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.
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.
This came across recently. An intrepid user has codified his hard-earned knowledge into a nice blog post that discusses how he set up his FreeSWITCH to use two NICs. Check it out and leave him feedback on how well it works. When we confirm that it works we can link to it from our wiki.
Some of you may be familiar with Eric des Courtis' mod_vmd - Voicemail Detect. He has rewritten this module as mod_avmd - Advanced Voicemail Detect. The new module uses an improved algorithm during audio analysis.
I asked Eric about why he wrote this module (as well as mod_vmd) and why he contributed back to FreeSWITCH. Like most open source developers, he understands that "giving back" can help others as well as the developer:
The good folks over at TransNexus have written a new module for FreeSWITCH that implements the Open Settlement Protocol: mod_osp. Jim Dalton from TransNexus describes it this way:
"mod_osp is a module that generates route queries and CDRs in the ETSI Open Settlement Protocol (OSP) format. OSP is basically a standard set of XML messages for requesting routing and access for inter-domain communications."
Complete information about the OSP protocol can be found at the TransNexus OSP Toolkit page. You may be wondering, "Why OSP? And why FreeSWITCH?" I asked Jim Dalton about the choice to create an OSP implementation in FreeSWITCH. He notes:
I thought that FreeSWITCH supporters everywhere would be happy to hear the news: we have completed the revised draft of the FreeSWITCH book and have submitted it to the publisher! It has been quite a learning experience for Anthony, Darren, and myself. We've all done a ton of work on this writing project and we sincerely hope that you will enjoy having an actual printed (or PDF) FreeSWITCH book.
I thought everyone would like to know that Katherine, aka "Callie" - the voice of FreeSWITCH - will be attending ClueCon this year! Katherine works for GM Voices, one of ClueCon's media sponsors and a great supporter of the FreeSWITCH project. We are all looking forward to meeting her in person. Please stay tuned for more updates. Don't forget - if we have 200 registered users by July 4th then we all get to see Brian shave his head at ClueCon! (You can rest assured that event will be captured on video for all the world to see.)
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.