230:1311 --->
INVITE sip:4166289921@tampa3.mixnetworks.net:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>
CSeq: 2 INVITE
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
Contact: <sip:lmadsen5#lmentinc@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="50ca84fc0a8b738de10f12e71b736c5e", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251
v=0
o=- 1165029551 1165029551 IN IP4 192.168.1.103
s=Polycom IP Phone
c=IN IP4 192.168.1.103
t=0 0
a=sendrecv
m=audio 2264 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 11 lines) ---
Sending to 74.100.136.230 : 1311 (NAT)
Using INVITE request as basis request - fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
Found user 'lmadsen5#lmentinc'
[Ktampa3*CLI>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.103:2264
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.103:2264
Looking for 4166289921 in origination (domain tampa3.mixnetworks.net)
list_route: hop: <sip:lmadsen5#lmentinc@192.168.1.103>
<--- Transmitting (NAT) to 74.100.136.230:1311 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:4166289921@8.7.177.3>
Content-Length: 0
<------------>
[Ktampa3*CLI>
Restrict CID: YES
[Ktampa3*CLI>
Really destroying SIP dialog '4ba566434a4c6da9654b168479c9676e@8.7.177.3' Method: INVITE
[Ktampa3*CLI>
Scheduling destruction of SIP dialog '072fec8d5dc80c1b1b49cc1f15f412c4@8.7.177.3' in 6400 ms (Method: INVITE)
Audio is at 8.7.177.3 port 12056
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 8.7.177.102:5060:
INVITE sip:38811278614166289921@8.7.177.102 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK26e1f821;rport
From: "LM Enterprises" <sip:4166289921@8.7.177.3>;tag=as7b869eba
To: <sip:38811278614166289921@8.7.177.102>
Contact: <sip:4166289921@8.7.177.3>
Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "LM Enterprises" <sip:4166289921@8.7.177.3>;privacy=off;screen=no
Date: Sat, 02 Dec 2006 03:19:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 21383 21383 IN IP4 8.7.177.3
s=session
c=IN IP4 8.7.177.3
t=0 0
m=audio 12056 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Ktampa3*CLI>
<--- SIP read from 8.7.177.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.7.177.3;branch=z9hG4bK26e1f821;received=8.7.177.3;rport=5060
From: "LM Enterprises" <sip:4166289921@8.7.177.3>;tag=as7b869eba
To: <sip:38811278614166289921@8.7.177.102>;tag=cba-03fa-4570b6af
Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3
CSeq: 102 INVITE
Date: Fri, 01 Dec 2006 23:11:43 GMT
Server: BRSIP v2.0.0.11
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
[Ktampa3*CLI>
<--- SIP read from 8.7.177.101:5060 --->
INVITE sip:14166289921@8.7.177.3 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051
From: <sip:4166289921@newtampa>;tag=cba-a8b6-4570f051
To: <sip:14166289921@8.7.177.3>
Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101
CSeq: 1165029458 INVITE
Contact: <sip:4166289921@8.7.177.101>
Date: Sat, 02 Dec 2006 03:17:37 GMT
User-Agent: BRSIP v2.0.0.11
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
Allow-Events: keep-alive, message-summary
Supported: timer
Session-Expires: 1800
Min-SE: 900
Expires: 300
Content-Type: application/sdp
Content-Length: 274
v=0
o=BRSDP 501942 501942 IN IP4 66.59.182.210
s=BRSDP Session
c=IN IP4 66.59.182.210
t=0 0
m=audio 17168 RTP/AVP 0 18 4 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (18 headers 12 lines) ---
Sending to 8.7.177.101 : 5060 (no NAT)
Using INVITE request as basis request - 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101
Found peer 'telic101'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 66.59.182.210:17168
Found description format PCMU for ID 0
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.59.182.210:17168
Looking for 14166289921 in from-voip (domain 8.7.177.3)
list_route: hop: <sip:4166289921@8.7.177.101>
<--- Transmitting (no NAT) to 8.7.177.101:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051;received=8.7.177.101
From: <sip:4166289921@newtampa>;tag=cba-a8b6-4570f051
To: <sip:14166289921@8.7.177.3>
Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101
CSeq: 1165029458 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14166289921@8.7.177.3>
Content-Length: 0
<------------>
[Ktampa3*CLI>
Scheduling destruction of SIP dialog '35da68b8301b518962f0e4786965fa54@8.7.177.3' in 13056 ms (Method: INVITE)
[Ktampa3*CLI>
Audio is at 8.7.177.3 port 16182
[Ktampa3*CLI>
Adding codec 0x100 (g729) to SDP
[Ktampa3*CLI>
Adding codec 0x2 (gsm) to SDP
[Ktampa3*CLI>
Adding codec 0x4 (ulaw) to SDP
[Ktampa3*CLI>
Adding codec 0x8 (alaw) to SDP
[Ktampa3*CLI>
Adding codec 0x10 (g726aal2) to SDP
[Ktampa3*CLI>
Adding codec 0x20 (adpcm) to SDP
[Ktampa3*CLI>
Adding codec 0x40 (slin) to SDP
[Ktampa3*CLI>
Adding codec 0x80 (lpc10) to SDP
[Ktampa3*CLI>
Adding codec 0x400 (ilbc) to SDP
[Ktampa3*CLI>
Adding codec 0x800 (g726) to SDP
[Ktampa3*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
[Ktampa3*CLI>
Reliably Transmitting (NAT) to 74.100.136.230:1212:
INVITE sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport
From: "4166289921" <sip:4166289921@8.7.177.3>;tag=as3f6ac536
To: <sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp>
Contact: <sip:4166289921@8.7.177.3>
Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "4166289921" <sip:4166289921@8.7.177.3>;privacy=off;screen=no
Date: Sat, 02 Dec 2006 03:19:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 512
v=0
o=root 21383 21383 IN IP4 8.7.177.3
s=session
c=IN IP4 8.7.177.3
t=0 0
m=audio 16182 RTP/AVP 18 3 0 8 112 5 10 7 97 111 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Ktampa3*CLI>
<--- SIP read from 74.100.136.230:1212 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport
From: "4166289921" <sip:4166289921@8.7.177.3>;tag=as3f6ac536
To: <sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp>
Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3
Date: Sat, 02 Dec 2006 03:19:20 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: <sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[Ktampa3*CLI>
<--- SIP read from 74.100.136.230:1212 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport
From: "4166289921" <sip:4166289921@8.7.177.3>;tag=as3f6ac536
To: <sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp>;tag=00036bdd39202a341e2b724f-43ee9bc1
Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3
Date: Sat, 02 Dec 2006 03:19:20 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: <sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[Ktampa3*CLI>
<--- Transmitting (no NAT) to 8.7.177.101:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051;received=8.7.177.101
From: <sip:4166289921@newtampa>;tag=cba-a8b6-4570f051
To: <sip:14166289921@8.7.177.3>;tag=as4fc2ecf2
Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101
CSeq: 1165029458 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14166289921@8.7.177.3>
Content-Length: 0
<------------>
[Ktampa3*CLI>
<--- SIP read from 8.7.177.102:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 8.7.177.3;branch=z9hG4bK26e1f821;received=8.7.177.3;rport=5060
From: "LM Enterprises" <sip:4166289921@8.7.177.3>;tag=as7b869eba
To: <sip:38811278614166289921@8.7.177.102>;tag=cba-03fa-4570b6af
Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3
CSeq: 102 INVITE
Contact: <sip:38811278614166289921@8.7.177.102>
Date: Fri, 01 Dec 2006 23:11:47 GMT
Server: BRSIP v2.0.0.11
Content-Type: application/sdp
Content-Length: 198
v=0
o=BRSDP 1018 1018 IN IP4 209.244.42.249
s=BRSDP Session
c=IN IP4 209.244.42.249
t=0 0
m=audio 62984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.244.42.249:62984
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.244.42.249:62984
[Ktampa3*CLI>
Audio is at 8.7.177.3 port 17040
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 74.100.136.230:1311 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>;tag=as7a40e8d1
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:4166289921@8.7.177.3>
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 21383 21383 IN IP4 8.7.177.3
s=session
c=IN IP4 8.7.177.3
t=0 0
m=audio 17040 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[Ktampa3*CLI>
Reliably Transmitting (no NAT) to 69.17.122.217:5060:
OPTIONS sip:69.17.122.217 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as23c24796
To: <sip:69.17.122.217>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
[Ktampa3*CLI>
Content-Length: 0
---
[Ktampa3*CLI>
Retransmitting #1 (no NAT) to 69.17.122.217:5060:
OPTIONS sip:69.17.122.217 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as23c24796
To: <sip:69.17.122.217>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Ktampa3*CLI>
Really destroying SIP dialog '072fec8d5dc80c1b1b49cc1f15f412c4@8.7.177.3' Method: INVITE
[Ktampa3*CLI>
Retransmitting #2 (no NAT) to 69.17.122.217:5060:
OPTIONS sip:69.17.122.217 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as23c24796
To: <sip:69.17.122.217>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Ktampa3*CLI>
Retransmitting #3 (no NAT) to 69.17.122.217:5060:
OPTIONS sip:69.17.122.217 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as23c24796
To: <sip:69.17.122.217>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Ktampa3*CLI>
<--- SIP read from 74.100.136.230:1311 --->
CANCEL sip:4166289921@tampa3.mixnetworks.net:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>
CSeq: 2 CANCEL
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
Contact: <sip:lmadsen5#lmentinc@192.168.1.103>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036
Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="50ca84fc0a8b738de10f12e71b736c5e", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 74.100.136.230 : 1311 (NAT)
<--- Reliably Transmitting (NAT) to 74.100.136.230:1311 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>;tag=as7a40e8d1
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 74.100.136.230:1311 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>;tag=as7a40e8d1
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:4166289921@8.7.177.3>
Content-Length: 0
<------------>
[Ktampa3*CLI>
Scheduling destruction of SIP dialog '548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3' in 6400 ms (Method: INVITE)
[Ktampa3*CLI>
<--- SIP read from 74.100.136.230:1311 --->
ACK sip:4166289921@8.7.177.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A
From: "lmadsen5#lmentinc" <sip:lmadsen5#lmentinc@tampa3.mixnetworks.net>;tag=1D2CE228-54CF5249
To: <sip:4166289921@tampa3.mixnetworks.net;user=phone>;tag=as7a40e8d1
CSeq: 2 ACK
Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103
Contact: <sip:lmadsen5#lmentinc@192.168.1.103>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036
Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="afca69fef6bfde509fbd878c65218fda", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog 'fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103' Method: ACK
[Ktampa3*CLI>
Reliably Transmitting (no NAT) to 8.7.177.99:5060:
OPTIONS sip:8.7.177.99 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK60ce581c;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as4f756162
To: <sip:8.7.177.99>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 36d3561d7cc655673b8e4313241efca1@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Ktampa3*CLI>
<--- SIP read from 8.7.177.99:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK60ce581c;rport=5060
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as4f756162
To: <sip:8.7.177.99>;tag=f838092f89d25b94c908dbbca208fb0a.075a
Call-ID: 36d3561d7cc655673b8e4313241efca1@8.7.177.3
CSeq: 102 OPTIONS
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 8.7.177.99:5060 "Noisy feedback tells: pid=3775 req_src_ip=8.7.177.3 req_src_port=5060 in_uri=sip:8.7.177.99 out_uri=sip:8.7.177.99 via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '36d3561d7cc655673b8e4313241efca1@8.7.177.3' Method: OPTIONS
[Ktampa3*CLI>
Reliably Transmitting (no NAT) to 8.7.177.2:5060:
OPTIONS sip:8.7.177.2 SIP/2.0
Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK537ae707;rport
From: "asterisk" <sip:asterisk@8.7.177.3>;tag=as02815eee
To: <sip:8.7.177.2>
Contact: <sip:asterisk@8.7.177.3>
Call-ID: 0ceb0d975cadfa897bbb7be30a259a73@8.7.177.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 Dec 2006 03:19:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0