sip:888@conference.freeswitch.org
https://conference.freeswitch.org

Welcome To FreeSWITCH

The World's First Cross-Platform Scalable FREE Multi-Protocol Soft Switch


FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.  It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools. More

PIKA T1/E1 and analog boards now compatible with FreeSWITCH

Submitted by admin on Mon, 04/14/2008 - 21:24
::

Today PIKA Technologies announced that their T1/E1 and Analog TDM hardware is now compatible with FreeSWITCH. You can read more here!

Thanks,

Brian West

Hey don't forget to meet us at Cluecon '08

Submitted by admin on Thu, 04/10/2008 - 02:36
::

Everyone get ready as we approach 1.0 the fun will begin!!!! So [[http://www.cluecon.com|click]] the [[http://www.cluecon.com|Cluecon '08]] link or the logo Logo over --([[http://www.cluecon.com|there]])>>>

ZDNet reports on "An Open Source Class-Five Switch"

Submitted by admin on Wed, 04/09/2008 - 00:59
::

Dave Greenfield of ZDNet blogs about freeswitch. Check it out and digg it:

http://digg.com/software/An_Open_Source_Class_Five_Telephony_Switch/

 

Its no joke

Submitted by admin on Wed, 04/02/2008 - 20:41
::

Now that April Fools Day has come to an end across the globe I just wanted to point out that the quote from yesterday is not a fabrication.

Its a real account of how one of our community members uses FreeSWITCH!

 

Never fear RC2 is here!

Submitted by admin on Tue, 04/01/2008 - 19:23
::

As we approach our 1.0 release you can expect FreeSWITCH to only get better moving forward. We have a quote from someone in our user base:

"Routinely my freeswitch routing servers reach > 400 sessions Per Second (where they are rated limited) and > 4000 concurrent Sessions (where they are also limited) with approx 20% CPU utilization and 30% ram utilization on a Dell 1950 w/ Dual Xeon E5335 2Ghz Quad Core CPUs and 4G of ram" Any SPS and Concurrency limiting is done not to protect my boxes but upstream peers I have had atleast 3 occations where Large Tier 1 / Tier 2 Intl and Domestic US LD Carriers have asked for mercy one called saying "Can you please slow the traffic up, you are melting down my {CENSORED} SBCs" well the vendors in question that I know for a Fact are Veraz and AcmePacket."

Please download, test and provide feedback.

For more info please check us out on IRC: #freeswitch on irc.freenode.net

You can download RC2 here!

 

RC1 Has arrived!

Submitted by admin on Thu, 03/20/2008 - 23:17
::

FreeSWITCH Community,

After nearly 2 years of non-stop development, FreeSWITCH has evolved from an idea and an empty directory into a fully operational soft-switch capable of withstanding the loads 8-10 times as large as anything else they were able to deploy in the past with other software according to the reports from our early adopters.

I am happy to announce the immediate availability of FreeSWITCH 1.0 “Phoenix“ (Release Candidate 1). The Phoenix symbolizes the rebirth of a new era and I hope this new release will be the first step in a new age of voice-over-IP development we can all benefit from.

We anticipate the official FreeSWITCH 1.0 “Phoenix” release date to be Monday, May 26th 2008. We will have a formal release presentation at this year’s ClueCon 2008 August 5th-7th 2008. http://www.cluecon.com

I would like to thank our community for the testing, bug reporting, code contributions, feedback and financial donations that helped us get this far, especially those who were involved from day 1.

Please take this opportunity to test drive FreeSWITCH as we will be making it available as binary packages for the first time to practice for the official 1.0 release. The code and arrangement of the configuration will not change apart from bug fixes and tweaks from this point forward. As always, the code will still be available in svn at the usual place.

For download instructions please stay tuned to:

http://wiki.freeswitch.org/wiki/Installation_Guide

The packages will be online soon.

Thanks Everyone!

Anthony Minessale II

New FreeSWITCH compat analog TDM cards

Submitted by admin on Sun, 02/03/2008 - 02:58
::

Check out NXTVOX http://www.nxtvox.com/ for an inexpensive

way to try out our new OpenZAP channel driver for FreeSWITCH.

The cards are compatable with both asterisk and FreeSWITCH! 

Secure VoIP

Submitted by admin on Sun, 02/03/2008 - 02:55
::

FreeSWITCH announces fully tested support for TLS sip signalling and SDES secure RTP making it possible to make secure VoIP calls with the standard release of the software! No patches or workarounds are necessary!

Now you can communicate with someone over the internet using FreeSWITCH and have a completely secure channel of communication. Used in conjunction with the G722 or speex16 HD codecs you can have crystal clear audio as well!

 

 

 

 

 

 

HD-Telephony: HIP or Hype?

Submitted by admin on Wed, 01/16/2008 - 20:03
::

Since the advent of the High Definition Television, The HD phenomena has spread far and wide in recent years making the “ordinary” “extraordinary” with the addition of a simple 2 letter prefix. The telecommunications industry is no exception with the strengthening concept of HD-Telephony. Is it all hype or is there an actual benefit to better sounding phone calls? How does this affect the hardware and software designed to keep us connected? As a software developer in this field, I hope to shed a little light on the topic and separate the facts from the misconceptions.

Release Candidate 1 on the way!

Submitted by admin on Tue, 11/06/2007 - 01:28
::

A few new features have been developed in the last few months.

We added:

* SRTP support: Make calls with secure RTP 

* mod_spidermonkey_curl: Now you can get and post http data from javascript.

* mod_dialplan_asterisk: A dialplan module similar to the asterisk dialplan syntax.

* mod_cdr_csv: A templated CSV CDR module.

* mod_tone_stream: Play tones anywhere a file can be played.

* USER channel, a way to call a user in the directory as a channel. 

* mod_voicemail: a full featured enterprise scale voicemail application tied to the core directory that can be used with a local sqlite database or shared across multiple boxes with ODBC. It supports mutiple greetings, fully customizable keys for every function, fully customizable email template, ability to forward messages to your email once you have listened to them, urgent messages with accurate urgent/non urgant message counts in mwi and templatable ivr prompts via the FreeSWITCH phrase macro interface.

*mod_limit: a small application to allow restriction of certian simotaneous resources in the dialplan which also works locally with sqlite and at an enterprise scale with ODBC.