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.
FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.
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. The G.729 codec is also available under a commercial license.
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.
FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.
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 Weekly News and Notes
Happy short week to those of you in North America!
The weekly FreeSWITCH news and notes took a hiatus while I was out on a medical leave. I am happy to report that I am back to work and recovering nicely. Many thanks to those who sent their well-wishes and happy thoughts. We have a great community and I am glad to be a part of it!
On last week's conference call we covered some Linux/FreeSWITCH install and configuration tips. A special thanks to Ken Rice for giving us some practical information on many of the useful files and utility items that are available in the FreeSWITCH source tree and how to implement them, including FreeSWITCH init scripts, Anthony's .emacs file, and even a monit configuration example. I hope you found these items as useful as I did.
We recently released FreeSWITCH 1.2.4 and Ken Rice tells me that more updates are in the works. More information will be available on this week's conference call. This week I will be presenting a Wiki how-to: adding a channel variable page. This will be especially useful because it illustrates a number of Mediawiki concepts. Also, we have a lot of missing channel variables so if everyone picks one or two to add we'll be able to expand the wiki coverage.
Have a great week!