Try setting your energy-level down, at 0 for instance. If it helps,
then increase until you find a happy medium.
On Dec 22, 2009, at 11:14 PM, Marc Orenberg wrote:
Hello. I've written an application using mod_conference which often
has two parties speaking at once and one party
I'm using FreeSWITCH in front of Asterisk without any issue.
Stick with the latest trunk. Can you set your loglevel to debug and
pastebin your log?
Here are some additional tips to help us help you :)
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Rob
On Mon, Dec 7, 2009 at 3:34 PM, Spencer
What about cron?
Create a cron entry like:
*/5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
But if you're just dumping global variables, you could easily retrieve them
directly from fs_cli without running an app and process the output however
you'd like:
LOL thats funny.
freeswitch, what is the meaning of life?
On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote:
freeswitch list wrote:
condition field=destination_number expression=^$
{caller_id_number}$
I knew this day would come. After the accumulation of all of the
knowledge from
Hey guys,
Having a problem with mod_local_stream.
I recently did a make current from 15334 to the latest trunk
(15630). After restarting, there now appears to be a memory leak. On
a test system (CentOS 5.4, 64-bit) with no calls or registrations,
Freeswitch gradually consumes all of the
that memory leak, could you test
mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/
~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/
mod_local_stream.c) with your current fs version to confirm this is
the cause please?
Mike
On Nov 23, 2009, at 4:53 PM, Rob
/
mod_local_stream.c) with your current fs version to confirm this is
the cause please?
Mike
On Nov 23, 2009, at 4:53 PM, Rob Forman wrote:
Hey guys,
Having a problem with mod_local_stream.
I recently did a make current from 15334 to the latest trunk
(15630). After restarting, there now appears
, there is no bin directory. I will keep
scratching away at it.
David
David V. Fansler
s/v Annabelle
dfans...@dv-fansler.com
www.dv-fansler.com
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Thursday, November 19
Hi Sam,
Take a look at mod_xml_curl. Pretty sure it'll do everything you're
looking for.
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Also, I would browse the modules and look for other nifty
functionality that already exists before setting out to write
something new.
Freeswitch was in. Is there a better
version of Linux to use?
thanks
David
David V. Fansler
s/v Annabelle
dfans...@dv-fansler.com
www.dv-fansler.com
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Wednesday
lol!
we have to play nice in the wiki but the mailing list is another story.
On Nov 18, 2009, at 3:20 PM, Brian West wrote:
Sounds like you need to take a baseball bat to their forehead.
/b
On Nov 18, 2009, at 3:13 PM, Nicolas Brenner wrote:
I had a voip provider which wouldn't accept
Hi David,
When using Apt, you would install packages with:
apt-get install package name
Or search for packages with
apt-cache search search term
If you're not root, you'll need to stick sudo in front of those
command. Honestly, you might want to find a better tutorial with
explicit
Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Thursday, November 05, 2009 7:52 AM
To: freeswitch-users@lists.freeswitch.org
I agree there is no such thing as unlimited. The three ways most SIP
providers will structure pricing is 1) per minute (ie $0.02/minute),
2) per channel (ie $15/month) or 3) unlimited with a channel limit
(ie $7/month for any amount of minutes but after two simultaneous
channels its ring
You can check the numbers of arguments passed with argc, and access
them via argv[0], argv[1], etc.
Its hinted at on the main Javascript wiki page, and also detailed in
the FAQ.
http://wiki.freeswitch.org/wiki/Javascript_FAQ
On Nov 8, 2009, at 10:34 AM, god.nirvana wrote:
hi all:
into the Conference App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Get a dedicated DSL. That'll work better than any sort of traffic
prioritization or shaping (I've tried).
Depending on your average channel use and codec, you could probably go
with the smallest package and be fine.
Rob
On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:
I am currently
No- make current won't work because it will try to do an svn
update, which won't be applicable since its not a checked out working
copy of the svn repository. He'll have to checkout the trunk with:
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
from,
The first caller isn't challenged for the pin (don't really know why--
maybe somebody else can elaborate on how the pin is designed to be
used). So to work around it, I validate the conference pin and
moderator pin independently via an IVR (or dynamically with a script
and odbc call) then
not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http
a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
odbc calls? What is it currently set to?
Rob
On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling card
solutoin. i found mod_odbc_query
Have you tried setting the effective_caller_id_number before
bridging? Such as:
action application=set data=effective_caller_id_number=9185551212/
Cheers,
Rob
On Oct 20, 2009, at 6:55 AM, Durk de Beer wrote:
Hello,
I us a call forward on freeswitch to forward calls to my mobile
phone. If
Try 300 seconds (5 minutes) and see if it improves.
On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:
2009/10/20 Rob Forman rob4manh...@gmail.com:
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
odbc calls? What is it currently set to?
Now 1 min
Rob
You could, but I would try just doubling whatever it is to see if
thats improves the issue first. The default is 32MB. You could
double it to 64MB and test again.
What is it currently set to (run: sysctl kernel.shmmax)? You can
change it on the fly with sysctl. Once you're done testing
Those are recommended values for Oracle. Freeswitch != Oracle. They
behave and use a server's resources very differently.
I wouldn't change more than you need for now. Tweak shmmax then go
from there.
Cheers,
Rob
On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:
2009/10/20 Rob
Hi Christian,
You can subscribe and monitor the heartbeat event either locally or
remotely:
http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example
You could also send an application level SIP packet, like a ping, to
Freeswitch externally. I have a small script if you can't find one.
I was in a production window so when the latest trunk worked I moved
on. I went back to troubleshoot later when I saw your email but I
couldn't reproduce it.
I like the tmpfs build- thanks for the tip Jason.
On Oct 9, 2009, at 1:11 AM, Jason White wrote:
Mark Campbell-Smith
I had that issue too where make current failed on mod_fax (under libs/
tiff). And yeah, it caused a problem where a bunch of modules
wouldn't load. You'll want to get it resolved before installing. I
ended up moving the existing source aside and re-checked out the
trunk, which compiled
Can you advise sip-providers offering t38?
Gafachi has T38 fax support.
On Oct 7, 2009, at 4:26 PM, Vladimir Elizarov wrote:
Kristian Kielhofner пишет:
Since no one else has responded I'll chime in with some general
advice.
It's troubling to see that your provider is using Asterisk to
And make sure verbose is set to true in ./conf/autoload_configs/
fax.conf.xml.
On Sep 16, 2009, at 11:50 AM, Steve Underwood wrote:
On 09/17/2009 12:08 AM, Travis Stutsman wrote:
In my attempts to receive a fax from a PSTN fax machine, the
transaction
fails with error code 13 Unexpected
Hi all,
I wrote a small 10-line python script wrapping txfax
(http://pastebin.freeswitch.org/10274
). Basically it originates a call with Session(), calls mod_fax txfax
application, then hangs up.
The weird thing is that this works fine the first I run it from
fs_cli. When I run a second
issue with a 10-line perl script,
starting FS without -hp option worked for me.
On Thu, Sep 10, 2009 at 7:28 PM, Rob Forman rob4manh...@gmail.com
wrote:
Hi all,
I wrote a small 10-line python script wrapping txfax
(http://pastebin.freeswitch.org/10274
). Basically it originates a call
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