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>:
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
............_.aINVITE sip:89176270590@sip.myprovider.com SIP/2.0
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
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
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
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
............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
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
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
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
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
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
...........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
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.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>:

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
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_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
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-- 
Daniel-Constantin Mierla
http://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-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda