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 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, 12, 16, 24, 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.

Patent Office Grants EFF Request for Reexamination of Dangerous VOIP Patent

Submitted by mcollins on Sat, 02/06/2010 - 16:38
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For those of you who may not know, the EFF (Electronic Frontier Foundation) has an endeavor called the Patent Busting Project. The goal is to get the USPTO (Patent and Trademark Office) to review bogus patents that do nothing more than stifle innovation and encourage patent trolls to sue legitimate businesses. The EFF has scored another victory, and this one affects many of us in the VoIP and telephony industries. Please check out this news release from the EFF. It's definitely good news for OSS telephony.
 

Report: VoIP to see 79% penetration in 3 years.

Submitted by mcollins on Thu, 02/04/2010 - 19:11
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Here's an interesting article over at Fierce VoIP: Within 3 years VoIP will have a 79% penetration rate in businesses. I'm assuming this is worldwide but the article doesn't say specificially. In any case, it's another good sign that FreeSWITCH (along with VoIP technologies, be they OSS or proprietary) will be used a lot in the coming months and years.
 
-Michael

Buy The FreeSWITCH Devs Dinner, Get Version 1.0.5!

Submitted by mcollins on Tue, 01/26/2010 - 23:19
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We are pleased to announce that the FreeSWITCH development team is going to meet in one place for the release of version 1.0.5. The official release will take place the week of February 8, 2010! To celebrate we invite and encourage the community to buy the development team dinner one night that week. To donate, simply click the PayPal button on the right side-bar of this page.
 
On behalf of the FreeSWITCH development team: thank you for supporting the project and thank you in advance for giving the guys a well-deserved dinner.
 
-Michael

Good Introduction to SIP Article at Ars Technica

Submitted by mcollins on Mon, 01/25/2010 - 19:11
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All those interested in learning more about VoIP and SIP should head over to Ars Technica and read An Introduction to the SIP Protocol, Part 1. It is a slightly technical introduction to how SIP works and is a good foundation for those who need to understand more about how SIP works with the rest of the network. I will post links to any followup articles that Ars puts up.
 
-Michael

Sangoma Expands Support For FreeSWITCH Project

Submitted by mcollins on Thu, 01/21/2010 - 18:55
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We'd like to extend our thanks to Sangoma, a Friend of FreeSWITCH, for continuing to support our favorite open source telephony project. There is a nice write-up over at TMCnet.com discussing Sangoma's future plans for support not only the FreeSWITCH project itself but also the FreeSWITCH community and the ClueCon open source telephony developer's conference.
 

FreeSWITCH and FreePBX v3 Blog Post

Submitted by mcollins on Tue, 01/19/2010 - 19:04
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This just came in a few days ago. It's a blog post over at chilling silence wherein the author describes his (rather pleasant) experiences dealing with FreeSWITCH and FreePBX v3. It's a nice read that we thought you might enjoy.
-Michael

VoIP Today Magazine Interview With Anthony Minessale

Submitted by mcollins on Fri, 01/15/2010 - 18:53
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Great news! VoIP Today has post a new interview with Anthony Minessale, primary author and lead developer of FreeSWITCH. Please give the interview a read and learn learn more about Anthony, FreeSWITCH, and ClueCon!
 

Ars Technica Article Discusses ENUM, Future of Telephone Numbers

Submitted by mcollins on Thu, 01/14/2010 - 21:34
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Our friends at Ars have put up a nice article about the history and future of telephone numbers. It centers on ENUM and discusses other related technologies like DNS and NAPTR. It's a great read that we recommend to all those wanting to know more about the VoIP landscape.

FreeSWITCH ESL Now Supports TCL

Submitted by mcollins on Thu, 01/14/2010 - 19:47
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The FreeSWITCH team would like to let everyone know that ESL - the FreeSWITCH Event Socket Library - now supports the TCL scripting language. It is available in the latest SVN as well as the build file at latest.freeswitch.org. Be sure to install the requisite TCL dev package for your platform.
 
Thanks to Mark Mann for writing the sample scripts.

FreeSWITCH Adds Broadsoft SCA Support

Submitted by mcollins on Tue, 01/12/2010 - 01:26
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The FreeSWITCH team is pleased to announce that we have added support for the Broadsoft SCA (Shared Call Appearance) method of sharing lines in a VoIP environment. The Broadsoft SCA method is known to work with Polycom, Snom, Cisco SPA (Small Business Pro 500 Series), and Aastra phones among others.
 
Shared Line Appearance and VoIP - Why So Much Drama?