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.
I was reading this story over on Slashdot about how SPIT (spam over internet telephony) will be worse than SPAM. I find it hard to believe that will be the case. If you use software like FreeSWITCH to frontend all your phone calls it can give you the ability to knock out* the people you don't want to talk to in a heart beat.
What is your take on this?
* No you can't literally knock people out via telephone... yet!
I'll be going on vacation till June 21st so our production may lag just a teenie bit.
Don't worry, we should be releasing 1.0.1 while I'm gone and with the slow down it gives our community time to prepare for the upcomming ClueCon 2008 http://www.cluecon.com
I just wanted to take the time to defend our use of XML if I can. =D
Originally, FreeSWITCH used the same .ini format that Asterisk uses in all of it's config. Actually the interface to parse .ini still exists and a module writer is able to use it if he pleases. The reasoning for XMLizing what we chose to XMLize becomes more clear when you begin to scale the system. FreeSWITCH parses it's XML registry when it first starts and keeps it in memory. This is one big entity that can be navigated similar to a file system. There are top level major sections: configuration, dialplan, directory and phrases. All of the bits and pieces of these sections are exploded out onto the disk in the default arrangement so you can edit the portion of the document you need and it also allows you to insert small XML representations of a single entity such as a SIP UA, a user on the system, the configuration for what modules you want to load. All of these files will be concatenated in the end into 1 big XML document that the entire core and its modules can access with a common API that gives you the entities as a tangible object that can be extended without more code.
It's been almost 3 years since the idea was hashed to write a new telephony platform. Today we have reached the milestone of 1.0.0. Thank you to everyone who helped make this release a success!
Let today mark the rebirth of Open Source Telephony as Phoenix is born from the ashes of past failures!
It looks like Zaptel is changing it's name to DAHDI.
I wonder how that will change the landscape of the Zaptel interface going forward for legacy support of existing devices. We may need a sugar DAHDI to help us port the changes. As we all know, Zaptel is the grand DAHDI of all hardware telephony interfaces and it will be a strange new world without it.
This will most likely be the last release before the upcomming 1.0 release next Monday!
See the Download link above for details.
How does FreeSWITCH compare to Asterisk? Why did you start over with a new application? These are questions I’ve been hearing a lot lately so I decided to explain it for all of the telephony professionals and enthusiasts alike who are interested to know how the two applications compare and contrast to each other. I have a vast amount of experience with both applications with about 3 years of doing asterisk development under my belt and well, being the author of FreeSWITCH. First I will provide a little history and my experience with Asterisk, then I will try to explain the motivations and the different approach I took with FreeSWITCH.
Today PIKA Technologies announced that their T1/E1 and Analog TDM hardware is now compatible with FreeSWITCH. You can read more here!
Everyone get ready as we approach 1.0 the fun will begin!!!! So [[http://www.cluecon.com|click]] the [[http://www.cluecon.com|Cluecon '08]] link or the logo Logo over --([[http://www.cluecon.com|there]])>>>