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
We are glad to report that the FreeSWITCH team has tagged version 220.127.116.11. You can download the tarball here. Anyone using 1.2.5.x should update as soon as possible. We appreciate all those who have helped us with testing and tracking down some sneaky and pernicious little bugs.
On last week's conference call we spent some time talking about the XML parser and some of its pre-processor directives. We discussed specifically how you can use the "exec" command to execute a shell script in the middle of XML processing. We also discussed a few tricks on how to look at the source code when you need to learn about some FreeSWITCH functionality that otherwise is not documented. This week's conference call subject is still pending, so stay tuned!
One other item I'd like to mention is that we've had several reports of FreeSWITCH success stories. We will be providing more information about those in upcoming stories on our Web site. We've got people using FreeSWITCH in various situations as well as software developers who've
added support for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to grow and flourish! Thank you all for being a part of it.
Take care and have a great week!