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.
Yesterday we talked about how Anthony added the ability to use mod_dingaling to make outbound calls using the new gmail voice interface. After taking a day off from doing ridiculously cool things with mod_dingaling, Anthony has now programmed the ability to receive calls in FreeSWITCH from a Google Voice account! This now means that you don't need to have a Gizmo number (or a landline) in order to receive incoming Google Voice phone calls into your FreeSWITCH system.
Our intrepid master hacker (Anthony Minessale) took it upon himself to see if he could make FreeSWITCH connect to gmail's new voice dialing interface. Someone suggested that it couldn't be done, so Tony went ahead and did it, and I think it took only about an hour for him to code the whole thing.
The FreeSWITCH wiki contains an article here that describes how it works. We've had users get multiple calls up and running. One even got 24 concurrent calls! We don't recommend abusing it like that, but just knowing that it can be done is pretty cool.
Chalk up another victory for the FreeSWITCH development paradigm. Wise engineering choices and disciplined coding practices continue to pay off with many new features being added with relative ease.
I thought you might appreciate this article at Network World. It talks about the potential that we've reached a tipping point in cloud-based VoIP. It specifically mentions Skype and OnSIP and that OnSIP is based on Asterisk, FreeSWITCH, and OpenSIPS.
Please go check it out and comment on the article. Give your feedback on using FreeSWITCH in the enterprise.
We are pleased to announce the continuation of
the FreeSWITCH Training courses! Another course is now available in
October in New York City. The course will cover the ins and outs of
FreeSWITCH, from setting it up to advanced routing topics and even some
coverage of building a FreeSWITCH C module. The training is being
organized by Darren Schreiber, co-author of the new Packt FreeSWITCH book.
Hello all! I felt it might be good to give the community a quick update on what's going on with the project...
ClueCon MMX was a huge success! We had a great time seeing everyone and we were privileged to have many great speakers and sponsors make some great announcements. Anthony Minessale took the wraps off of the magic sofia recover feature that has actually been lurking in the FreeSWITCH source tree since January of this year. We also had Mathieu Rene announced RTMP, a new Flash-based softphone application for FreeSWITCH.
It's official: The FreeSWITCH book has arrived!
The official title of the book is FreeSWITCH 1.0.6, however don't let the number fool you. The information applies to both FreeSWITCH 1.0 and 1.2 branches. Under the hood there may be some differences but the user interface has remained very stable. This book is perfect for anyone learning the upcoming FreeSWITCH 1.2 release.
ClueCon is only two weeks away!
This is a reminder that there are still a few rooms left at the Trump. When getting your room be sure to refer to booking number 4347 and ask for the discounted rate of $225 per night. This is a great value! ClueCon is the premier open source telephony developer conference in North America. Nowhere else can you enjoy such a great mix of technology and business professionals, a four-star hotel, and a price under $700. ClueCon has it all!
Congratulations to FreeSWITCH community member João Mesquita (IRC: jmesquita) who has been sponsored by Khomp to travel to Brasil and speak about FreeSWITCH at the FISL conference.
João will be making two presentations at the conference. If you are going to be near Porto Alegre next week then by all means check in and see how he does. For reference, here are his presentations, in Brazilian Portugese:
palestra: FreeSWITCH™ - Telefonia do futuro
sala: 41-A fisl 1
dia: 23 Jul 2010horário: 10:00duração: 60 minutos
We're about 1/4 of the way to getting to the target amount for the
UDRP dispute on freeswitch.com. We're still looking for your support to
get the domain pointing here as it should be....Remember to put UDRP and your name in the comment note so we can properly thank you!
Thanks to everyone who has contributed so far:
Jay Binks www.NetSip.com.au
Everyone go check out this post - it's a great interview with Anthony conducted by Suzanne Bowen of DIDX. Don't forget to check out the classic video of Anthony jamming on the harmonica with Matt Williams (IRC: hmmhesays) playing guitar.