freeswitch (1.0.rc6)
	Changed to not allow pass_2833 on transcoded calls (it never worked, now it will tell you)
	Enhanced sofia sip nat handling
	Fix libedit build on solaris
	Fix session timers in mod_sofia
	Fix conference fire-call
	Change: add var_event down into the endpoints so chans with no parents can still pass options
	Added enable-post-var param to xml_rpc
	Fix mod_lua build on solaris
	Many fixes found by Klockwork (www.klocwork.com)
	Add unregister event in mod_sofia
	Enhance python configure detection
	Add vm_boxcount api func
	Fixed att_xfer issue
	Fix sip now includes the Allow-Events header in more places

freeswitch (1.0.rc5)

	Changed internal state names to avoid confusion
	Fixed video negotiation
	Enhanced accuracy of windows timer
	Fixed mod_ldap build
	Added dialplan and context to sql table for channels
	Multiple fixes to mod_lua and mod_perl
	Fixed logic bug in fifo causing segfault
	internal changes to sip stack so we can remove a hash redundant to the stack
	Fixed multiple memory leaks in mod_sofia
	Fixed event fetch segfault on sip subscribe
	Fixed segfault on timer rollover in sofia on 64bit
	Fixed audio timing issues in mod_portaudio
	Changed names of sip profiles in default config to avoid confusion
	Fixed memory usage leak-like behavior when playing files requiring resampling
	Removed some unused api's
	Fix rtp timeout when playing moh
	Removed some un-needed libraries and files from tree
	Fixed multiple issues in sip stack including multiple segfaults
	Added support for sip transfers on bypass_media and proxy_media calls
	Added say application
	Fixed --disable-debug configure option
	Enhanced switch_cpp wrapper (and perl, python, lua, java)
	Fixed segfault on inavalid stun response
	Fixed configure help output
	Fixed segfault on mp3 playback
	Fixed assert on invalid sdp (missing m= line)
	Added configurable windows service name
	Fixed proxy mode call transition to non proxy call
	Fixed solaris build of voipcodecs
	Fixed sofia seg when call failure edge case

freeswitch (1.0.rc4)

	Add tab completion in cli
	Add "inline" dialplan
	Fixed segfault in enum
	Enhance enum to fork dial equal priority entries
	Added auto-reload to enum
	Fixed odbc bug is mod_sofia presence handling
	Add presence for conference and dial an eavesdrop
	Fix stack overflow segfault when recursively parking calls
	Fixed race is sofia registration handling
	Enhance sofia registration, unregister on keep-alive OPTIONS failure
	Added internal routing loop detection/avoidance
	Fixed race in bgapi in event socket
	Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
	Fixed re-setting sound prefix to no prefix after a pharse
	Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
	Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
	Add originate_timeout to originate vars
	Fixed hanging channels in mod_portaudio
	Added auto time sync on vps migration to different hardware
	Fixed seg on transfer when both legs are not sip
	Added configurable dtmf duration defaults
	Enhanced voicemail, allow interruption of hello message
	Fixed voicemail to not light up light on saved messages
	Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
	Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
	Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
	Added hold/unhold dialplan apps
	Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
	Backport fixes from sofia-sip tree
	Fixed MSVC build
	Fixed segfault on sip SUBSCRIBE with Expires: 0
	Added mod_say_zh
	Added --with-pyton and --with-pyton-config configure options
	Added mod_lua
	Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
	Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
	Added back mod_perl
	Added sofia gateway option ping to adjust options ping frequency
	Added .net event socket lib to contrib
	Fixed passing of exact response codes of sip across a bridge
	Added mod_reference, reference endpoint module
	Enhanced build so you can now make commented out modules using "make mod_name"
	
freeswitch (1.0.rc3)

        Enhance xml menu system
        Fixes upstream from sofia-sip library
        Enhance mod_fifo
        Added close method to ODBC spidermonkey class
        Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
        Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses

freeswitch (1.0.rc2)

	Fixed speex protocol negotiation issues (8k vs 16k)
	Fixed mod_iax race conditions
	Fixed ptime negotiation issues when re-packetizing
	Added ip based acl lists

freeswitch (1.0.rc1)


	
