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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 2008-04-23 17:33:00 +00:00
parent 7ae460ea23
commit 727cf3a673
4 changed files with 79 additions and 10 deletions
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35
debian/changelog vendored
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@ -1,3 +1,38 @@
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
-- 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

18
debian/files vendored
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@ -1,9 +1,9 @@
freeswitch_1.0~rc3-1_i386.deb net extra
freeswitch-spidermonkey_1.0~rc3-1_i386.deb net extra
freeswitch-dev_1.0~rc3-1_i386.deb net extra
freeswitch-codec-passthru-g7231_1.0~rc3-1_i386.deb net extra
freeswitch-codec-passthru-amr_1.0~rc3-1_i386.deb net extra
freeswitch-codec-passthru-g729_1.0~rc3-1_i386.deb net extra
freeswitch-lang-en_1.0~rc3-1_i386.deb net extra
freeswitch-lang-de_1.0~rc3-1_i386.deb net extra
freeswitch-lang-fr_1.0~rc3-1_i386.deb net extra
freeswitch_1.0~rc4-1_i386.deb net extra
freeswitch-spidermonkey_1.0~rc4-1_i386.deb net extra
freeswitch-dev_1.0~rc4-1_i386.deb net extra
freeswitch-codec-passthru-g7231_1.0~rc4-1_i386.deb net extra
freeswitch-codec-passthru-amr_1.0~rc4-1_i386.deb net extra
freeswitch-codec-passthru-g729_1.0~rc4-1_i386.deb net extra
freeswitch-lang-en_1.0~rc4-1_i386.deb net extra
freeswitch-lang-de_1.0~rc4-1_i386.deb net extra
freeswitch-lang-fr_1.0~rc4-1_i386.deb net extra

34
docs/ChangeLog
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@ -1,3 +1,37 @@
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
freeswitch (1.0.rc3)
Enhance xml menu system

2
freeswitch.spec
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@ -5,7 +5,7 @@ Name: freeswitch
Summary: FreeSWITCH open source telephony platform
License: MPL
Group: Productivity/Telephony/Servers
Version: 1.0.rc1
Version: 1.0.rc4
Release: 1
URL: http://www.freeswitch.org/
Packager: Michal Bielicki