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?
Yes, you can. But this will break subsequent CSeq numbers in all requests within the dialog. Also, you will need to restore proper CSeq for replies to the INVITE for which the CSeq was altered.
-ovidiu On Oct 30, 2014 9:59 AM, "Yuriy Gorlichenko" ovoshlook@gmail.com wrote:
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
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?
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
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?
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
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 mailto:eremina.net@gmail.com>:
No way. Use sems or b2b. 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" <ovoshlook@gmail.com <mailto: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? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <mailto: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
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?
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-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
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
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?
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-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
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
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?
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-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
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
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@gmail.com mailto: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 <mailto:sip%3A89176270590@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 <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>> Contact:<sip:gw2@sip.myservice.com:5068 <http://sip:gw2@sip.myservice.com:5068>> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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..@..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 <http://sip:webinar.device-200@17.6.43.24:50600>>;tag=as5255aaa8 To: <sip:89176270590@sip.myservice.com:5068 <http://sip:89176270590@sip.myservice.com:5068>> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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@sip.myprovider.com <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>>;tag=as066163db Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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..@.@p ............t..ACK sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@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 <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>>;tag=as066163db Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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..@.<. ...............INVITE sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@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 <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>> Contact:<sip:gw2@sip.myservice.com:5068 <http://sip:gw2@sip.myservice.com:5068>> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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@sip.myprovider.com <mailto:sip%3A89176270590@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 <mailto:sip%3A89176270590@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 <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>> Contact:<sip:gw2@sip.myservice.com:5068 <http://sip:gw2@sip.myservice.com:5068>> Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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@sip.myprovider.com <mailto:sip%3A89176270590@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 <mailto:sip%3Agw2@sip.myprovider.com>>;tag=as5255aaa8 To: <sip:89176270590@sip.myprovider.com <mailto:sip%3A89176270590@sip.myprovider.com>>;tag=as2ce5c2f5 Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@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@gmail.com <mailto: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 <mailto: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.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@gmail.com <mailto:eremina.net@gmail.com>>: No way. Use sems or b2b. 30.10.2014 19:59 пользователь "Yuriy Gorlichenko" <ovoshlook@gmail.com <mailto: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? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <mailto: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 <mailto:sr-users@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@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Ovidiu Sas -
Pavel Eremin -
Yuriy Gorlichenko