[SR-Users] cSeq increasing
Yuriy Gorlichenko
ovoshlook at gmail.com
Thu Oct 30 23:25:27 CET 2014
Does I need to use $dlg_var(cseq_diff) before UAC_AUTH()?
If yes - How. Documentation say only that this var stores Difference
between CSeq...
2014-10-31 1:58 GMT+04:00 Yuriy Gorlichenko <ovoshlook at gmail.com>:
> Daniel. I installed new Kamailio 4.2.
>
> I set dialog module params like this:
>
> modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "track_cseq_updates", 1)
>
> Call still unsuccessfull. CSeq still the same
>
> IP 10.0.1.12.5068 > 21.47.2.3.5060: UDP, length 1111
> E..sH3.. at .=.
> ............_.aINVITE sip:89176270590 at sip.myprovider.com SIP/2.0
> Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as5255aaa8;lr=on>
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.0
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
> Max-Forwards: 70
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>
> Contact:<sip:gw2 at sip.myservice.com:5068>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 12.6.1
> Date: Thu, 30 Oct 2014 21:50:46 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 1822659339 1822659339 IN IP4 2.10.4.20
> s=Asterisk PBX 12.6.1
> c=IN IP4 2.10.4.20
> t=0 0
> m=audio 30162 RTP/AVP 8 3 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> a=rtcp:30163
>
> IP 10.0.1.12.5068 > 17.6.43.24.50600: UDP, length 380
> E...(p.. at ..5
> ....J.I......:.SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;branch=z9hG4bK4203f70a;rport=50600;received=17.6.43.24
> From: <sip:webinar.device-200 at 17.6.43.24:50600>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myservice.com:5068>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> Server: MS Lync
> Content-Length: 0
>
>
>
>
> IP 21.47.2.3.5060 > 10.0.1.12.5068: UDP, length 671
> E...Q?..3.CB....
> ...........SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.0;received=2.10.4.20;rport=5068
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>;tag=as066163db
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> Server: FastTel SoftSwitch
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="sip.myprovider.com",
> nonce="7d150eae"
> Content-Length: 0
>
>
> IP 10.0.1.12.5068 > 21.47.2.3.5060: UDP, length 364
> E...H4.. at .@p
> ............t..ACK sip:89176270590 at sip.myprovider.com SIP/2.0
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.0
> Max-Forwards: 70
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>;tag=as066163db
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 ACK
> Content-Length: 0
>
>
> IP 10.0.1.12.5068 > 21.47.2.3.5060: UDP, length 1293
> E..)H5.. at .<.
> ...............INVITE sip:89176270590 at sip.myprovider.com SIP/2.0
> Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as5255aaa8;lr=on>
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.1
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
> Max-Forwards: 70
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>
> Contact:<sip:gw2 at sip.myservice.com:5068>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 12.6.1
> Date: Thu, 30 Oct 2014 21:50:46 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 314
> Authorization: Digest username="gw2", realm="sip.myprovider.com",
> nonce="7d150eae", uri="sip:89176270590 at sip.myprovider.com",
> response="68af82b65cbbcd29a27873c7288a246f", algorithm=MD5
>
> v=0
> o=root 1822659339 1822659339 IN IP4 2.10.4.20
> s=Asterisk PBX 12.6.1
> c=IN IP4 2.10.4.20
> t=0 0
> m=audio 30162 RTP/AVP 8 3 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> a=rtcp:30163
>
> IP 10.0.1.12.5068 > 21.47.2.3.5060: UDP, length 1293
> E..)H6.. at .<.
> ...............INVITE sip:89176270590 at sip.myprovider.com SIP/2.0
> Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as5255aaa8;lr=on>
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.2
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
> Max-Forwards: 70
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>
> Contact:<sip:gw2 at sip.myservice.com:5068>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 12.6.1
> Date: Thu, 30 Oct 2014 21:50:46 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 314
> Authorization: Digest username="gw2", realm="sip.myprovider.com",
> nonce="7d150eae", uri="sip:89176270590 at sip.myprovider.com",
> response="68af82b65cbbcd29a27873c7288a246f", algorithm=MD5
>
> v=0
> o=root 1822659339 1822659339 IN IP4 2.10.4.20
> s=Asterisk PBX 12.6.1
> c=IN IP4 2.10.4.20
> t=0 0
> m=audio 30162 RTP/AVP 8 3 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> a=rtcp:30163
>
>
>
> IP 21.47.2.3.5060 > 10.0.1.12.5068: UDP, length 671
> E...Q at ..3.CA....
> ...........SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP sip.myservice.com:5068
> ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.1;received=2.10.4.20;rport=5068
> Via: SIP/2.0/UDP 17.6.43.24:50600
> ;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
> From: <sip:gw2 at sip.myprovider.com>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com>;tag=as2ce5c2f5
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> CSeq: 102 INVITE
> Server: FastTel SoftSwitch
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="sip.myprovider.com",
> nonce="5f11cf69"
> Content-Length: 0
>
> 2014-10-30 20:26 GMT+04:00 Yuriy Gorlichenko <ovoshlook at gmail.com>:
>
>> Thanks for answer. Now will insttall it for tests.
>>
>> 2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla <miconda at gmail.com>:
>>
>>> This feature (increasing/decreasing cseq for calls authenticated to the
>>> next hop by kamailio) is available with 4.2.0, by using dialog and uac
>>> modules.
>>>
>>> See more details at:
>>> -
>>> http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html
>>>
>>> Let me know if works ok for you, as I did not test it yet extensively.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 30/10/14 16:11, Yuriy Gorlichenko wrote:
>>>
>>> As I understand UAC module can not be used at production as module
>>> foroutgoing calls from kamailio to provider with this limitations?
>>>
>>> 2014-10-30 18:24 GMT+04:00 Pavel Eremin <eremina.net at gmail.com>:
>>>
>>>> No way. Use sems or b2b.
>>>> 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" <ovoshlook at gmail.com>
>>>> написал:
>>>>
>>>>> Does it possible increase cSeq manually (for example remove and then
>>>>> append headers?) for UAC module when send INVITE messages with Auth, or
>>>>> kamailio have pseudovar for this header?
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>
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