freeswitch (1.0.0)
Enhanced sofia sip nat handling
Many fixes found by Klockwork (www.klocwork.com)
Added disable_app_log variable
Fixed mod_local_stream with rates on windows
Fixed finding of files in rate dirs on windows
Fixed memory corruption from sofia_contact function
Added sofia profile param NDLB-received-in-nat-reg-contact
Added sofia profile param aggressive-nat-detection
Fixed video sip calls in proxy media mode
Added bridge_terminate_key var
Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
Enhanced nat handling in proxy media mode in sip
Add progress media to timetable so you can calculate pdd
Fixed seg when using unicast on socket when call has no read_codec
Fixed missed log events on busy box
Added -bleg to intercept
Enhance configure detection of python
Fixed build on solaris and freebsd for several modules
Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
Added param "vm-mailto-notify" to allow sending a notification email
Fixed mod_java build
Fixed mwi failures for some devices that don't subscribe
Removed fsapi functions (killchan, transfer, session_displace, reject)
Removed fsapi functions (session_record, broadcast, hold, media)
Many updates to sofia-sip library including over 100 fixes
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)