Hello,
I just checked in a new mod_managed. It breaks backwards
compatibility, but adds scripting and reloading support.
I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make &
make install seemed to take care of everything.
Let me know if y
Anthony Minessale wrote:
> limit is for inbound calls
> you cannot call it after you already made the call.
> The correct approach would be to not make the call at all.
>
> you could maybe use the limit FSAPI interface with apiExecute to check
> if the limit was exceeded and
> then not bother to pl
Dear All,
Is it possible to apply different limits on a same IP by different
destinations.
Example:
IP: 1.2.3.4
Destination: Germany Mobile (491) => max-channels=10
Destination: Germany (49) => max-channel=20
Sometimes the supplier provides limited capacity on different destinations;
so in thi
They don¹t just 503 the call once you hit the limit?
And look at the options for using limit... Be a little creative and you can
do just want you want to do with some regex
From: Saeed Ahmad
Reply-To:
Date: Sun, 26 Jul 2009 11:27:54 +0200
To:
Subject: [Freeswitch-users] Mod_Limit - Limiting d
Can mod limit also be used to apply limit on outbound supplier ip? i was
unable to find a way to do that...
- Saeed
On Sun, Jul 26, 2009 at 11:47 AM, Ken Rice wrote:
> They don’t just 503 the call once you hit the limit?
>
> And look at the options for using limit... Be a little creative and you
Hi,
I am having random Linux Kernel crash problems while running FreeSWITCH as
Skype to/from SIP gateway on one of our production servers. This machine is
running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS
svn revision number 13754.
At time of Kernel crash i could find f
Performance problems and other issues (eg crashes on ALSA drivers) has
been reported for Skypiax on CentOS, albeit various users got good
success on same CentOS. The section down below, "Extreme" Performances
on Linux solves all problems for the user that got issues on CentOS.
http://wiki.freeswit
Ciao Muhammad,
I've got many problems with ALSA drivers, including various kind of crashes.
To make a looong story short, use the alsa_drivers version 1.0.20,
they have not yet crashed on me.
Also, if you want to test it, you can compile the customized snd-dummy
driver you find in the svn code,
Thanks. Let me try it and let you know the results.
Thank you.
On Sun, Jul 26, 2009 at 5:40 PM, Giovanni Maruzzelli wrote:
> Performance problems and other issues (eg crashes on ALSA drivers) has
> been reported for Skypiax on CentOS, albeit various users got good
> success on same CentOS. The
On Sun, Jul 26, 2009 at 2:19 PM, Muhammad
Shahzad wrote:
> Hi,
>
> I am having random Linux Kernel crash problems while running FreeSWITCH as
> Skype to/from SIP gateway on one of our production servers. This machine is
> running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS
Hi Michael,
Thank you for your job with mod_managed, I get lastest version with mod_managed
but the files PluginInterfaces.cs, PluginManager.cs and ScriptPluginManager.cs
were not downloaded.
Diego
--- On Sun, 7/26/09, Michael Giagnocavo wrote:
From: Michael Giagnocavo
Subject: Re: [Fre
Every event has a header 'Event-Date-Local', which has the local date
and time. If want to actively retrieve the date and time, you can send
this API request to the server: api strftime
Regards,
Raul
On Sat, 2009-07-25 at 09:20 -0500, Milena wrote:
> Hello everyone,
>
> I'm using the inbound ev
Ah, that’s embarrassing. I added them and tried building FreeSWITCH.Managed
from svn and it worked fine now. (I’ll kick off a new complete build in a
minute.)
-Michael
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego
Hi Guys,
I'm going to use OpenSER as SIP Platform (Registrar, Proxy, Presence) and
FreeSWITCH in my application. So I need to subscribe the FreeSWITCH for
presence information of users who involve in the application. After some
looking of Sofia.conf.xml , it seems there is not support to doing so,
Dear all,
I am learning how to implement a IVR in Freeswitch.In our organization we
are using Perl scripting language for doing this.So In freeswitch also I
need to use Perl.
So far I heard two methods for executing IVR.
One is in dial plan using perl application.( In perl I create IVR
Dear All,
I have tried to call a Asterisk extension from FreeSWITCH. I have done
the following configurations,
* I have enabled mod_iax module in
modules.conf.xml file.
* Next I have configure following extension in
dialplan.
mod_iax isn't loaded. I suggest using sip anyways.
Mike
On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:
Dear All,
I have tried to call a Asterisk extension from FreeSWITCH. I
have done the following configurations,
* I have enabled mod_iax module in
module
I have loaded mod_iax now that error didn't come.
But, When I call I have received following message in the console.
"[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause:
FACILITY_REJECTED"
What configuration I missed?
How to use sip to connect the Asterisk?
Please give sol
Hi,
I get core dump segmentation fault in freeswitch machine frequently. can
any one assist me what is error in the freeswitch. i have pasted the logs in
freeswitch pastebin.
This is the link http://pastebin.freeswitch.org/9851
--
Thanks with Regards,
N.Baskar
__
Hi FS Users,
I just want to try multiple gateways. It works actually like this...
But I test call like 513 at 222.333.444.555, it also calls the
second bridge 111.222.333.333.
It there any way to determine which prefix will call to a bridge
specified.
E.g.
fo
Hi FS Users,
I just want to try multiple gateways. It works actually like this...
But I test call like 513 at 222.333.444.555, it also calls the
second bridge 111.222.333.333.
It there any way to determine which prefix will call to a bridge
specified.
E.g.
fo
Hi FS Users,
I just want to try multiple gateways. It works actually like this...
But I test call like 513 at 222.333.444.555, it also calls the
second bridge 111.222.333.333.
It there any way to determine which prefix will call to a bridge
specified.
E.g.
fo
Hi FS Users,
I just want to try multiple gateways. It works actually like this...
But I test call like 513 at 222.333.444.555, it also calls the
second bridge 111.222.333.333.
It there any way to determine which prefix will call to a bridge
specified.
E.g.
fo
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