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.
I will be attending SpeechTek in NYC next week(August 20-23). If you're in the area please feel free to find me (Brian West). I'll be hanging around in the LumenVox/Cepstral area. Also be on the look out for some existing announcements coming from these guys. If your all about speech then this is the event to be at. Hope to see you there!
Here is a little bit of information on how some companies are starting to use FreeSWITCH in production.
ClueCon was last week and we are all pretty tired. =D
Thanks to everyone who attended and we look forward to seeing you in Aug 2008!
A lot of great information was released including the new OpenZAP library that provides zaptel and sangoma support for FreeSWITCH on both analog and ISDN. The project is still new so we need to do a lot more development and testing but we already have basic functionality underway. Visit our IRC channel #freeswitch on irc.freenode.net or see http://fisheye.freeswitch.org/browse/OpenZAP for details.
Moshe Yudkowsky also provided this informative report on his blog:
I wanted to remind everyone that ClueCon is in a few weeks and if you plan to attend you should reserve your place now.
Sangoma will be giving away a few FREE T1 and analog cards to some lucky attendees
Zap Micro will be giving a FREE 4 port analog card with 1 FXO and 1 FXS module to all attendees.
Rhino will also be giving away a FREE Cards to some lucky attendees.
A while ago I made this conference application for Asterisk 1.2. Since I don't use it much these days, I thought I'd share it with everyone so download it. You can also just build it right from the net with the astxs utility I created (included in the asterisk distribution). What? Isn't this the FreeSWITCH homepage? you ask. Well, I did my fair share of Asterisk development before I decided to write FreeSWITCH. In fact, I'm still the #3 most decorated developer in thier Karma Hall of Fame even though I have been busy for almost a year and a half doing development here. It supports a bunch of features like: silence supression, playing files, and a bunch more things you can do with the FreeSWITCH conference (but not all of them =D) "lock", "unlock", "mute", "unmute", "kick", "mark", "list", "killsound", "play", "dial", "admin", "unadmin", "vol", "silence", "verbose", "dtmf" To install it right from the net follow this simple instruction. From the 1.2 source tree, where you normally type make, execute this command: export ASTSRC=`pwd` perl ./contrib/scripts/astxs -install http://www.freeswitch.org/asterisk_stuff/app_confcall.c Also get the config from this url: http://www.freeswitch.org/asterisk_stuff/confcall.conf
FreeSWITCH's Open Source Softswitch Controls TelcoBridges' Carrier-Grade Telecom Platform to Route PSTN Calls to Truphone WiFi Subscribers
Montreal, Quebec, Canada (PRWEB) June 5, 2007 -- TelcoBridges and FreeSWITCH announce that Truphone has selected FreeSWITCH and TelcoBridges' proven, reliable and highly redundant carrier-grade telephony platform to enable VoIP calls on mobile phones. Truphone is a mobile internet network operator that brings VoIP to mobile phones via WiFi using SIP. The three companies have collaborated closely to port FreeSWITCH's open source telephony application code to TelcoBridges' hardware platform and have adapted it successfully to fulfill Truphone's requirements for a media gateway, bridging calls between the Internet, using VoIP, and the PSTN.
I’ve been working on FreeSWITCH for nearly 2 years and on the dawn of our first release I wanted to take some time to share the story behind the software project and provide a glimpse of what’s to come. This story will also appear in the first issue of OST Magazine so get a copy, it's FREE!
I would like to announce that FreeSWITCH will be entering
into a BETA status within the week so we can produce a series of release candidates which will ultimately produce a formal release by the end of the summer or sooner if possible.
Our software is growing rapidly and we've come a long way from our modest initial public release in January of 2006. I hope everyone enjoys the opportunity to participate in the development process which is one of the best benefits of open source software in my opinion. Not only do we have a few finishing touches to put on the code, we also have to institute a version policy, make sure the WIKI is accurate and, of course, find all the bugs so we can focus our energy on forward development and stay away from nasty unresolved bugs.
In order to make the debugging process successful we ask that everyone use our jira tracker for all bug reports http://jira.freeswitch.org, feature requests, or feature contributions. We are glad to help but it's beginning to be more than we can handle in real-time so we really need to document issues on the tracker so we won't forget!
We also ask anyone who receives help to please pay it forward and document it on the WIKI http://wiki.freeswitch.org. There are plenty of logistical and clerical necessities to make FreeSWITCH a success so anyone who is interested in helping out by being a bug marshal or managing the svn for one of the modules or any way you think you can help, please mail the dev list http://lists.freeswitch.org or visit us on IRC. Please also feel free to start a new page on the wiki with your irc nick and your paypal address or wishlist urls so when you help someone they can show their gratitude properly.
Thank you all for participating and making FreeSWITCH a fun project!
There is an update to the pre-release of the N800 version of FreeSWITCH at
The new version includes up to date code and mod_speex built just for arm and the new jitterbuffer to be used with mod_alsa (a clone of portaudio that uses libasound just for the N800)
There is also a mini-webserver running on port 8080
with l/p freeswitch/works will give you a small
web based softphone of sorts (enter 888 and press dial to call our developers conference)
This week we added an optional jitter buffer to the RTP stack that
you can turn on with a channel variable from your dialplan.
On an inbound call for use on the inbound channel:
(setting this before the call is answered is mandatory)
<action application="set" data="jitterbuffer_msec=180"/>
Or to set it on the subsequent outbound call:
export sets a variable on both the current channel and on
any channels it creates, the 'nolocal:' disables setting it on the
current channel and only sets it on the subsequent outbound channels
<action application="export" data="nolocal:jitterbuffer_msec=180"/>
<action application="bridge" data="firstname.lastname@example.org"/>