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 1.0.5 Update
Many of you have been anticipating the release of FreeSWITCH version 1.0.5, just as we have. We wanted to let you all know that we're still working on 1.0.5 and we hope to release it soon. We had planned on releasing last Tuesday, however when we checked the JIRA tracker that morning there were quite a few bug reports. All of those reports are being addressed, however we could use some community assistance.
HOW YOU CAN HELP
The best way to help the developers to speed the process is to test the latest SVN trunk. Users who are already on SVN trunk are encouraged to run the make current process as soon as they reasonably can. Report back any issues you encounter. The other really useful thing that will help out is to review the open JIRA reports. This entails reviewing the JIRA report, reading the comments, and trying to reproduce the behavior reported. Some have opened JIRA reports but have not followed up on them. In some cases the developers have follow up questions on these open reports that have gone unanswered. If you have opened a JIRA please review it to make sure that the developers are not waiting for more information from you.
Please visit http://jira.freeswitch.org and review the bug reports that are outstanding. If you sign up for an account then you can also open new bug reports and comment on existing ones. The more people we have testing and verifying the reported the behavior, the sooner we will be able to release 1.0.5.
Thanks for all of your help!