<--- SIP read from 10.1.1.10:5060 --->
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edff55c71fd8abf6f
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 1 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=Tornado-M20 144768429195310 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 20332 RTP/AVP 18 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (11 headers 13 lines) ---
Sending to 10.1.1.10 : 5060 (no NAT)
Using INVITE request as basis request - 25dea83edf46a590fd8abf6e@10.1.1.10

<--- Reliably Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edff55c71fd8abf6f;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>;tag=as377f2687
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e3b6309"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '25dea83edf46a590fd8abf6e@10.1.1.10' in 32000 ms (Method: INVITE)
Found user '100'
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
ACK sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edff55c71fd8abf6f
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>;tag=as377f2687
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 1 ACK
User-Agent: Lenoxa.com Tornado-M20


<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edf8116f1fd8abf6c
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 2 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>
Content-Type: application/sdp
Content-Length: 290
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="0e3b6309",uri="sip:101@10.1.1.7:5060",response="8c01e39c52e359c62fd9ce39a803a8bb",algorithm=MD5

v=0
o=Tornado-M20 144768429195310 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 20332 RTP/AVP 18 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 13 lines) ---
Sending to 10.1.1.10 : 5060 (NAT)
Using INVITE request as basis request - 25dea83edf46a590fd8abf6e@10.1.1.10
Found user '100'
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.10:20332
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format G726-32 for ID 97
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.10:20332
Peer video RTP is at port 10.1.1.10:61184
Looking for 101 in from-internal (domain 10.1.1.7)
list_route: hop: <sip:100@10.1.1.10:5060>

<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edf8116f1fd8abf6c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0


<------------>
Video is at 10.1.1.9 port 16396
Audio is at 10.1.1.9 port 19826
Adding codec 0x100 (g729) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:15:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19826 RTP/AVP 18 101
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
m=video 16396 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---

<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edf8116f1fd8abf6c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>;tag=as4c312661
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0


<------------>
Retransmitting #1 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:15:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19826 RTP/AVP 18 101
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
m=video 16396 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
Retransmitting #2 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:15:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19826 RTP/AVP 18 101
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
m=video 16396 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
Retransmitting #3 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:15:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19826 RTP/AVP 18 101
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
m=video 16396 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>


<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
list_route: hop: <sip:101@10.1.1.7:5060>
set_destination: Parsing <sip:101@10.1.1.7:5060> for address/port to send to
set_destination: set destination to 10.1.1.7, port 5060
Transmitting (NAT) to 10.1.1.7:5060:
ACK sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK2d1540f8;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 ACK 
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.1.1.9 port 18290
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83edf8116f1fd8abf6c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>;tag=as4c312661
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
t=0 0
m=audio 18290 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
=sendrecvCLI> 

<------------>

<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK11667c3f;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=Tornado-M20 222889596158018 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 23272 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 17242 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2

<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:23272
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer 
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:23272
Peer video RTP is at port 10.1.1.7:17242
set_destination: Parsing <sip:101@10.1.1.7:5060> for address/port to send to
set_destination: set destination to 10.1.1.7, port 5060
Transmitting (NAT) to 10.1.1.7:5060:
ACK sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK1a8075cd;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
To: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
Contact: <sip:100@10.1.1.9>
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
ACK sip:101@10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ec2d019f1fd8abf6d
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
To: <sip:101@10.1.1.7>;tag=as4c312661
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 2 ACK
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="0e3b6309",uri="sip:101@10.1.1.9",response="9dbbd3cd452916741bc684a411723b2f",algorithm=MD5


<------------->
--- (9 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2edae3b1fd8abfda
Max-Forwards: 70
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 99 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>;expires=60
Expires: 60
Authorization: Digest username="101",realm="asterisk",nonce="2f87ad99",uri="sip:10.1.1.9",response="782dbcf5fedf4d065ff6558231328571",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.7 : 5060 (no NAT)

<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2edae3b1fd8abfda;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 99 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2edae3b1fd8abfda;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>;tag=as4c712fd6
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 99 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c14ddc4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a699a4cd354fcb30fd8abe32@10.1.1.7' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2e971621fd8abfdb
Max-Forwards: 70
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 100 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>;expires=60
Expires: 60
Authorization: Digest username="101",realm="asterisk",nonce="5c14ddc4",uri="sip:10.1.1.9",response="5a49db1c5fdd33ac0860d10a804716e5",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.7 : 5060 (NAT)

<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2e971621fd8abfdb;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 100 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0


<------------>
trixbox1*CLI> 
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2e971621fd8abfdb;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>;tag=as4c712fd6
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 100 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:101@10.1.1.7:5060>;expires=60
Date: Wed, 10 Oct 2007 17:15:41 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a699a4cd354fcb30fd8abe32@10.1.1.7' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from 10.1.1.7:5060 --->
BYE sip:100@10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2d846ab1fd8abfd8
Max-Forwards: 70
From: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
To: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 1 BYE
User-Agent: Lenoxa.com Tornado-M20


<------------->
--- (8 headers 0 lines) ---
Sending to 10.1.1.7 : 5060 (NAT)

<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53e2d846ab1fd8abfd8;received=10.1.1.7
From: <sip:101@10.1.1.7:5060>;tag=a699a53e2712cd73fd8abfdd
To: "Pruebas 1" <sip:100@10.1.1.9>;tag=as3c6a561c
Call-ID: 0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0


<------------>
Really destroying SIP dialog '0b91ab8074eb368139c3d2e41ba31b95@10.1.1.9' Method: BYE
Scheduling destruction of SIP dialog '25dea83edf46a590fd8abf6e@10.1.1.10' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:100@10.1.1.10:5060> for address/port to send to
set_destination: set destination to 10.1.1.10, port 5060
Reliably Transmitting (NAT) to 10.1.1.10:5060:
BYE sip:100@10.1.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK5d796487;rport
From: <sip:101@10.1.1.7>;tag=as4c312661
To: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK5d796487;rport=5060;received=10.1.1.9
From: <sip:101@10.1.1.7>;tag=as4c312661
To: "100" <sip:100@10.1.1.10>;tag=25dea83ed848f492fd8abf61
Call-ID: 25dea83edf46a590fd8abf6e@10.1.1.10
CSeq: 102 BYE
User-Agent: Lenoxa.com Tornado-M20


<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '25dea83edf46a590fd8abf6e@10.1.1.10' Method: ACK
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecab5efb1fd8abf6a
Max-Forwards: 70
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 85 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>;expires=60
Expires: 60
Authorization: Digest username="100",realm="asterisk",nonce="1189b994",uri="sip:10.1.1.9",response="00cb47fb3a4bfce459d8b2d714497f38",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.10 : 5060 (no NAT)

<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecab5efb1fd8abf6a;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 85 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecab5efb1fd8abf6a;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>;tag=as0e67ce58
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 85 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="539e8dcf"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '25dea836b6dd54c0fd8abe37@10.1.1.10' in 32000 ms (Method: REGISTER)
trixbox1*CLI> 
<--- SIP read from 10.1.1.10:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecad628d1fd8abf6b
Max-Forwards: 70
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 86 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>;expires=60
Expires: 60
Authorization: Digest username="100",realm="asterisk",nonce="539e8dcf",uri="sip:10.1.1.9",response="034a29b2591c080cde5949a65e78e103",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.10 : 5060 (NAT)

<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecad628d1fd8abf6b;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 86 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0


<------------>
trixbox1*CLI> 
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83ecad628d1fd8abf6b;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>;tag=as0e67ce58
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 86 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:100@10.1.1.10:5060>;expires=60
Date: Wed, 10 Oct 2007 17:15:45 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '25dea836b6dd54c0fd8abe37@10.1.1.10' in 32000 ms (Method: REGISTER)