[sr-dev] git:master: uac: use transaction to propagate uac_auth() flag
Yuriy Gorlichenko
ovoshlook at gmail.com
Mon Nov 3 22:12:19 CET 2014
Daniel, yes! With this patch CSeq increases.
Now We tested it with provider that have kamailio at the remote proxy side.
Today I will test once agin but with other provider, that have not kamailio
as proxy.
I will write about results of tests at this list.
And if you remember - I wrote some weeks ago about problem with AT&T
provider (whenn called party (provider endpoint) picking up phone - our
kamailio sends CANCEL packet. I thought It happen because CSeq not incrased
and remote party answers with wrong CSeq... But now CSeq is valid put ussie
still lives. (Codecs are ok, It was first guess)
I take sip trace and will set it here (bellow). This trace with valid CSeq
and CANCEL from kamailio. If you have some time please look at this.. This
trouble happens only with AT&T.
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129
E....... at .l.
...6........q..INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651
E.......*.Z.6...
.........}.SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com
>;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410
E....... at .o.
...6.........j%ACK sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com
>;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402
E....... at .k.
...6..........`INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri="
sip:98765432100 at phone.myprovider.com", qop=auth, nc=00000001,
cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3",
algorithm=MD5
v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402
E....... at .k.
...6.........._INVITE sip:98765432100 at phone.myprovider.com SIP/2.0
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Contact:<sip:12345678 at sip.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com",
nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri="
sip:98765432100 at phone.myprovider.com", qop=auth, nc=00000001,
cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3",
algorithm=MD5
v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529
E..-....*.[.6...
.........xiSIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529
E..-....*.[.6...
.........xkSIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998
E.......*.Y16...
.........^.SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 862
E..z.Q.. at .p.
...
..).....f.&SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 719
E...-... at .2.
..)
...........SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfe23.1634293c5fc80fc5bd67bdb34a4f3f1a.0;received=10.0.2.4;rport=5068
Via: SIP/2.0/WSS
mhk1a1e768j5.invalid;rport=55902;received=85.21.140.253;branch=z9hG4bK4895645
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=d5f4kgk0kj;lr=on>
From: "skynet.device-3" <sip:skynet.device-3 at sip.myservice.info
>;tag=d5f4kgk0kj
To: <sip:98765432100 at sip.myservice.info>;tag=as5b947c5c
Call-ID: u8399fm4dtpcsjv5mksc
CSeq: 1363 INVITE
Server: Asterisk PBX 12.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:98765432100 at 10.0.1.41:50600>
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986
E.......*.Y<6...
...........SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 850
E..n.R.. at .p.
...
..).....Zj.SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294
E..*....*.X.6...
.........:.SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1184
E....S.. at .o.
...
..)......Z.SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30040 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30041
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 1030
E.."-... at .1.
..)
.........5~SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfe23.1634293c5fc80fc5bd67bdb34a4f3f1a.0;received=10.0.2.4;rport=5068
Via: SIP/2.0/WSS
mhk1a1e768j5.invalid;rport=55902;received=85.21.140.253;branch=z9hG4bK4895645
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=d5f4kgk0kj;lr=on>
From: "skynet.device-3" <sip:skynet.device-3 at sip.myservice.info
>;tag=d5f4kgk0kj
To: <sip:98765432100 at sip.myservice.info>;tag=as5b947c5c
Call-ID: u8399fm4dtpcsjv5mksc
CSeq: 1363 INVITE
Server: Asterisk PBX 12.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:98765432100 at 10.0.1.41:50600>
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 1968867400 1968867400 IN IP4 10.0.1.41
s=Asterisk PBX 12.5.0
c=IN IP4 10.0.1.41
t=0 0
m=audio 13996 RTP/AVP 8 0 126
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282
E.......*.X.6...
........
..SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 103 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50
s=FreeSWITCH
c=IN IP4 173.203.60.50
t=0 0
m=audio 25430 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1172
E....U.. at .o.
...
..)........SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=UrBQrZp9vSm3p
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 173.203.60.50:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
Remote-Party-ID: "98765432100" <sip:98765432100 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415020637 1415020638 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30058 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30059
IP 10.0.2.4.5068 > 176.74.218.73.50600: UDP, length 387
E....... at .D.
....J.I........OPTIONS sip:176.74.218.73:50600 SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfc2f.3a89e451000000000000000000000000.0
To: <sip:176.74.218.73:50600>
From: <sip:proxy at 10.0.2.4>;tag=58e3fe21d19a1a54b2095194d1fd8b5f-165a
CSeq: 10 OPTIONS
Call-ID: 4e9bbcd8278bd272-13326 at 10.0.2.4
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (4.3.0-dev1 (x86_64/linux))
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 379
E....... at .q.
...
..)........OPTIONS sip:10.0.1.41:50600 SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK0d2f.46825c47000000000000000000000000.0
To: <sip:10.0.1.41:50600>
From: <sip:proxy at 10.0.2.4>;tag=58e3fe21d19a1a54b2095194d1fd8b5f-17b2
CSeq: 10 OPTIONS
Call-ID: 4e9bbcd8278bd273-13326 at 10.0.2.4
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (4.3.0-dev1 (x86_64/linux))
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 515
E...-... at .3.
..)
.........|SSIP/2.0 404 Not Found
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK0d2f.46825c47000000000000000000000000.0;received=10.0.2.4;rport=5068
From: <sip:proxy at 10.0.2.4>;tag=58e3fe21d19a1a54b2095194d1fd8b5f-17b2
To: <sip:10.0.1.41:50600>;tag=as7b23cb12
Call-ID: 4e9bbcd8278bd273-13326 at 10.0.2.4
CSeq: 10 OPTIONS
Server: Asterisk PBX 12.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
IP 176.74.218.73.50600 > 10.0.2.4.5068: UDP, length 596
E..p.>..3.,..J.I
........\.-SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfc2f.3a89e451000000000000000000000000.0;received=68.34.22.11;rport=1024
From: <sip:proxy at 10.0.2.4>;tag=58e3fe21d19a1a54b2095194d1fd8b5f-165a
To: <sip:176.74.218.73:50600>;tag=as26ab300c
Call-ID: 4e9bbcd8278bd272-13326 at 10.0.2.4
CSeq: 10 OPTIONS
Server: Asterisk PBX 12.6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="newasterisk.dev.webinar.ru",
nonce="2a3b4448"
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W.6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1318
E..B.>.. at .m0
...
..)......i}SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30040 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30041
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412
E....... at .o.
...6.........j.CANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 CANCEL
Content-Length: 0
Reason: SIP;cause=200
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 580
E..`-... at .3s
..)
........L..ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:50600;branch=z9hG4bK010bb317;rport
Route: <sip:sip.myservice.info:5068
;nat=yes;ftag=as288f6857;lr=on>,<sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 1016
E...-... at .1.
..)
.........,rSIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfe23.1634293c5fc80fc5bd67bdb34a4f3f1a.0;received=10.0.2.4;rport=5068
Via: SIP/2.0/WSS
mhk1a1e768j5.invalid;rport=55902;received=85.21.140.253;branch=z9hG4bK4895645
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=d5f4kgk0kj;lr=on>
From: "skynet.device-3" <sip:skynet.device-3 at sip.myservice.info
>;tag=d5f4kgk0kj
To: <sip:98765432100 at sip.myservice.info>;tag=as5b947c5c
Call-ID: u8399fm4dtpcsjv5mksc
CSeq: 1363 INVITE
Server: Asterisk PBX 12.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:98765432100 at 10.0.1.41:50600>
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 1968867400 1968867400 IN IP4 10.0.1.41
s=Asterisk PBX 12.5.0
c=IN IP4 10.0.1.41
t=0 0
m=audio 13996 RTP/AVP 8 0 126
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6...........ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 564
E..P.[.. at .p.
...
..).....<..ACK sip:98765432100 at 10.0.1.41:50600 SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bKfe23.6c9ecb93d17682ab8c63a2cbf9b0bafa.0
Via: SIP/2.0/WSS
mhk1a1e768j5.invalid;rport=55902;received=85.21.140.253;branch=z9hG4bK4225613
Max-Forwards: 69
To: <sip:98765432100 at sip.myservice.info>;tag=as5b947c5c
From: "skynet.device-3" <sip:skynet.device-3 at sip.myservice.info
>;tag=d5f4kgk0kj
Call-ID: u8399fm4dtpcsjv5mksc
CSeq: 1363 ACK
Allow: ACK,CANCEL,BYE,OPTIONS,UPDATE,INVITE,MESSAGE
Supported: outbound
User-Agent: undefined 0.4.2
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W~6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1318
E..B.b.. at .m.
...
..)......i}SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30040 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30041
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 580
E..`-... at .3p
..)
........L..ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:50600;branch=z9hG4bK1f95579c;rport
Route: <sip:sip.myservice.info:5068
;nat=yes;ftag=as288f6857;lr=on>,<sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........jACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246
E.......*.\.6...
...........OPTIONS sip:68.34.22.11:1024 SIP/2.0
Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0
From: sip:keepalive at myprovider.com;tag=29ebf5d
To: sip:68.34.22.11:1024
Call-ID: 6360a4bb-53bebf7-5686ca6 at 10.202.129.254
CSeq: 1 OPTIONS
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330
E..f.... at .p.
...6........R.<SIP/2.0 405 Method not allowed
Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23
From: sip:keepalive at myprovider.com;tag=29ebf5d
To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7
Call-ID: 6360a4bb-53bebf7-5686ca6 at 10.202.129.254
CSeq: 1 OPTIONS
Server: MS Lync
Content-Length: 0
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W|6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1318
E..B.... at .l.
...
..)......i}SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30040 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30041
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 580
E..`-... at .3o
..)
........L}.ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:50600;branch=z9hG4bK5e2d222b;rport
Route: <sip:sip.myservice.info:5068
;nat=yes;ftag=as288f6857;lr=on>,<sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........eACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200
IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428
E.......*.W{6...
.........H.SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.myservice.info:5068
;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 104 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83
s=FreeSWITCH
c=IN IP4 98.129.251.83
t=0 0
m=audio 22108 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
IP 10.0.2.4.5068 > 10.0.1.41.50600: UDP, length 1318
E..B.... at .k.
...
..)......i}SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on>
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at phone.myprovider.com>;tag=etQmF8eNZ7NKm
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 INVITE
Contact: <sip:98765432100 at 98.129.251.83:5060;transport=udp>
User-Agent: myprovider
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-myproviderOutboundGateway: sip:+98765432100 at 67.231.8.195
X-myproviderOutboundCarrierID: 29927192767592
X-myproviderCarrierRate: 0.00900
X-myproviderCloudRate: 0.00300
Remote-Party-ID: "Outbound Call" <sip:5060 at phone.myprovider.com
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1415023959 1415023960 IN IP4 68.34.22.11
s=FreeSWITCH
c=IN IP4 68.34.22.11
t=0 0
m=audio 30040 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:30041
IP 10.0.1.41.50600 > 10.0.2.4.5068: UDP, length 580
E..`-... at .3n
..)
........L..ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:50600;branch=z9hG4bK6081ac89;rport
Route: <sip:sip.myservice.info:5068
;nat=yes;ftag=as288f6857;lr=on>,<sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647
E....... at .n.
...6..........;ACK sip:98765432100 at 98.129.251.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600
Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f>
Max-Forwards: 70
From: "John Connor" <sip:skynet.device-3 at 10.0.1.41:50600>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>;tag=etQmF8eNZ7NKm
Contact: <sip:skynet.device-3 at 10.0.1.41:50600>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.5.0
Content-Length: 0
IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406
E....... at .o.
...6.........?xCANCEL sip:98765432100 at phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1
Max-Forwards: 70
From: "John Connor" <sip:12345678 at phone.myprovider.com>;tag=as288f6857
To: <sip:98765432100 at ints.myservice.info:5068>
Call-ID: 6717947d5e5f445314ba762f154be72f at 10.0.1.41:50600
CSeq: 102 CANCEL
Content-Length: 0
Reason: SIP;cause=200
2014-11-03 21:42 GMT+04:00 Daniel-Constantin Mierla <miconda at gmail.com>:
> Module: sip-router
> Branch: master
> Commit: 952227ef749da464c7990fa7f056764daf4bda0d
> URL:
> http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=952227ef749da464c7990fa7f056764daf4bda0d
>
> Author: Daniel-Constantin Mierla <miconda at gmail.com>
> Committer: Daniel-Constantin Mierla <miconda at gmail.com>
> Date: Mon Nov 3 18:41:01 2014 +0100
>
> uac: use transaction to propagate uac_auth() flag
>
> - needed by dialog to increase the cseq
> - local request is no longer accessible at forwarding callback
>
> ---
>
> modules/uac/auth.c | 6 ++++--
> 1 files changed, 4 insertions(+), 2 deletions(-)
>
> diff --git a/modules/uac/auth.c b/modules/uac/auth.c
> index 1717cc7..3bdee90 100644
> --- a/modules/uac/auth.c
> +++ b/modules/uac/auth.c
> @@ -469,8 +469,10 @@ int uac_auth( struct sip_msg *msg)
> goto error;
> }
>
> - /* mark msg wit uac auth for increase of cseq via dialog */
> - msg->msg_flags |= FL_UAC_AUTH;
> + /* mark request in T with uac auth for increase of cseq via dialog
> + * - this function is executed in failure route, msg_flags will be
> + * reset afterwards by tm fake env */
> + if(t->uas.request) t->uas.request->msg_flags |= FL_UAC_AUTH;
>
> return 0;
> error:
>
>
> _______________________________________________
> sr-dev mailing list
> sr-dev at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>
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