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 <[email protected] > <mailto:[email protected]>>: > > 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..@.=. > ............_.aINVITE sip:[email protected] > <mailto:sip%[email protected]> 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:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>> > Contact:<sip:[email protected]:5068 > <http://sip:[email protected]:5068>> > Call-ID: [email protected]:50600 > <http://[email protected]: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...([email protected] > ....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:[email protected]:50600 > <http://sip:[email protected]:50600>>;tag=as5255aaa8 > To: <sip:[email protected]:5068 > <http://sip:[email protected]:5068>> > Call-ID: [email protected]:50600 > <http://[email protected]: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:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as066163db > Call-ID: [email protected]:50600 > <http://[email protected]: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..@.@p > ............t..ACK sip:[email protected] > <mailto:sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP > > sip.myservice.com:5068;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.0 > Max-Forwards: 70 > From: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as066163db > Call-ID: [email protected]:50600 > <http://[email protected]:50600> > CSeq: 102 ACK > Content-Length: 0 > > > IP 10.0.1.12.5068 > 21.47.2.3.5060: UDP, length 1293 > E..)H5..@.<. > ...............INVITE sip:[email protected] > <mailto:sip%[email protected]> 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:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>> > Contact:<sip:[email protected]:5068 > <http://sip:[email protected]:5068>> > Call-ID: [email protected]:50600 > <http://[email protected]: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:[email protected] > <mailto:sip%[email protected]>", > 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..@.<. > ...............INVITE sip:[email protected] > <mailto:sip%[email protected]> 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:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>> > Contact:<sip:[email protected]:5068 > <http://sip:[email protected]:5068>> > Call-ID: [email protected]:50600 > <http://[email protected]: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:[email protected] > <mailto:sip%[email protected]>", > 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 > [email protected].... > ...........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:[email protected] > <mailto:sip%[email protected]>>;tag=as5255aaa8 > To: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as2ce5c2f5 > Call-ID: [email protected]:50600 > <http://[email protected]: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 <[email protected] > <mailto:[email protected]>>: > > Thanks for answer. Now will insttall it for tests. > > 2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla > <[email protected] <mailto:[email protected]>>: > > 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 >> <[email protected] <mailto:[email protected]>>: >> >> No way. Use sems or b2b. >> >> 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" >> <[email protected] <mailto:[email protected]>> >> 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 >> [email protected] >> <mailto:[email protected]> >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - >> sr-users mailing list >> [email protected] >> <mailto:[email protected]> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> [email protected] >> <mailto:[email protected]> >> 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 > [email protected] > <mailto:[email protected]> > 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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