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.
NOTE: The following is a guest blog entry posted on behalf of Dean Hansen of DTH Software.
The DTH VoIP Billing system has been designed for telephony companies providing class 4 or class 5 service, including multi-tenant hosted PBXs. We are a North American-based company with customers in over a dozen countries and we continue to expand our international presence.
As you may have heard, there is a cyber-squatter sitting on the freeswitch.com domain name. Repeated efforts to contact the person or entity who has this domain registration have all failed. We are now pursuing a dispute resolution with ICANN. Unfortunately this process is not inexpensive for an open source project. Ideally we need to raise $2600US in order to have our case heard by a three-person panel.
HOW YOU CAN HELP
FreeSWITCH now has T38 support.
The old mod_fax and many of the codecs in FreeSWITCH have now merged to one module called mod_spandsp which takes atvantage of all the DSP features found in the spandsp library including T.38 endpoint and gateway functionality.
Some squatter has had freeswitch.com for years and it's time we had it back.
We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute.
They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action)
Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause.
Everyone who donates will get their name up on our thank you page on our site.
Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form.
FYI, I thought you all might appreciate this post. It highlights the fact that a number of VoIP security-related companies are seeing an influx of cash and that VoIP security is getting lots of attention in the enterprise. It also contains a dozen links to various VoIP news items relating to security and VoIP hacking.
We are pleased to announce the first formal FreeSWITCH training course. 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 upcoming FreeSWITCH book.
Organized as a 3-day bootcamp, the course is a full 3-day intensive training providing in-depth coverage of FreeSWITCH installation, configuration, maintenance and programming. You will learn, step by step, how to configure, manage and program the FreeSWITCH telephony soft switch in great detail. Tips and tricks in debugging and optimizing FreeSWITCH are also covered.
Mike Jerris already posted this on the mailing list but for posterity's sake I am putting it here as well. Please fill out Digium's survey about OSS PBX software. We are a bit curious to see if the 5% reported last year is at all realistic. You can take the survey here.
Thank you for supporting FreeSWITCH and OSS telephony.
Here is an item some of you might find interesting. An intrepid user has created a nice script for installing FreeSWITCH and FreePBX V3 on a Debian Linode using StackScripts. Check out his blog post on the subject. If you use something like this for packaging and/or distribution please send us some feedback on how well it works for you.
Our friends at Sangoma have released a new piece of hardware, the D100, which does on-board transcoding of calls. Check out the writeups over at TMCNet and DigitalJournal.
If you plan on purchasing one of these for your system then please let us know how it works out. We would love to have some feedback on how well this card performs in real-world scenarios.