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.
Our buddy Dan York has written a piece about Microsoft's assimilation acquisition of Skype. For better or for worse it is a done deal. I don't know about you but I must confess that I'm keeping my eye on this one. It could be great or it could be a train wreck. In either case I've got the popcorn ready...
We wanted to keep everyone informed of the schedule for upcoming FreeSWITCH training. Darren Schreiber, co-author of the FreeSWITCH book and CEO of the 2600hz project, is the lead instructor. There are beginner and advanced classes as well as a 5-day complete training. Visit VoIP.com for all the details. Also, don't forget that Darren and Company can also offer customized training suited to your specific business and technical needs. Contact them at firstname.lastname@example.org to obtain more information.
Thanks to Darren and all the great folks at 2600hz for helping to expand the reach of the FreeSWITCH community all over the world!
Those of you running Linux/Unix systems no doubt make use of the time zone database. Now it seems a foolish group of astrologers/software developers has actually sued the long-time maintainer of this database, among others. They claim that they have a copyright on the timezone data. All the selacious details are available in this Techdirt post. Keep your eyes open on this one, it should be interesting.
An interesting item just came across my desk. A post over at Boston Review talk's about Susan Landau's new book: Surveillance or Security?: The Risks Imposed by New Wiretapping Technologies. It is quite an interesting read. If you are at all interested in Internet security and privacy, especially with VoIP communications, I highly recommend reading this article.
We would like to let everyone know that the 2600hz team has more official FreeSWITCH training sessions here in the United States:
- Nov 7-9, 2011 in NYC
- Jan 18-20, 2011 in Austin, TX
Pricing, accommodations and other information is available on the VoIPKB website.
Last week Anthony Minessale added the new mod_sms into the FreeSWITCH tree. This module allows you to route SMS messages in a FreeSWITCH server. How does it work?
Mod_sms binds on the global event system and listens for MESSAGE events. When it catches a MESSAGE event it routes it to the new chatplan. The chatplan is to chat messages as the dialplan is to phone calls. The chatplan "routes" the message to the appropriate endpoint. If there is no match then the message is handled the way any other chat message would be, that is, it is simply a message directly from one client to another.
Special thanks to Seven Du who has started a nice wiki page for mod_sms. Please try out this new module and give us your feedback. We would love to hear ideas on how you are using this in a production environment and what problems you are solving with it.
Here is a nice blog post from SunTel Technologies. I thought you all might enjoy a nice read.