[SR-Users] cSeq increasing
Daniel-Constantin Mierla
miconda at gmail.com
Fri Oct 31 00:19:41 CET 2014
Do you call dlg_manage() for the initial INVITE?
Cheers,
Daniel
On 30/10/14 23:25, Yuriy Gorlichenko wrote:
> 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
> <mailto: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
> <mailto:sip%3A89176270590 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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>
> Contact:<sip:gw2 at sip.myservice.com:5068
> <http://sip:gw2@sip.myservice.com:5068>>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <http://sip:webinar.device-200@17.6.43.24:50600>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myservice.com:5068
> <http://sip:89176270590@sip.myservice.com:5068>>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>;tag=as066163db
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <http://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
> <mailto:sip%3A89176270590 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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>;tag=as066163db
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <mailto:sip%3A89176270590 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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>
> Contact:<sip:gw2 at sip.myservice.com:5068
> <http://sip:gw2@sip.myservice.com:5068>>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <http://sip.myprovider.com>", nonce="7d150eae",
> uri="sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 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
> <mailto:sip%3A89176270590 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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>
> Contact:<sip:gw2 at sip.myservice.com:5068
> <http://sip:gw2@sip.myservice.com:5068>>
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <http://sip.myprovider.com>", nonce="7d150eae",
> uri="sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 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
> <mailto:sip%3Agw2 at sip.myprovider.com>>;tag=as5255aaa8
> To: <sip:89176270590 at sip.myprovider.com
> <mailto:sip%3A89176270590 at sip.myprovider.com>>;tag=as2ce5c2f5
> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c at 17.6.43.24:50600
> <http://1b74d0a5402e76fb249fe8dc427ce99c@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
> <http://sip.myprovider.com>", nonce="5f11cf69"
> Content-Length: 0
>
> 2014-10-30 20:26 GMT+04:00 Yuriy Gorlichenko <ovoshlook at gmail.com
> <mailto: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 <mailto: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 <mailto:eremina.net at gmail.com>>:
>>
>> No way. Use sems or b2b.
>>
>> 30.10.2014 19:59 пользователь "Yuriy Gorlichenko"
>> <ovoshlook at gmail.com <mailto: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
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>> <mailto:sr-users at lists.sip-router.org>
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>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) -
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>> <mailto:sr-users at lists.sip-router.org>
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>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/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
> <mailto:sr-users at lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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