[Freeswitch-users] Performance Tuning

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

Re: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch

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 [mailto:freeswitch-users-boun...@lists.

Re: [Freeswitch-users] upgrading to latest SVN

-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 se

Re: [Freeswitch-users] upgrading to latest SVN

h.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo wrote: > Getting error below..not sure whats wrong..which line n

[Freeswitch-users] upgrading to latest SVN

Getting error below..not sure whats wrong..which line number in what file does this refer to? [r...@ss_freeswitch log]# freeswitch 2009-11-19 20:15:44.725118 [INFO] switch_event.c:568 Activate Eventing Engine. 2009-11-19 20:15:44.727095 [DEBUG] switch_event.c:556 Create event dispatch thread 0 C

Re: [Freeswitch-users] Setting up Conference with Moderator

u have 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 hav

Re: [Freeswitch-users] Setting up Conference with Moderator

t; and asks for main conf pin (123456) once again. I would like the caller to be disconnected if they get into the Moderator menu and enter wrong Moderator PIN 3 times. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +1720239169

Re: [Freeswitch-users] Changing User-Agent String

gent String It needs 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.ca<mailto:mr...@avgs.ca> On 17-Nov-09, at 9:49 PM, Ujjval Karihaloo wrote: http:

[Freeswitch-users] Changing User-Agent String

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

Re: [Freeswitch-users] Simple Conference Setup issue

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 and

[Freeswitch-users] Simple Conference Setup issue

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: Help appreciated ___

Re: [Freeswitch-users] Setting up Conference with Moderator

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 AT&T 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 Ce

Re: [Freeswitch-users] Setting up Conference with Moderator

...@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

Re: [Freeswitch-users] Setting up Conference with Moderator

] Setting up Conference with Moderator On Tue, Nov 3, 2009 at 6:27 PM, Ujjval Karihaloo mailto:ujj...@simplesignal.com>> wrote: Was that sarcasm or you really mean it? No, he's serious. There are some issues that are quite endemic t

Re: [Freeswitch-users] Setting up Conference with Moderator

: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be AT&T /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 AT&T cell phone, I

Re: [Freeswitch-users] Setting up Conference with Moderator

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

Re: [Freeswitch-users] Setting up Conference with Moderator

ou pastebin your logs > 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 >> direct

Re: [Freeswitch-users] Setting up Conference with Moderator

Rob On Oct 29, 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-use

Re: [Freeswitch-users] Setting up Conference with Moderator

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 U

Re: [Freeswitch-users] Estimating Call Capacity

: 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 a

[Freeswitch-users] Estimating Call Capacity

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

Re: [Freeswitch-users] Setting up Conference with Moderator

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

[Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers

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 remove

Re: [Freeswitch-users] Setting up Conference with Moderator

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

Re: [Freeswitch-users] Setting up Conference with Moderator

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 To

Re: [Freeswitch-users] Core Dump question!

freeswi...@internal> version FreeSWITCH Version 1.0.4 (exported) freeswi...@internal> 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@01CA535E.2ACD2880

[Freeswitch-users] Setting up Conference with Moderator

Hi I have the Basic Conferencing working. Here is my Dial Plan. I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN.. action application="conference" data="conference.c...@wideband+flags{moderat

Re: [Freeswitch-users] Core Dump question!

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@01CA5359.87C

Re: [Freeswitch-users] Core Dump question!

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

Re: [Freeswitch-users] FS Registration Contact

Ujjval Karihaloo Sent: Thursday, October 22, 2009 2:00 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] FS Registration Contact 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

[Freeswitch-users] FS Registration Contact

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

Re: [Freeswitch-users] Connecting to FS CLI...just hangs..

2009 at 4:42 PM, Ujjval Karihaloo mailto: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|grep

[Freeswitch-users] Core Dump question!

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.

[Freeswitch-users] Connecting to FS CLI...just hangs..

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 3103

[Freeswitch-users] Asterisk vs Freeswitch

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 ca

[Freeswitch-users] Conferencing setup with FS

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 http://lists.freeswitch.org/mailman/listinfo/freeswitch

Re: [Freeswitch-users] New install

: [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

[Freeswitch-users] New install

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 n