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.
FreeSWITCH 1.0.1 "Phoenix" Released!
The first round of improvements are in from our 1.0 line!
There are several new features including:
Open Source ASR with mod_pocketsphinx
Open Source TTS with mod_flite
Improved ACL with user matching.
Enhanced SNOM support.
And many more!

Open Source VoIP: Asterisk or FreeSWITCH?
blogs.zdnet.com —
When the time came for a new PBX, Brian Snipes chose to do something a bit unconventional. The IT manager at law firm Hare, Wynn, Newell, and Newton LLP didn’t purchase a commercial PBX, nor did he settle on the open source market leader, Asterisk. No, Snipes chose to become the first enterprise to deploy the new open source IP PBX, FreeSwitch.
ClueCon 2008 is just under 3 weeks away!
I wanted to let everyone know that ClueCon 2008 is just under 3 weeks away!
Everyone who is planning to attend:
Please make sure you are registered and have your rooms booked.
We have an overflowing schedule of speakers but we could still probably
squeeze in a few more lightening talks if anyone wants to speak.
This year is shaping up to be the best yet with developers or reps from
most of the popular Open Source VoIP applications represented including:
FreeSWITCH(tm) (naturally) ;)
OpenSER
OPAL
sipX/SIPfoundry
Asterisk
Yate
Bayonne
SofaSWITCH
We also have reps or developers from several hardware vendors including:
Sangoma
PIKA
Digium
Dialogic
SNOM
OCTASIC
We also have 2 professional voice firms attending including:
GM Voices (creator of the famous Cepstral Callie and all the FreeSWITCH(tm) prompts)
Allison Smith (The voice of Asterisk(tm))
http://www.cluecon.com
sipXecs incorporates FreeSWITCH into next release
SIPfoundry announced today that they are using FreeSWITCH in their next release.
sipXecs adds a fully featured conferencing server. That conferencing server is based on FreeSWITCH and over the last couple of months a very good cooperation emerged
between the two respective developer communities. There are lot's of
synergies between FreeSWITCH and sipXecs where FreeSWITCH provides a
very powerful state-of-the-art media framework. sipXecs adds
ease-of-use Web based management, cluster management, and all the other
services provided by sipXecs already.
see sipfoundry.org for details:
http://www.sipfoundry.org/component/option,com_ezine/Itemid,82/task,read...
Spit Will Be Worse Than Spam?? NOT!!!
I was reading this story over on Slashdot about how SPIT (spam over internet telephony) will be worse than SPAM. I find it hard to believe that will be the case. If you use software like FreeSWITCH to frontend all your phone calls it can give you the ability to knock out* the people you don't want to talk to in a heart beat.
What is your take on this?
* No you can't literally knock people out via telephone... yet!





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