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.
Good news! I was looking for some information on a ClueCon 2010 talk and it turns out that I had not uploaded any of the day 3 videos! They are on viddler.com, starting with video 94. These talks are all now available:
Notes on the Perception of Imaginary Differences
The blue.box Project
IAX - The Myth, The Reality - And The Status
Telefaks*de Application Server and FreeSWITCH Operator Panel
Creating Phone 2.0 Applications with Adhearsion
What I Learned Supporting Asterisk and AGI Applications For a Year
As you know, ClueCon is fast approaching: August 9 - 11, 2011. That's only three months away! The ClueCon team wants to see as many as possible get registered ASAP, so as an added bonus they are doing an early bird special: extra entries into the drawings for the coolest laptop and iPads ever!
Here's how it works: Get registered before the end of May 2011 and you'll receive a total of four entries into the giveaway drawings. After May 31 the number of entries will decrease:
Register by June 7 and receive three entries
Register by June 14 and receive two entries
All registrations after June 14 will receive a single entry.
Don't wait to sign up for ClueCon - a number of folks have already signed up and have their four entries guaranteed. Get signed up right away so that you maximize your chances of winning the coolest laptop and iPads!
For those of you who were not able to attend last week's conference call (April 27, 2011) I would like to let you know that both audio and video are now available:
* Audio - FreeSWITCH Weekly Conference Main Page
* Video - Stream it at Safi Systems Website or download it here
Thanks again to Zac Wolfe and the gang over at Safi Systems for their hard work. The product is actually very nice and is a great addition to the burgeoning FreeSWITCH ecosystem. If you haven't already seen it then please visit Safi Systems to learn more.
You may recall that Twilio launched a "Roll-Your-Own Google Voice" campaign a while back. They took the open-source Asterisk PBX and merged it with their proprietary cloud platform to create a powerful cloud-based communications software platform with handy APIs and a slick interface. This generated a fair amount of buzz, however the costs involved (up to $0.06/minute or more) were a barrier to entry for some. Additionally, the service doesn't offer the ability to use SIP directly. Voxeo also offers a full-featured, cloud-based telecom platform with its Tropo service.
Tropo and Twilio are great services indeed. But wouldn't it be nice if there was a 100% FOSS offering that made use of FreeSWITCH? As of today there is! The gang over at the 2600hz project have released an offering they've dubbed "Whistle." It is a truly open cloud-based platform that has many features, among which are native SIP support, service configuration, and much more. Whistle leverages the power of open source: FreeSWITCH, CouchDB, RabbitMQ, Erlang. The result is a platform that can run in the cloud or on your own servers - or both. It allows you to tap into FreeSWITCH's powerful event-driven subsystems across multiple deployments of FreeSWITCH and connect those events to API-driven software, all while distributing components and servers anywhere across the Web.
The 2600hz team has evidently been busy. Not only have they produced this software, they have full API documentation on the Whistle system as well as a RESTful abstraction layer called Crossbar. They also have an automated installer that you can use if you wish to avoid the complicated manual install process. They've even built a new website with udpated blue.box documentation. All of this is released under the same OSI-approved license (MPL 1.1) that FreeSWITCH uses.
The FreeSWITCH team is happy to see others taking this soft-switch and putting it to creative use. The Whistle platform is a great example of what FreeSWITCH - and indeed all open source software - is capable of producing. It is amazing to think that just six years ago, Anthony Minessale was staring into that empty text editor, contemplating how to build a telephony engine. In just a few years, FreeSWITCH has gone from an idea to a best-of-breed software library. Like the Hemi that moves a large truck, FreeSWITCH is the VoIP engine in a growing array of products: CudaTel, Ooma, and now Whistle.
The future of open source telephony is looking good!
We have good news about FreeSWITCH: our community is growing leaps and bounds! The reason we know this is because each day we have new ones coming in and asking lots of questions. Additionally, not-so-new users are also asking lots of good questions about how to put FreeSWITCH to good use in various scenarios. However, we are finding that we just don't have enough hours in the day to help everyone. This is a good problem to have - it means that FreeSWITCH is growing beyond being the best-kept secret in open source telephony.
So, how can you help? We have several ways of helping new ones. First and foremost we have our documentation resources: the wiki and the FreeSWITCH book. (We are also slaving away on the FreeSWITCH Cookbook, but that's another story...) We are constantly in need of having our community members update the wiki. The FreeSWITCH developers are constantly adding new features. For example, see the agenda for this week's conference call - all those features were added within the past 3+ weeks, and that's not even the whole of it! As you can see, we have a real need for community members to donate time and energy in keeping our documentation updated.
Secondly, we have direct contact within our community. We have two main ways of keeping in contact: via the mailing list and by gathering together in the IRC channel. We also have the weekly conference call. The most important thing that you can do to help the community is to participate in these venues. Our two most pressing needs are: answering questions to the mailing list and answering questions in IRC. This is where your help is needed most and is also where you can have an immediate impact. If you are not subscribed to the freeswitch-users list then please join today. You don't have to read every message, but we encourage you to scan the subject lines looking for threads where you can help out. Likewise, if you have not joined IRC, please do so. Join irc.freenode.net and register a nickname, then come into #freeswitch. Set your IRC client to beep at you when certain key words are mentioned. For example, if you have experience with the event socket library, tell your IRC client to beep whenever "ESL" is mentioned in the channel. You can then scan the conversation and possibly offer your expertise.
If you look back at how you learned FreeSWITCH I think you'll have fond memories of how people helped you figure things out. I know I do. I first started asking lots of question when OpenZAP appeared in the summer of 2007. Anthony, Brian, and Mike all gave me individual attention. Even though the project has grown immensely over the past four years, they still give individual attention. But we all know that they cannot spend all day answering questions and still have time build a great project. Let's all give back by paying it forward. Pick one or more facets of the community and jump in. Even if you don't know very much you still know more than those who are just getting started. It is never too early start helping others.
Thank you to all who work to make FreeSWITCH such a great project and community. Thank you as well to all of those who read this and decide to give back by helping others. You are the ones who make open source software so rewarding.
SafiSystems has just announced that their Safi Communications Suite 1.5.5beta now has preliminary support for FreeSWITCH. Community members are encouraged to download and test drive this new version.
Safi Communications Suite (SCS) is described as "a powerful multi-platform IVR (Interactive Voice Response), call-flow, and scripting environment." Included is the SafiWorkshop IDE - a tool that allows users to "design, test, debug and deploy advanced logic and call routing applications from a single, unified development environment." SCS and the SafiWorkshop are available for free from the SafiSystems download page.
Thank you to SafiSystems for helping to grow the open source telephony environment!
After countless attempts at stability, the FreeSWICH Team has decided the only way to ensure the proper operation of FreeSWITCH is to include libc 4.1.11 and kernel 184.108.40.206 into the git repository. "Previousy, we have been plagued with countless unknowns on various distros. Now we can be sure FreeSWITCH runs properly and efficiently," says lead developer Anthony Minessale. He adds: "The decision was easy, the hard part will be porting them to Win64 but we're up for the challenge." Brian West was thrilled with the idea and is already working the new design into his mod_NEXT!! application designed to quickly hangup on clueless callers. The team hopes to have the adjustment complete in time for ClueCon 2011.
FreeSWITCH has been around for about 6 years now and this new move will be the most signifigant since the hamster wheel driven timer module designed to keep accurate asyncronous RTP timing through a series of electrodes fitted onto several peices of rodent execercise equipment.
As the video below mentions, it is indeed ClueCon season! We invite all to visit our redesigned ClueCon web site and get registered right away. We will be holding this year's conference at the Sofitel in downtown Chicago, August 9-11. Those staying at the hotel will enjoy a $300 discount off of their conference admission! We have a negotiated a fantastic rate of $155 per night.
On another note, we are also accepting new sponsors as well as proposals for speaking topics. If you would like to speak at ClueCon 2011 then please send your proposal to firstname.lastname@example.org. Keep in mind that our audience enjoys more technical presentations, so don't be afraid to really talk tech!
Looking forward to seeing you all again this August!