Here is from freeswitch.org

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 PBX, 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.

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, 16 or 32 kilohertz and can bridge channels of different rates.

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.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipX, The Asterisk Open Source PBX and Call Weaver.

Possible Uses

• Rating & Routing Server
• Transcoding B2BUA
• IVR & Announcement Server
• Conference Server
• Voicemail Server
• SBC (Session Border Controller)
• Basic Topology Hiding Session Border Controller
• Zaptel, Sangoma, Rhino, PIKA Hardware Support (Analog and PRI)

Features

• Centralized User/Domain Directory (directory.xml)
• Nano Second CDR granularity
• Call recording (In Stereo caller/callee left/right)
• High Performance Multi-Threaded Core engine
• Configuration via CURL to your http server (xml_curl).
• XML Config files for easy parsing.
• Protocol Agnostic
• Configurable RFC2833 Payload type
• Inband DTMF generation and detection.
• Software based Conference (no hardware requirement)
• Wideband Conferencing
• Media / No Media modes
• Proper ENUM/ISN dialing built in
• Detailed CDR in XML
• Radius CDR
• Subscription server
  • Shared Line Appearances
  • Bridged Line Appearances
• Enterprise/Carrier grade Eventing Engine. (XML Events, Name Value Events, Multicast Events)
• Loadable File formats and streaming
• Stream to Shoutcast
• Multi-lingual Speech Phrase Interface
• ASR/TTS support (native and via MRCP)
• Basic IP/PBX features
• Automated Attendant
• Custom Ring Back Tones
• XML RPC support
• Multiple format CDR’s supported
• SQL Engine provides session persistence
• Thread Isolation
• Parallel Hunting
• Serial Hunting
• Mozilla Public License
• Support
  • Paid support available
  • Free support via IRC & e-mail
• Many supported codecs
  • G.722 (wideband)
  • G.711
  • G.726 (16k,24k,32k,48k) AAL2 and RFC3551
  • G.723.1 (passthru)
  • G.729 (passthru)
  • AMR (passthru)
  • iLBC
  • speex (narrow and wideband)
  • lpc10
  • DVI4 (ADPCM) 8khz and 16khz

Applications

• Voicemail
  • Multitenancy – Enterprise/Carrier configuration
  • Time of Day Greetings
  • Urgent Message Tagging
  • EMail Delivery
  • Playback and Rerecord messages before delivery.
  • Keys are templates so you can rearrange to fit your needs.
  • Callback support from inside voicemail.
  • Podcast of Voicemail (RSS)
  • Message Waiting Indicator (MWI)
• Support for Queues (via mod_fifo)
• Parking (via mod_fifo)
• Conference
  • Software based Conferencing without any hardware requirements.
  • Wideband conferences.
  • Multiple on-demand or scheduled conferences with entry/exit announcements
  • Play files into the conference or a single member.
  • Relationships
  • TTS integration
  • Transfers
  • Outbound Calling
  • Configurable Key Lay
  • Volume, Gain and Energy level per call.
  • Bridge to Conference trasition
  • Multi Party outbound dialing.
• RSS Reader
• T.30 Audio Fax (via mod_fax)

Protocols

• SIP
  • UDP, TCP, SCTP and TLS transports for full sip compliance.
  • IPv6 Support
  • SIP Session timers
  • RTP Timers
  • RFC3263 (SRV and NAPTR)
  • SRTP via SDES (works with polycom, snom, linksys and grandstream)
  • Blind SIP Registration
  • STUN Support
  • Jitter buffer
  • NAT Support
  • Distributed sip registrations
  • Late Codec Negotiation
  • Multiple sip registrations per user account.
  • Multitenancy – Multiple sip UAs
  • SIP Reinvites.
  • Can act as an SBC (session border controller)
  • Manage Presence
  • SIP/SIMPLE (can gateway to other chat protocols)
  • SIP Multicast Paging support for Linksys and Snom
  • Intercom/AutoAnswer support.
  • Call features like Call Hold(Re-INVITE), Blind Transfer(REFER), Call Forward(302) etc.

• IAX (Via a modified libiax2.)

• Jingle
  • Interop with GoogleTalk and Telepathy

• h.323 (Currently only supports H.323 via the Woomera protocol. This should change soon.)

Languages

• Javascript (Using the Spidermonkey Javascripting engine.)
  • ODBC Support from inside your Javascript
  • Extendable modules for Javascript
  • Tone Generation

• Python
• Perl
• Lua

Cross Platform

• Builds native on Windows in MSVC
• Builds on Mac OS X, Linux, Solaris and *BSD.

Minimum/Recommended System Requirements

• 32bit OS (64bit recommended)
• 512MB Ram (1GB recommended)
• 50MB of Disk Space

System requirements depend on your deployment needs. We recommend you plan for 50% duty cycle.

Performance

• Tested under load for over 100 hours
• 10,000,000+ calls
• At rates exceeding 50 CPS

Performance will vary depending on application. You will need to test for your particular situation.