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
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 place
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 this
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 saeedahmad1...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Sun, 26 Jul 2009 11:27:54
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 kr...@freeswitch.org wrote:
They don’t just 503 the call once you hit the limit?
And look at the options for using limit... Be a
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
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.
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 gmar...@celliax.orgwrote:
Performance problems and other issues (eg crashes on ALSA drivers) has
been reported for Skypiax on CentOS, albeit various users got good
success on
On Sun, Jul 26, 2009 at 2:19 PM, Muhammad
Shahzadshaherya...@googlemail.com 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
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 m...@giagnocavo.net wrote:
From: Michael
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
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
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
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