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@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:89176270590@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@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com> Contact:<sip:gw2@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:89176270590@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:gw2@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@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:89176270590@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@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@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:89176270590@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@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com> Contact:<sip:gw2@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:89176270590@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:89176270590@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@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com> Contact:<sip:gw2@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:89176270590@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@..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@sip.myprovider.com>;tag=as5255aaa8 To: <sip:89176270590@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 <ovoshlook@gmail.com>:
Thanks for answer. Now will insttall it for tests.
2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla <miconda@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....
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@gmail.com>:
No way. Use sems or b2b. 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" <ovoshlook@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?
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_______________________________________________ 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-users@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
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