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freeswitch: FreeSWITCH open source telephony platform
- Summary
- 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, IAX2 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.
Changelog
- * Mon Feb 4 14:00:00 2008 - michal.bielicki@voiceworks.pl
- More fixes to specfile
- First go at SFE files
- * Sun Feb 3 14:00:00 2008 - michal.bielicki@voiceworks.pl
- abstraction of prefix
- more wrong stuff deleted
- abstraction of mkdir, mv, rm, install etc into macros
- * Fri Jan 18 14:00:00 2008 - michal.bielicki@voiceworks.pl
- fixes, fixes and more fixes in preparation for rc1