freeswitch/debian/changelog
Brian West f57c69ceb0 thanks CtRiX
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@8306 d0543943-73ff-0310-b7d9-9358b9ac24b2
2008-05-08 16:44:33 +00:00

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freeswitch (1.0~8303) unstable; urgency=low
* pacchetto
-- root <root@fs.navynet.it> Tue, 6 May 2008 09:46:26 +0000
freeswitch (1.0~rc4-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Wed, 23 Apr 2008 12:58:00 -0400
freeswitch (1.0~rc3-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
freeswitch (1.0~rc2-1) unstable; urgency=low
* 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
-- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
freeswitch (1.0~rc1-1) unstable; urgency=low
* loads of fixes
new cdr-csv module
new spidermonkey-curl module
-- Michal Bielicki <michal.bielicki@voiceworks.pl> Mon, 14 Jan 2008 23:37:04 +0100
freeswitch (1.0~beta3-1) unstable; urgency=low
* Additional scripts for changing the user to freeswitch
Added Startup Scripts
Monit integration
Settings file for integration into init
init.d file
added user freeswitch to own and run all off freeswitch
cleaned up config file control
new upstream release
split off codec pakcages
split off spidermonkey packages
-- Michal Bielicki <michal.bielicki@voiceworks.pl> Tue, 27 Nov 2007 13:20:21 +0100
freeswitch (1.0~beta2-1) unstable; urgency=low
* New upstream release
-- Paul van Genderen <paulvg@member.fsf.org> Wed, 17 Oct 2007 19:32:09 +0200
freeswitch (1.0~beta1-1) unstable; urgency=low
* New packages.
-- Robert McQueen <robot101@debian.org> Sun, 12 Nov 2006 17:32:23 -0500