We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, 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, 16 or 32 kilohertz and can bridge channels of different rates.
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.
Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.
a Spec Sheet is available on our Wiki.
Celliax Announces Port of Skype Solution To FreeSWITCH
Celliax has recently announced they will be adding support for FreeSWITCH.
http://www.celliax.org/node/503
FreeSWITCH Comes in 8th out of 103 Alternatives to Cisco and Avaya.
We've come in at 8th place in this impressive list of VoIP platforms.
Not too bad considering 1st,2nd,4th and 7th are all derivitaives of Asterisk in one way or another.
http://blog.voipsupply.com/uncategorized/need-an-ip-pbx-101-alternatives...
VoIP Providers Needs are Changing over Time
Providers need an inexpensive solution for hosting multi-tenant, secure, scalable VoIP services. Such a solution is now available in the marketplace. When VoIP technology first moved into the mainstream, solutions were rare and expensive. Solutions introduced included Session Border Controllers (SBCs), softswitches, and Gateways.
Cluecon 2008 Videos Online Now!!!
Thanks Then No Thanks.... No Place For Me In The Future Of Telephony
I have been so busy coding for the last few years I haden't even noticed. Before I started FreeSWITCH, I was an avid Asterisk developer putting all my spare time into the betterment of the project. When the book was released Asterisk: The Future of Telephony by O'Reilly Media I thought it was awesome that they mentioned one time Asterisk bug marshall and now FreeSWITCH QA Manager Brian West as well as myself in the book:
• Brian K. West, for your commitment to the community, Asterisk, our book, and
open-source telephony
Anthony Minessale (a.k.a. anthm) is one of the unsung heroes of Asterisk develop-
ment. The number of people who have contributed to Asterisk development are many; the number who can claim to have matched Anthony’s efforts are few.
WOW! guess what? They cut both mentions out of the book. Yet they certianly didn't cut all the contributions we gave them from the code base or the book. I would not have beleived it but they actually seem to have purposely removed our names from the book. I guess my efforts were not quite as memorable as they first thought. Yet I believe most of the things on my list of contribtions at http://www.cluecon.com/anthm.html still remain.
I have to say, This is the biggest dissapointment to date from Asterisk since the time Mark Spencer told me I was going to receive a gift from the Digium for all my hard work on the Asterisk project back in 2005 and then never actually gave it to me...........Sigh......







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