after doing some testing with fs, i can see in the console, when
entering top, that fs uses 9.9% of the memory.
when i do some more calls, the used memory will raise - the memory
will not beeing released, till i do a restart of fs.
is this a normal behavior or do i have some problems?
thanks
This is normal behavior... FS allocates memory into pools and re-uses that
same memory over and over... It is quite normal to see memory usage increase
as usage of FS increases to a point where it levels off for that load... As
the loa decreases memory is not released but used for later when
Freeswitch can handle a large volume of call... I suggest you review your
configs to make sure you don't have any of the default or arbitrary other
limits in there... We routinely run 1500 concurrent calls on dual quad
core hardware at call rates far above what you tested at
Ken
From: Dennis
thanks for the good explaination and for making me feel better :-)
2008/12/23 Ken Rice kr...@suspicious.org:
This is normal behavior... FS allocates memory into pools and re-uses that
same memory over and over... It is quite normal to see memory usage increase
as usage of FS increases to a
Hi,
I am getting the following error when compiling latest Freeswitch with svn
Revision no - 10914 on Debian etch 64bit. Freeswitch version 1.0.1 is
building successfully.
Making all in .
gcc -I/opt/src/freeswitch/src/include
-I/opt/src/freeswitch/libs/libteletone/src -fPIC -Werror -g -ggdb -g
if i do the same test with the 9998, it does not to seem much better:
20.0(4000 ms)/1.000s 5061 68.25 s 1174 xx.xx.xx.xx:5060(UDP)
0 new calls during 1.008 s period 9 ms scheduler resolution
710 calls (limit 1000) Peak was 782 calls, after 58 s
0
There are a number of issues you can be running into... It really depends on
how your app works, what your actual configuration of freeswitch is, disk IO
subsystem, ulimits, etc etc
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec
because the latest result was with the 9998, it can't be out app (at
the moment).
so there are no other typical things or settings i could look for?
2008/12/23 Ken Rice kr...@suspicious.org:
There are a number of issues you can be running into... It really depends on
how your app works, what
Whats this 9998 to which you refer?
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec 2008 10:43:30 +0100
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
because the latest
the 9998 is an extension in the default.xml to test with media flowing
through the line.
2008/12/23 Ken Rice kr...@suspicious.org:
Whats this 9998 to which you refer?
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 23 Dec 2008 10:43:30 +0100
Oh! Well who knows how that will affect the performance... I have never
tested it with that... Try the echo tester but be sure you are using the
media refector with sipp or you arent doing anything useful
From: Dennis oderm...@googlemail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Thanks guys, it works.
Brian West wrote:
action application=export![CDATA[sip_h_Diversion=123456...@10.10.10.254
;reason=unconditional]]/action
/b
On Dec 22, 2008, at 9:55 AM, rod wrote:
Dear All,
I've been playing with the freeswitch options for one month now, and
I've been able
sorry, i do not really understand what you mean with: Try the echo
tester but be sure you are using the media refector with sipp or you
arent doing anything useful.
what is the echo tester and what is media refector and how could i use it?
i would like to find out, how many people can talk to
Today I did some more testing and packet sniffing.
When calling from google talk to google talk, packets are traveling only inside
lan, there are some
queries which goes outside, but nothing more.
When using Gtalk to call someone on sip, then those packets are sent outside
and I never see them
The echo tester was refering to the echo app in freeswitch
The media reflector is part of sipp that just echos media back to the
source... That's the proper way to test media handling capabilities
otherwise you are only seeing 1/2 the media stream
From: Dennis oderm...@googlemail.com
Hi,
I am running some stress testings on freeswitch. When the number of
RTP ports reached around 1248 - 1250, freeswitch starts to pop out No
RTP ports available! error:
2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
sofia_glue_tech_choose_port() No RTP ports available!
OS is Centos 5.2 64 bits
ah, that sounds interesting. so the echo app is the 9996, right?
how can i start/use the media reflector? is it something, i have to
call sipp with? sorry for this question, but i am very new in this
business.
right now i call sipp with: sipp -sn uac xx.xx.xx.xx -s 123456 -r 50
-l 400 -d 4000
Hi,
When I make a unsuccesfull call using session.originate, I'd like to
have a 10 minutes pause and then try again.
For our dialer we are using JS scripts, and setTimeout is not defined,
session.execute(sleep,...) doesn't work because the session has to be
originated first. And I don't really
I've decided to do this properly:
clean fresweetch reinstall.
My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP
10.99.8.221)
Other windows machine has gtalk with and also a sip phone registered as 1001
(IP 10.99.8.111).
First case - SIP to SIP. Calling from 1000
Please install c++ compiler.
/b
On Dec 23, 2008, at 3:10 AM, B Karthik wrote:
Hi,
I am getting the following error when compiling latest Freeswitch
with svn Revision no - 10914 on Debian etch 64bit. Freeswitch
version 1.0.1 is building successfully.
Making all in .
gcc
How exactly are you load testing this? Can you provide us an example?
/b
On Dec 23, 2008, at 4:39 AM, Juan Backson wrote:
Hi,
I am running some stress testings on freeswitch. When the number of
RTP ports reached around 1248 - 1250, freeswitch starts to pop out No
RTP ports available!
You don't have a default user in domain 192.168.14.10, in the default
config I used this so that you can set some vars on every call with
one call to set_user and it would set all the vars from the default
user on the current session. Its best to rip it out the set_user call
if you have
Thanks, commenting ext-rtp fixed my issue.
In case of further problems I'll do what you suggested.
Thank you again for all help.
On Tue, 23 Dec 2008 16:03:38 +0100, Anthony Minessale
anthony.miness...@gmail.com wrote:
when 2 devices talk via googles gtalk when they are both behind the same
On Tue, 2008-12-23 at 06:42 -0800, Michael S Collins wrote:
Hehe, you just stepped on a land mine! There was A LOT of discussion
ok. I promise that this my last post on this subject :)
about this. The simple fact of the matter is that there was no way to
make everyone happy so the devs
Carole,
Are you calling the hangup app from the Dialplan?
-MC
Sent from my iPhone
On Dec 23, 2008, at 7:04 AM, Brian West br...@freeswitch.org wrote:
Well in this context the phones need to hangup... they aren't going to
do so automatically. So you'll need to hang up on them or they will
Have you checked out 'sched_api'?
-MC
Sent from my iPhone
On Dec 23, 2008, at 4:25 AM, Alexandru Nedelcu a...@sinapticode.ro
wrote:
Hi,
When I make a unsuccesfull call using session.originate, I'd like to
have a 10 minutes pause and then try again.
For our dialer we are using JS
I think Carole is calling a group of people into a conference..
leaving and expecting everyone to get kicked.
/b
On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote:
Carole,
Are you calling the hangup app from the Dialplan?
-MC
Sent from my iPhone
Hi all, if you have the same problem as me failing to build the latest trunk
on OS X 10.5.6
please see the error message below:
making all in .
Compiling src/switch_xml.c ...
cc1: warnings being treated as errors
src/switch_xml.c: In function 'switch_xml_find_child_multi':
src/switch_xml.c:310:
Update and try again.
/b
On Dec 23, 2008, at 12:12 PM, Chris Chen wrote:
Hi all, if you have the same problem as me failing to build the
latest trunk on OS X 10.5.6
please see the error message below:
making all in .
Compiling src/switch_xml.c ...
cc1: warnings being treated as errors
On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
Besides, when all of the GUIs get built
you won't be hacking XML very much - if at all.
And the XML will still be there for those of us who prefer editing
configuration files to using GUIs. I'm very much a Unix shell type of
You don't have a default user in domain 192.168.14.10, in the default
config I used this so that you can set some vars on every call with
Thanks for pointing it out and explaining the purpose.
It looks like the domain is coming from set_domain in default.xml
which gets it from
You have to remember the default assumes a lot. You go to changing
things you have to then change the way things are assumed.
/b
On Dec 23, 2008, at 4:51 PM, John Wehle wrote:
You don't have a default user in domain 192.168.14.10, in the default
config I used this so that you can set some
On Tue, Dec 23, 2008 at 05:51:38PM -0500, John Wehle wrote:
It looks like the domain is coming from set_domain in default.xml
which gets it from sip_auth_realm. I guess the question is if
force-register-domain is being used then:
a) Should sip_auth_realm be set by FreeSWITCH to the
The default is just an example that tries to get you started... the
key thing you need to remember for ALL calls coming in that aren't
authenticated the domain_name variable needs to be set before you
transfer into the default context.
/b
On Dec 23, 2008, at 5:27 PM, Jason White wrote:
yeah, vim and emacs will always be available as your GUI. ;)
-MC
On Tue, Dec 23, 2008 at 2:31 PM, Jason White ja...@jasonjgw.net wrote:
On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
Besides, when all of the GUIs get built
you won't be hacking XML very much - if at all.
FYI,
If anyone would like to show their appreciation for the FS Dev team please
email me off list so we can talk about options. I think we can all agree
that Tony Co. have worked very hard to make FreeSWITCH a success for all
of us. Let's see if we can give a little love back to the core team!
Can this command be used to run a bash script?
I wanted to do some sox processing on some recordings after the bridge
ends and thought I should use this command. But would like to do it in
bash.
Is there a better way?
If this is the right way, what is the syntax for calling the bash
Frank @ Impact fr...@impactfax.com wrote:
Can this command be used to run a bash script?
Based on information at the wiki, this should be possible; use the system
command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system
___
I'm pretty sure that this is doable. Could you give us a hint as to
what arguments you want to send? For example, do you have one or more
channel variables you'd like to pass to the shell script?
-MC
Sent from my iPhone
On Dec 23, 2008, at 6:25 PM, Jason White ja...@jasonjgw.net wrote:
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