freeswitch - Open Source Telephony Platform
FreeSWITCH is an open source telephony platform designed to facilitate the
creation of voice and chat driven products scaling from a soft-phone up to
a soft-switch. It can be used as a simple switching engine, a media gateway
or a media server to host IVR applications using simple scripts or XML to
control the callflow.
We support various communication technologies such as SIP, H.323 and
GoogleTalk making it easy to interface with other open source PBX
systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk.
We also support 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 runs 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 sipXecs, OpenSER,
Asterisk, CodeWeaver and OpenPBX.
- * 2011-06-30 - Lott Caskey <lottc[AT]fugitol[DOTCOM]> 1.0.7-4
- Update to latest git revision.
- Add conditions for shared objects.
- Remove yaml.
- Add option to build sounds. Make sounds depend on FS >= current version-release.
- * 2011-01-24 - Lott Caskey <lottc[AT]fugitol[DOTCOM]> 1.0.7-1
- Change to RedHat FHS file layout
- Added mod_yaml
- Use shared objects.
- * 2011-01-18 - - michal.bielicki[AT]seventhsignal.de
- Fedora adjustments
- * 2010-10-15 - - michal.bielicki[AT]seventhsignal.de
- added mod_curl
- * 2010-10-09 - - michal.bielicki[AT]seventhsignal.de
- added mod_silk
- added mod_codec2
- moved from openzap to freetdm to make way for inclusion of libsng_isdn and wanpipe
- added mod_freetdm
- added mod_cidlookup
- added more runtime dependencies