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 <ovoshl...@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..@.=. > ............_.aINVITE sip:89176270...@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:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com> > Contact:<sip:g...@sip.myservice.com:5068> > Call-ID: 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..@..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@17.6.43.24:50600>;tag=as5255aaa8 > To: <sip:89176270...@sip.myservice.com:5068> > Call-ID: 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:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com>;tag=as066163db > Call-ID: 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", > 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:89176270...@sip.myprovider.com SIP/2.0 > Via: SIP/2.0/UDP sip.myservice.com:5068 > ;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.0 > Max-Forwards: 70 > From: <sip:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com>;tag=as066163db > Call-ID: 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..@.<. > ...............INVITE sip:89176270...@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:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com> > Contact:<sip:g...@sip.myservice.com:5068> > Call-ID: 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", > nonce="7d150eae", uri="sip:89176270...@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..@.<. > ...............INVITE sip:89176270...@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:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com> > Contact:<sip:g...@sip.myservice.com:5068> > Call-ID: 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", > nonce="7d150eae", uri="sip:89176270...@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....@..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:g...@sip.myprovider.com>;tag=as5255aaa8 > To: <sip:89176270...@sip.myprovider.com>;tag=as2ce5c2f5 > Call-ID: 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", > nonce="5f11cf69" > Content-Length: 0 > > 2014-10-30 20:26 GMT+04:00 Yuriy Gorlichenko <ovoshl...@gmail.com>: > >> Thanks for answer. Now will insttall it for tests. >> >> 2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla <mico...@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....@gmail.com>: >>> >>>> No way. Use sems or b2b. >>>> 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" <ovoshl...@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@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@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-us...@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@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >
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