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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.
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FreeSWITCH 1.0.5pre8 Has Arrived!
Submitted by
mcollins on Tue, 12/08/2009 - 18:09
::
It's that time - everyone needs to update to the latest 1.0.5 pre-release. Those running SVN trunk please make current as soon as you reasonably can. If you're still running 1.0.4 please make your preparations for upgrading. 1.0.5pre8 is more stable than 1.0.4 so please make arrangements to update systems. We are starting to get a deluge of bug reports for 1.0.4 that have long since been fixed in the 1.0.5 pre-releases (and, naturally, in the latest SVN trunk).
The 1.0.5pre8 tarball can be downloaded here.
If you find bugs please report them to JIRA immediately! Also, be sure to follow up on your bug reports. We've had to close some bugs because the those reporting have not responded to requests for further information.
Thanks for being a great community! With your help we will have 1.0.5 out the door very soon.
-Michael
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