rendered paste body<--- 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=z9hG4bK25dea83034a90041fd8abf2f
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>
Call-ID: 25dea830344d8fe0fd8abf2e@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 144768401604646 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 26896 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 - 25dea830344d8fe0fd8abf2e@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=z9hG4bK25dea83034a90041fd8abf2f;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as3196d415
Call-ID: 25dea830344d8fe0fd8abf2e@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="5a29cd53"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '25dea830344d8fe0fd8abf2e@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=z9hG4bK25dea83034a90041fd8abf2f
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as3196d415
Call-ID: 25dea830344d8fe0fd8abf2e@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=z9hG4bK25dea830344bfd61fd8abf2c
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>
Call-ID: 25dea830344d8fe0fd8abf2e@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="5a29cd53",uri="sip:101@10.1.1.7:5060",response="a90b793beb6e05208d7a71f3e25b9ccb",algorithm=MD5
v=0
o=Tornado-M20 144768401604646 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 26896 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 - 25dea830344d8fe0fd8abf2e@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 10.1.1.10:26896
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 10.1.1.10:26896
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=z9hG4bK25dea830344bfd61fd8abf2c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>
Call-ID: 25dea830344d8fe0fd8abf2e@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 12922
Audio is at 10.1.1.9 port 15096
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=z9hG4bK21b96388;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:08:32 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 15096 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 12922 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=z9hG4bK25dea830344bfd61fd8abf2c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@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=z9hG4bK21b96388;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:08:32 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 15096 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 12922 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=z9hG4bK21b96388;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:08:32 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 15096 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 12922 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=z9hG4bK21b96388;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:08:32 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 15096 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 12922 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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 222889568567354 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 17118 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 23718 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 10.1.1.7:17118
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 10.1.1.7:17118
Peer video RTP is at port 10.1.1.7:23718
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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 222889568567354 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 17118 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 23718 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 10.1.1.7:17118
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 10.1.1.7:17118
Peer video RTP is at port 10.1.1.7:23718
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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 222889568567354 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 17118 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 23718 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 10.1.1.7:17118
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 10.1.1.7:17118
Peer video RTP is at port 10.1.1.7:23718
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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 222889568567354 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 17118 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 23718 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 10.1.1.7:17118
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 10.1.1.7:17118
Peer video RTP is at port 10.1.1.7:23718
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=z9hG4bK21b96388;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 222889568567354 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 17118 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 23718 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 10.1.1.7:17118
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 10.1.1.7:17118
Peer video RTP is at port 10.1.1.7:23718
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=z9hG4bK37da3d1e;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Contact: <sip:100@10.1.1.9>
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@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 10266
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=z9hG4bK25dea830344bfd61fd8abf2c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@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 10266 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>
<------------>
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
CANCEL sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830344bfd61fd8abf2c
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 2 CANCEL
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="5a29cd53",uri="sip:101@10.1.1.9",response="b6bcc4e879e57061341ba801addffe71",algorithm=MD5
trixbox1*CLI>
<------------->
--- (9 headers 0 lines) ---
Sending to 10.1.1.10 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830344bfd61fd8abf2c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 2 CANCEL
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>
<--- SIP read from 10.1.1.10:5060 --->
CANCEL sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830344bfd61fd8abf2c
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 2 CANCEL
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="5a29cd53",uri="sip:101@10.1.1.9",response="b6bcc4e879e57061341ba801addffe71",algorithm=MD5
trixbox1*CLI>
<------------->
--- (9 headers 0 lines) ---
Sending to 10.1.1.10 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830344bfd61fd8abf2c;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 2 CANCEL
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>
<--- 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=z9hG4bK25dea83038ca8621fd8abf2d
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 2 ACK
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="5a29cd53",uri="sip:101@10.1.1.9",response="58d4429004a03ae69ccc29a234dc2098",algorithm=MD5
<------------->
--- (9 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
BYE sip:101@10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830381adb71fd8abf2a
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 3 BYE
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="5a29cd53",uri="sip:101@10.1.1.9",response="6c646e5a61744468d3eee7d446fe7d17",algorithm=MD5
<------------->
--- (9 headers 0 lines) ---
Sending to 10.1.1.10 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea830381adb71fd8abf2a;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83034e40a52fd8abf21
To: <sip:101@10.1.1.7>;tag=as5fbfb607
Call-ID: 25dea830344d8fe0fd8abf2e@10.1.1.10
CSeq: 3 BYE
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
<------------>
Scheduling destruction of SIP dialog '6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9' in 6400 ms (Method: INVITE)
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
Reliably Transmitting (NAT) to 10.1.1.7:5060:
BYE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK474db0a5;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as48dec991
To: <sip:101@10.1.1.7:5060>;tag=a699a5319c4c95f3fd8abf86
Call-ID: 6d0dad92052fdd0d22a9fd98404e5684@10.1.1.9
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0