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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.
|freeswitch-1.2.8-1.mga3.x86_64.html||FreeSWITCH open source telephony platform||Mageia Cauldron for x86_64||freeswitch-1.2.8-1.mga3.x86_64.rpm|
|freeswitch-1.2.8-1.mga3.i586.html||FreeSWITCH open source telephony platform||Mageia Cauldron for i586||freeswitch-1.2.8-1.mga3.i586.rpm|
|freeswitch-1.2.8-1.mga3.x86_64.html||FreeSWITCH open source telephony platform||Mageia 3 for x86_64||freeswitch-1.2.8-1.mga3.x86_64.rpm|
|freeswitch-1.2.8-1.mga3.i586.html||FreeSWITCH open source telephony platform||Mageia 3 for i586||freeswitch-1.2.8-1.mga3.i586.rpm|
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