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.
G.729A License (Linux)
G.729 is a patented audio codec often used in VoIP.
FreeSWITCH provides a licensed commercial module for $10 per channel. (that's one encoder and one decoder.)
You can buy licenses separately and combine them later when they are for the same machine.
If your machine stops working, contact us at email@example.com to get new licenses for your replacement machine.
Click "Add to Cart" below to add a single channel license to your cart. When your cart is displayed you may change the quantity.
1 single channel of G.729A License (x86_64 & i386 ONLY) All sales final.
DOWNLOADS ARE AT http://files.freeswitch.org/g729
NOTICE: If you need to order channels for different servers ie. 30 for one and 20 for another please process them as separate orders. The licenses can NOT be split up after the license code is issued and once activated can NOT be moved to a new system. They are locked to the system you activate them on. Currently doesn't work on OpenVZ