Hello,
can you send me the logs (for trunk) with debug=3 ... I may be able to
catch it without testing it myself, which can take a bit longer.
Cheers,
Daniel
On 30/10/14 16:21, Kristian F. Høgh wrote:
> Hi,
>
> Tested on 4.0.6 (+contact header REFER patch) and trunk.
> When appending header on o
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 Go
Hi Juha
Correct as usual :).
It is odd this happened only after upgrading to 1.4.6. I am guessing
something got more strict about the formating. Also strange is that the
tables for lcr_gw have type INT for both those columns (uri_scheme and
transport). None the less changing my code to remove the
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 :
> Daniel. I installed new Kamailio 4.2.
>
> I set dialog module params like this:
>
> modparam("dialog", "d
Thanks for answer. Contact is Ok. It is just literal mistake at dump. This
error happens because CSeq not incremented by kamailio. Already talking
about this with Daniel at the another List.
2014-10-31 1:37 GMT+04:00 Gonzalo Gasca :
> What about the Contact header,
> Contact:
> Can you verify is
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
E..sH3..@.=.
_.aINVITE sip:891
What about the Contact header,
Contact:
Can you verify is a valid one.
On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko
wrote:
> Hello. I use kamailio for calling to porvider. My providr seccefuully
> registered from UAC module, but when I try to call through it? it back 401
> Unauthorised.
Hi Ricardo,
I have a similar setup working:
sipml5 -wss-> Kamailio -udp-> GW (FS)
I use Freeswitch with UDP and works fine, as you can see initial Invite
with SDP for Webrtc clients using sipMl5 is normally pretty big
(audio+video) and normally if you are proxying that message the remote end
shou
Can somebody please tell me what would be a verified combination of kamailio
and rtpproxy?I mean, for example, kamailio 4.1 and rtpproxy 1.x that can be
downloaded from:http://
I found at least 3 different places where one can download rtpproxy:
http://sourceforge.net/p/sippy/rtpproxy/ci/master
Thank you for the response. So i tried following example on link you sent but i
get the following error:
ERROR: [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No
module matching found
i also tried :
kamailio2gig:~# kamctl fifo bm_enable_timer test 1
500 command 'bm_enable_tim
Thanks for answer. Now will insttall it for tests.
2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla :
> 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:
> -
>
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 d
Hi,
Tested on 4.0.6 (+contact header REFER patch) and trunk.
When appending header on originating INVITE, the ACK from kamailio gets messed
up.
I generate call, using myself as proxy
kamctl fifo dlg_bridge sip:201@[domain] sip:202@[domain] sip:[kamailio-ip]:5060
In route[tm:local-request], i us
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 :
> No way. Use sems or b2b.
> 30.10.2014 19:59 пользователь "Yuriy Gorlichenko"
> написал:
>
>> Does it possible increase
No way. Use sems or b2b.
30.10.2014 19:59 пользователь "Yuriy Gorlichenko"
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?
>
>
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" wrote:
> Does it possible increase cSeq manually
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 l
Hello,
if you want to use kamailio as a load balancer with failure re-routing,
then see dispatcher module -- it has a sample config that does re-routing:
-
http://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher.ex.config
If you want to install kamailio load balancer as active
Hello,
the problem can be UDP fragmentation -- the gateway stack is not able to
handle UDP fragments. If the gateway supports tcp, then use this
transport layer.
Cheers,
Daniel
On 29/10/14 21:42, Ricardo Martinez wrote:
>
> Hello Daniel.
>
> I have printed the $mb in the kamailio debug and the $
Hello,
see benchmark module:
- http://kamailio.org/docs/modules/stable/modules/benchmark.html
Cheers,
Daniel
On 30/10/14 04:09, bob sacamano wrote:
> hi, I am trying to see how long kamailio takes to process messages I
> saw the commands:
> set_time_stamp
> diff_
Dear Nandini,
The easiest and recommended way is try our PRO solution:
https://www.sipwise.com/products/sppro/features/
which contains heartbeat and load balancing our of the box.
Also it has perfect support from our team and will safe a lot of your
time and money, as works perfectly out of the
This is my setup:
UAC -> firewall (1:1 nat) -> Kamailio:5060 - > asterisk:5080
. |
.sip providers
I'm running kamailio 4.1.x & ast 11 based on aspitos kamailio configuration
tutorial.
My problem is port number in via, contact and from headers is 5080 instead of
Hello Daniel.
I have printed the $mb in the kamailio debug and the $ml :
The SIP message in the client side has 2759 bytes.
This is what I get from the kamailio at the entrance leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG:
Dear All,
Greetings,
I want to implement kamailio server as failover and load balancing.
Is there any configuration needed in kamailio.cfg.
As I have searched in blogs, For load balancing, Virtual IP is added and
also heartbeat software is needed.
Please let me Know how to configure kamailio s
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