Hello,
is there an rsync endpoint available or is there a possiblity of setting
this up? Creating a mirror via HTTP is a rather quick and dirty solution
and while the deb-repo can be mirrored using debmirror, the rpm repo is
hard to sync to a non-CentOS-based machine due to missing dependencies
such as yum and reposync in latest Debian-based systems.
Would be great to get some input in regards to this topic.
Cheers
On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote:
> Sure. Attached. Problem appears to be that the topos query can't find
> callid-totag (from the response).
>
> I'll try the same scenario with the mysql backend to see if it behaves
> different.
Config works fine with mysql as topos backend. So the bug is restricted
to topos-redis.
Forwarding my reply to the list, using gmail's reply button set Henning as
the sole recipient :-\
---------- Forwarded message ---------
From: George Diamantopoulos <georgediam(a)gmail.com>
Date: Sat, 26 Jun 2021 at 02:25
Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)?
To: Henning Westerholt <hw(a)skalatan.de>
Hello Henning,
Thanks for your reply. Here's what has come up after a few hours:
shm55: https://pastebin.com/h9JCePmc
shm54: https://pastebin.com/Nx5xEEnA
It seems to me htable is the culprit? Are you seeing anything different? 54
has been running for 77020 seconds, 55 for 28521 (significantly less).
I'm going to turn it off until we figure something out...
BR,
George
On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote:
> Hello,
>
>
>
> Good observation. Please run the memory statistics CLI commands to get
> more hints about the module that might cause it (as per below link). Then
> please report more details. If you can point to a particular module, you
> can also open an issue on our tracker.
>
>
>
> https://www.kamailio.org/wiki/tutorials/troubleshooting/memory
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George
> Diamantopoulos
> *Sent:* Friday, June 25, 2021 4:53 PM
> *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
> *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)?
>
>
>
> Hello all,
>
>
>
> I'm still investigating the (most likely non-kamailio-related) memory leak
> of my previous message to the list, there have been no developments so far.
> I'll update if anything changes.
>
>
>
> This concerns a new finding which seems to affect kamailio 5.5.x. I have
> two kamailio instances receiving the same traffic via round-robin. I
> upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared
> of any issues arising. I was lucky to do so, because with identical
> configuration, 5.5.x seems to run out of SHM very quickly. Here are links
> to graphs produced by our monitoring system:
>
>
>
> Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png
>
> New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png
>
>
>
> The configuration uses mtree, htable, vars and vns extensively. Has anyone
> come across anything similar? Let me know if I can provide any further
> information to help disect this. Thanks!
>
>
>
> BR,
>
> George
>
Hi to all,
Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values.
Regards,Hossein
Hi.
I'm looking for a solution to the following problem, and wondering whether
Kamailio (with which I have very little experience so far) may be it, or
whether I'm thinking in totally the wrong direction.
The problem:
I have a SIP client application which runs on a standard Linux system, and is
extremely basic - it can REGISTER in order to receive inbound calls, and it
can send INVITEs in order to place outbound calls, and that's about it. It
handles media as well - in essence, it's a simple softphone. I can't change
it for a more capable one, before anyone suggests that :)
Specifically, this thing cannot send REINVITEs in order to put calls on hold,
nor can it handle anything to do with transfers (blind or attended).
I'm looking for something which does have these SIP capabilities which I could
put in between this application and the SIP server through which it is placing
and receiving calls, so that I can (through some sort of API) cause calls in
progress to be put on hold, resumed, blind transferred, or hold-and-attended-
transferred.
Does that sound like something Kamailio can do?
If so, can anyone point me in a helpful direction re how to go about it?
If not, can anyone suggest something else which could sit in between a very
basic SIP client and a PBX server, in order to inject these sorts of commands
into the path and thereby give me these sorts of extra call capabilities?
Thanks,
Antony.
--
I'm not impossible, just highly implausible.
Please reply to the list;
please *don't* CC me.
It is possible to update value of max_while_loops dynamically in config
file?
For example, there could be a need to increase the max in some while
loop, but after the loop is done, decrease it again for other loops.
There exists an rpc command to do that, but I haven't found how to do it
in config file.
I tried
@cfg_seti.core.max_while_loops = 200;
but got syntax error.
-- Juha
Hello guys,
I know SEMS can provide conference, voicemail, and other services.
In theory it’s also a B2BUA. Could I use a python script to provide a
simple routing service? I.e.: receive an invite and send it somewhere else
based on some routing logic?
Thanks all
David
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Hello guys,
I'm trying to dynamically add a branch after doing lookup.
The user is found, but in some cases I need to add a branch and do parallel
forking.
So i'm basically doing:
route[LOCATION] {
if (!lookup("location")) {
....
}
if (something) {
route(BRANCH_TO_EXTRA);
}
route(RELAY);
}
route[BRANCH_TO_EXTRA] {
$fs = MY_SOCKET;
append_branch("sip:$tU@" + $sel(cfg_get.pstn.gw_ip) + ":" +
$sel(cfg_get.pstn.gw_port));
return;
}
For some reason only the branch appended is being used (I have
append_branches=1)
Ideas?
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Hello,
following situation.
I have a Kamailio (5.4) using rtpengine to loadbalance calls.
If a call from Alice comes in, Kamailio decides to send the call to Carrier
B from Bob.
Bobs Phone is ringing and the carrier B send a 183 Session Progress with
SDP and To-tag=abcd. The SDP has G722 as codec and port 1234.
A few moments later carrier B send a second 183 Session Progress with SDP
and TO-tag=fghi. The SDP has G711 as codec and port 5678. This is done, to
play some funky music as ringtone -.-
If Bob answers the call, carrier B sends a 200 OK WITHOUT SDP and
TO-tag=abcd. So this should instruct our Kamailio to switch to the first
G722 and port 1234.
But sadly, this is just not working as expected.
We tried to set the flags media-handover and port-latching for the
rtpengine options and additionally set a to-tag when using rtpenging_manage.
But this doesn't solve the codec change, so we have only audio when Bob
answers the call, but no ringtone-music. If we allow G711 only in the
outgoing INVITE to Bob, we have also tha ringtone-muisic, because there is
no codec-change.
Carrier B tells us, they are using a fork-mechanism.
Is there something we can do, to support the codec change in 183? Or
enforce carrier B to send SDP in 200 OK? Or anything else?
Carrier B can not change anything in the ringtone-music-backend. They are
stuck on G711.
Thanks!
