getting ready for rc4
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@8178 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale
parent
7ae460ea23
commit
727cf3a673
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@ -1,3 +1,38 @@
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freeswitch (1.0~rc4-1) unstable; urgency=low
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* Add tab completion in cli
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Add "inline" dialplan
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Fixed segfault in enum
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Enhance enum to fork dial equal priority entries
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Added auto-reload to enum
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Fixed odbc bug is mod_sofia presence handling
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Add presence for conference and dial an eavesdrop
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Fix stack overflow segfault when recursively parking calls
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Fixed race is sofia registration handling
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Enhance sofia registration, unregister on keep-alive OPTIONS failure
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Added internal routing loop detection/avoidance
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Fixed race in bgapi in event socket
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Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
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Fixed re-setting sound prefix to no prefix after a pharse
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Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
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Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
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Add originate_timeout to originate vars
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Fixed hanging channels in mod_portaudio
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Added auto time sync on vps migration to different hardware
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Fixed seg on transfer when both legs are not sip
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Added configurable dtmf duration defaults
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Enhanced voicemail, allow interruption of hello message
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Fixed voicemail to not light up light on saved messages
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Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
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Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
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Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
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Added hold/unhold dialplan apps
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Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
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Backport fixes from sofia-sip tree
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Fixed MSVC build
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Fixed segfault on sip SUBSCRIBE with Expires: 0
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-- Michael Jerris <mike@jerris.com> Wed, 23 Apr 2008 12:58:00 -0400
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freeswitch (1.0~rc3-1) unstable; urgency=low
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* Enhance xml menu system
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fixes upstream from sofia-sip library
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@ -1,9 +1,9 @@
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freeswitch_1.0~rc3-1_i386.deb net extra
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freeswitch-spidermonkey_1.0~rc3-1_i386.deb net extra
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freeswitch-dev_1.0~rc3-1_i386.deb net extra
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freeswitch-codec-passthru-g7231_1.0~rc3-1_i386.deb net extra
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freeswitch-codec-passthru-amr_1.0~rc3-1_i386.deb net extra
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freeswitch-codec-passthru-g729_1.0~rc3-1_i386.deb net extra
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freeswitch-lang-en_1.0~rc3-1_i386.deb net extra
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freeswitch-lang-de_1.0~rc3-1_i386.deb net extra
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freeswitch-lang-fr_1.0~rc3-1_i386.deb net extra
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freeswitch_1.0~rc4-1_i386.deb net extra
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freeswitch-spidermonkey_1.0~rc4-1_i386.deb net extra
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freeswitch-dev_1.0~rc4-1_i386.deb net extra
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freeswitch-codec-passthru-g7231_1.0~rc4-1_i386.deb net extra
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freeswitch-codec-passthru-amr_1.0~rc4-1_i386.deb net extra
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freeswitch-codec-passthru-g729_1.0~rc4-1_i386.deb net extra
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freeswitch-lang-en_1.0~rc4-1_i386.deb net extra
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freeswitch-lang-de_1.0~rc4-1_i386.deb net extra
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freeswitch-lang-fr_1.0~rc4-1_i386.deb net extra
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@ -1,3 +1,37 @@
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freeswitch (1.0.rc4)
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Add tab completion in cli
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Add "inline" dialplan
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Fixed segfault in enum
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Enhance enum to fork dial equal priority entries
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Added auto-reload to enum
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Fixed odbc bug is mod_sofia presence handling
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Add presence for conference and dial an eavesdrop
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Fix stack overflow segfault when recursively parking calls
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Fixed race is sofia registration handling
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Enhance sofia registration, unregister on keep-alive OPTIONS failure
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Added internal routing loop detection/avoidance
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Fixed race in bgapi in event socket
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Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
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Fixed re-setting sound prefix to no prefix after a pharse
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Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
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Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
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Add originate_timeout to originate vars
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Fixed hanging channels in mod_portaudio
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Added auto time sync on vps migration to different hardware
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Fixed seg on transfer when both legs are not sip
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Added configurable dtmf duration defaults
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Enhanced voicemail, allow interruption of hello message
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Fixed voicemail to not light up light on saved messages
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Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
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Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
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Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
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Added hold/unhold dialplan apps
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Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
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Backport fixes from sofia-sip tree
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Fixed MSVC build
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Fixed segfault on sip SUBSCRIBE with Expires: 0
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freeswitch (1.0.rc3)
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Enhance xml menu system
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@ -5,7 +5,7 @@ Name: freeswitch
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Summary: FreeSWITCH open source telephony platform
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License: MPL
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Group: Productivity/Telephony/Servers
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Version: 1.0.rc1
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Version: 1.0.rc4
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Release: 1
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URL: http://www.freeswitch.org/
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Packager: Michal Bielicki
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