Looking at Performance Tune my Freeswitch
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
Is refers to the following:
Turn off every module you don't need
Turn presence off in the profiles
libsofia only handles 1 thread per profile, so if that is your bottle neck use
Polycom Firmware matrix (Look at the polycom website) does not allow firmware
higher than 2.3.2 (I think) to be loaded on the old 501 phones...So first
confirm you are on a supported firmware release...
From: freeswitch-users-boun...@lists.freeswitch.org
the full
control you need.
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
Rob
On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
I have used the following setting in ivr.conf.xml to setup
conferencing with moderator.
However, the issue I have is - the user enters 123456
] upgrading to latest SVN
Ujjval Karihaloo ujj...@simplesignal.com wrote:
Getting error below..not sure whats wrong..which line number in what file
does this refer to?
freeswitch/log/freeswitch.xml.fsxml
This will be due to a syntax error somewhere in your configuration
-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Thursday, November 19, 2009 8:49 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] upgrading to latest SVN
I really didn't change anything.
I was running 1.0.4 and now built from SVN...I see
to go in the profile, not in sofia's global config.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.camailto:mr...@avgs.ca
On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote:
http://wiki.freeswitch.org/wiki
digits=1 param=conference
123...@default+flags{} /
/menu
/menus
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buser-agent-string.5D
As per the above link, we can change the User Agent String, but I added this
param name but does not seem to work.
[u...@freeswitch autoload_configs]$ vi sofia.conf.xml
configuration name=sofia.conf
I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch
and when I have 2 people dial in, looks like the Music on Hold never stops.
Here is what my public.xml looks like:
extension name=test !-- your provider or any name you'd like to call
it --
condition
10, 2009 3:50 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Simple Conference Setup issue
What does your config look like?
/b
On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote:
I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch
, at 9:54 AM, Ujjval Karihaloo wrote:
Yes, I think I did. However here is what furthur testing revelas. If
I dial in from ATT cell phone, I do not see any DTMF using Don's
IVR.xml.conf to call my conf app. But when I dial the same number
using a Verizon Cell, it works.
When I dial a number
...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator
: [Freeswitch-users] Setting up Conference with Moderator
you know I have heard this before... It seems to ONLY be ATT
/b
On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote:
Yes, I think I did. However here is what furthur testing revelas. If
I dial in from ATT cell phone, I do not see any DTMF
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the Conference App, I see DTMF from the provider
Rob:
Once I have the Moderator and Participants logged on, how do I invoke the
moderator previlidges, LIk esay muting everyone/someone or kicking someone out
of the Conf and the like?
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88
, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when
using the below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when using the
below configurationso it basically timesout.
UK
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the
With the following spec for CPU and Memory can someone help me guesstimating
how many simultaneous calls and Calls/sec a FS server can handle - Used as a
Conferencing Server.
cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Estimating Call Capacity
You'll have to do your own load testing. Nobody can really tell you
exactly how many you'll get.
/b
On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:
With the following spec for CPU and Memory can
One more try for help for COnferencing using moderator,
Anyone successfully accomplished this? Help appreciated!
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo
[ujj
Hi,
I used the downlaoded TAR ball and my calls worked, however, when upgrading to
the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call
- using Conferencing app)..
Here are teh bad and good 200 OKI see a lot of additional headers startin
gwith X:FS , can I
Any ideas on this one. Look slike only way rite now is to have a different Dest
Phone number for a moderator.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Thursday, October 22, 2009 9:02 PM
It just hangsand I CTRL-C out of it.
[r...@ss]# ./fs_cli -H 127.0.0.1
^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]
Freeswitch process is running:
[r...@ss bin]# ps -ef|grep free
root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch
root 8952
freeswi...@ss_freeswitch sofia_gateway_data
Segmentation fault (core dumped)
Just ran the gateway command above w/o any parameters,,, and it core dumped..
I am sure mistakes like that happen...but I not sure if it should core dump
and shutdown.
at 4:42 PM, Ujjval Karihaloo
ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote:
It just hangsand I CTRL-C out of it.
[r...@ss]# ./fs_cli -H 127.0.0.1
^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]
Freeswitch process is running:
[r...@ss bin]# ps -ef
Hi All,
I have FS registered to an ITSP. The contact is showing as follows..
Contact: sip:gw+i...@1.1.1.1:5080;transport=udp
Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1
I want the Phone number (FromUser)to show in the contact header in the REGISTER
msg going to the
params to send in the contact--
param name=extension-in-contact value=true/
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Thursday, October 22, 2009 2:00 PM
To: freeswitch-users@lists.freeswitch.org
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it
happened.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
How do I tell if it's the latest...I downloaded is yesterday..and installed it
from freeswitch.org
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA5359.87CB41C0]http
conference.conf.xml file in the auto_loads
directory.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA535A.E43E9870]http://www.simplesignal.com/
inline: image001
Is there benchmark test results on how many simultaneous calls Freeswtich can
do (with RTP anchored through it) vs the Asterisk.
For any hardware/CPU/Mem that anyone may have performed this performance
testing.
Any numbers on average how much Freeswitch scores over the Asterisk in terms of
Hi ,
I cannot seem to find a Document online for setting up conferencingon
FreeSwitch. Can someone point me to one?
Thx.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Hi,
I just installed freeswitch as a replacement for our Asterisk Server. I want
to untimately do Conferencing with it as I have heard is it pretty good at it.
I have it compiled and up and running. However, when I provision a
Sofphone/Xlite to register with it to run basic tests, it does
: [Freeswitch-users] New install
make sure your firewall is not up
/b
On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote:
Hi,
I just installed freeswitch as a replacement for our Asterisk Server. I want
to untimately do Conferencing with it as I have heard is it pretty good at it.
I
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