Welcome To FreeSWITCH

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch.  It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow.

We support various communication technologies such as Skype, SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16, 32 or 48 kilohertz and can bridge channels of different rates.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver.

a Spec Sheet is available on our Wiki.

FreeSWITCH 1.0.4pre9 Now Available

Submitted by mcollins on Mon, 06/29/2009 - 23:05
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The inexorable march to the official release of FreeSWITCH 1.0.4 continues. Version 1.0.4pre9 is now availble for immediate download. All those not using the latest SVN are encouraged to upgrade as soon as possible. The tarball is available here.
 
This latest pre-release is now the most stable version of FreeSWITCH available. There have been numerous improvements and bug fixes. The community has been providing a lot of feedback and the developers are doing a good job of addressing any issues that are found. There are new features as well. One in particular that you may appreciate is a new FS API command: show distinct_channels.
 

Bootable ISO's, Created Nightly!

Submitted by mcollins on Thu, 06/18/2009 - 23:57
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A special thanks goes out to Kristian Kielhofner of Astlinux fame - and Yahoo! message board infamy - for creating bootable ISO's with FreeSWITCH. Kristian has graciously added some automation to his setup and each night a new ISO is created. The link is here:
 
http://mirror.astlinux.org/freeswitch/daily/
 
Many thanks to all of the folks out there who are working so hard to make the FreeSWITCH community the best in the OSS world!
-MC

FreeSWITCH In The UK - How-To By IBM's Rob Smart

Submitted by mcollins on Wed, 06/10/2009 - 22:31
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FYI,
IBM's Rob Smart has been busy with FreeSWITCH! For those of you living in the U.K. I recommend this nice blog post explaining how to set up FreeSWITCH as your home PBX/voice communications server. Now if we can just get Rob and IBM to sponsor ClueCon... :)
-MC

OpenSim Developers Creating Amazon EC2 Image

Submitted by mcollins on Wed, 06/10/2009 - 22:25
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Some of you may be aware of the fact that the OpenSimulator project has some enterprising community members who've used FreeSWITCH as the voice engine. We just wanted to report that they are also getting ready to have a VoIP-enabled image on Amazon EC2. Check it out and spread the word.
-MC

Getting Ready to Welcome The Newest Member of the FreeSWITCH Family: Kaiden Anthony Chandler

Submitted by mcollins on Mon, 06/01/2009 - 18:02
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We are pleased to let FreeSWITCHers everwhere know that long-time FreeSWITCH contributor Raymond Chandler and his wife Samantha are expecting their fifth child next month. We all want to give young Kaiden Anthony Chandler a warm welcome to the FreeSWITCH family. To that end, we are happy to let you all know that Samantha is registered on WalMart's baby registry here. Raymond (IRC: intralanman) spends a lot of time and energy working on FreeSWITCH and his family is very supportive. Samantha (IRC: lanwifie) spends a fair amount of time in the channel when she's not being a very busy mom. Let's all show our appreciation to the Chandler family by getting something for Kaiden!

FreeSWITCH 1.0.4pre8 Now Available

Submitted by mcollins on Tue, 05/26/2009 - 15:20
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The FreeSWITCH team is pleased to announce the immediate availability of version 1.0.4pre8. This latest release is the most stable and secure version of FreeSWITCH to date. All are encouraged to update as soon as possible. The latest files are available at files.freeswitch.org.
 
The newest release of FreeSWITCH adds new features and improvements in stability, security, and performance. For example, support for ZRTP (www.zfone.com) was recently added. ZRTP is an opportunistic encryption protocol that allows for automatic encryption of RTP streams when both endpoints are ZRTP-enabled. Combining ZRTP with TLS, which encrypts signaling information, allows for complete end-to-end security in FreeSWITCH.
 

FreeSWITCHeR: A Ruby Event Socket Abstraction Library

Submitted by mcollins on Thu, 05/21/2009 - 16:54
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Ruby fans everywhere will be happy to know that, in the words of Bougyman, "Ruby Loves FreeSWITCH!" How so? Consider the FreeSWITCHeR project, an open source Ruby library that acts as an abstraction layer for the FreeSWITCH event socket.

FreeSWITCHeR is a FreeSWITCH event socket abstraction library built upon the Ruby event machine. It allows for connecting to - and controlling of - multiple FreeSWITCH instances from a single controller. The following is a sample "get digits" listener program that uses the FreeSWITCH outbound event socket paradigm:

FreeSWITCH Now Supports ZRTP!

Submitted by mcollins on Thu, 05/21/2009 - 16:08
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The development team is pleased to announce that preliminary support for ZRTP as been added to the Linux, Unix, and Mac OS X versions of FreeSWITCH. Windows support will be added shortly.
 
ZRTP is a key agreement protocol for establishing SRTP streams. ZRTP is not limited to a specific signaling protocol because the key exchange is done within the RTP stream, therefore ZRTP works with SIP and H.323. ZRTP is opportunistic: if ZRTP is available at both ends of a call then an SRTP connection is automatically negotiated, otherwise a standard RTP stream is used.
 

ZDNet Article Focuses on FreeSWITCH 1.0.4, Skype Integration

Submitted by mcollins on Tue, 05/19/2009 - 19:46
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If you haven't heard already there is a nice article about FreeSWITCH. Please follow this link to digg.com and digg the article. Then please go to the vote section of the article and click the thumbs up button to upvote this article.
Also, for those of you who have other link sharing sites we have links for those:

  • reddit.com - click through and upvote the "Open Source Softswitch" entry
  • mixx.com - click through and comments, kudos, etc.

Please add to your del.icio.us accounts if you have them.

HD Telephony Article Mentions FreeSWITCH

Submitted by mcollins on Wed, 05/06/2009 - 16:34
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FYI,

Some of you are familiar with ZDNet's Dave Greenfield. He has a new blog post discussing HD Telephony. Evidently HD Telephony is becoming a big buzzword. We need everyone in the FreeSWITCH community to spread the word that not only is FreeSWITCH capable of HD voice, it does it as well or better than anyone on the planet.  If you think that last statement is hyperbole then google around for HD telephony or HD VoIP and see what your options are. Look at the expensive companies - Cisco, Avaya, Nortel, etc. - and see what they offer for HD voice. Compare that to using FreeSWITCH with CELT, Speex WB, or Polycom's Siren codecs, not to mention Skype integration with mod_skypiax. For zero cost software you can get HD voice. Of course, you'll still need some hardware, but who wants to pay for hardware and software both, especially when the paid-for software can't even keep up with FreeSWITCH?

If you have used or are using HD voice with FreeSWITCH then please go put a nice comment on Dave's blog. Also, if you know someone going to the HD VoIP Summit in New York on May 21 then definitely put a bug in their ear about FreeSWITCH + HD codecs being a fantastically low barrier-to-entry for HD voice in the enterprise.

-Michael