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+== USAGE ==
+
+Start the freeswitch daemon with /etc/rc.d/freeswitch.
+Add 'freeswitch' to DAEMONS in /etc/rc.conf and it will start at boot time.
+All configuration is done in /etc/freeswitch/.  
+/usr/bin/fs_cli will bring up the console to freeswitch once it's running.
+
+== SUPPORT ==
+
+See http://wiki.freeswitch.org for up-to-date configuration documentation.
+Official (paid) support available through http://freeswitch.org or 
+consulting at freeswitch dot org.
+#freeswitch on Freenode IRC network.
+
+== DESCRIPTION ==
+
+From http://freeswitch.org:
+
+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,
+Mac 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. 
+
+== FEATURES ==
+
+From http://wiki.freeswitch.org/wiki/Specsheet:
+
+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), and Khomp Brazilian telephony hardware manufacturer
+ * Fax server
+ * T.38 gateway, termination, and origination mode
+ * T.30 to T.38 and T.38 to T.30 gateway
+  See also: mod_spandsp
+  And, of course, a PBX
+
+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 (mod_xml_curl).
+ * XML Config files for easy parsing.
+ * Protocol Agnostic
+ * ZRTP support for transparent RTP based key exchange and encryption
+ * Configurable RFC 2833 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 and play from Shoutcast and Icecast
+ * Multi-lingual Speech Phrase Interface
+ * ASR/TTS support (native and via MRCP)
+ * Basic IP/PBX features
+ * Automated Attendant
+ * Custom Ring Back Tones (early_media)
+ * XML-RPC support
+ * Multiple format CDRs 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
+ * CELT (32 kHz ahd 48 kHz)
+ * G.722.1 (wideband)
+ * G.722.1C (wideband 32 kHz)
+ * G.722 (wideband)
+ * G.711
+ * G.726 (16k, 24k, 32k, 48k) AAL2 and RFC 3551
+ * G.723.1 (passthrough)
+ * G.729AB (Requires a license unless using passthrough)
+ * AMR (passthrough)
+ * iLBC
+ * Speex (narrow and wideband)
+ * LPC-10
+ * DVI4 (ADPCM) 8 kHz and 16 kHz
+ * SILK
+ * Video Codecs (passthrough):
+ * Theora
+ * H.261
+ * H.263
+ * H.264
+ * MP4
+ * More Codec Information: http://wiki.freeswitch.org/wiki/Codecs
+ * Live Migration of calls from one FreeSWITCH box to another. See 
+   http://wiki.freeswitch.org/wiki/Freeswitch_HA
+
+Applications
+
+ * Voicemail
+ * Multitenancy - Enterprise/Carrier configuration
+ * Time of Day Greetings
+ * Urgent Message Tagging
+ * E-mail 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 or mod_callcenter)
+ * 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 transition
+ * Multi Party outbound dialing.
+ * Inbound Call Center Queues
+ * RSS Reader
+ * Fax endpoint, gateway and passthrough mode.
+ * T.30 (G.711) Audio Fax (via mod_spandsp) formerly known as mod_fax.
+ * T.38 faxing (gateway, endpoint and passthrough)
+
+Protocols
+
+ * SIP
+ * UDP, TCP, SCTP and TLS transports for full SIP compliance.
+ * IPv6 Support
+ * SIP Session timers
+ * RTP Timers
+ * RFC 3263 (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.
+ * Jingle
+ * Interop with Google Talk, Google Voice, and Telepathy
+ * H.323 with mod_opal (opalvoip.org)
+ * mod_h323 - H.323 Endpoint module based on the h323plus library.
+ * mod_skinny - Skinny Call Control Protocol (SCCP)
+
+Languages
+
+ * JavaScript (Using the SpiderMonkey JavaScript 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
+
+ * 32-bit OS (64-bit 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.