[sr-dev] git:master: uac: use transaction to propagate uac_auth() flag

Yuriy Gorlichenko ovoshlook at gmail.com
Mon Nov 3 22:36:01 CET 2014


Daniel, I also tested patch with Russian service MULTIFON. Works fine.

2014-11-04 1:12 GMT+04:00 Yuriy Gorlichenko <ovoshlook at gmail.com>:

> 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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