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.
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
Config works fine with mysql as topos backend. So the bug is restricted
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>
Thanks for your reply. Here's what has come up after a few hours:
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...
On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote:
> 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.
> *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George
> *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!
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.
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.
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-
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?
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
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.
@cfg_seti.core.max_while_loops = 200;
but got syntax error.
I'm new to Kamailio, so bear with me as I stumble through this. First, I'll describe what I'm trying to achieve at a high level and then perhaps somebody can advise me on whether Kamailio is a good fit for this solution or not. I'd like to be able to deploy a small appliance type server to our customer's sites that just runs Kamailio and a VPN connection back to our datacenter. At our datacenter, we run virtualized instances of Asterisk for each of our customers. The idea is that Kamailio would act as a transparent proxy through to the Asterisk instance under nominal conditions and as a basic SIP router in the case that the Asterisk instance is unavailable. This degraded functionality would then at least allow extension to extension calling even if the Internet or Asterisk instance is down.
I'm currently using dispatcher with a single entry in preparation for a time when we might want to failover to another Asterisk instance. I'm forwarding all REGISTER and INVITE messages to the server chosen from ds_select_dst. Initially this all seems to work as I can register with a softphone and pjsip show endpoints shows my softphone connected. However, when I attempt to call any extension (my own or another) Asterisk responds to the INVITE message with a "401 Unauthorized" message and the typical "The person at extension XXXX is unavailable...".
I know that more details might be necessary to troubleshoot this, but I didn't want to include everything in one post and risk cluttering it up with unnecessary information. If anyone can confirm that this is a reasonable way to approach the problem, I can then provide whatever relevant data is necessary to get deeper into it. (I've used sngrep, logging, asterisk cli, etc.)
Thanks in advance for any help.