update changelog

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

+ -- Removed MP3 format and codec
+ -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
+ -- Fixed various compiler warnings and clean up source tree
+ -- Preliminary AES Support
+ -- Fix SIP REINVITE
+ -- Outbound SIP registration behind NAT using externip
+ -- More CLI documentation and clean up
+ -- Pin numbers on MeeMe
+ -- Dynamic MeetMe conferences are more consistent with static conferences
+ -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
+ -- ODBC support for logging CDRs
+ -- Indications for Norway and New Zeland
+ -- Major redesign of app_voicemail
+ -- Syslog support
+ -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
+ -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console 
+ -- Properly reaping any zombie processes
+ -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
  -- Make PRI Hangup Cause available to the dialplan
  -- Verify included contexts in extensions.conf
  -- Add DESTDIR support for building RPMs and packages
  -- Do route lookups on OpenBSD
- -- Add support for building on FreeBSD
+ -- Add support for building on FreeBSD and OS X
  -- Add support for PostgreSQL in Voicemail
  -- Translate SIP hangup cause to PRI hangup cause where needed
  -- Better support for MOH in IAX2
@@ -14,6 +32,7 @@
  -- More flexible CDR handling
  -- Distinguish between BUSY and FAILURE on outbound calls
  -- Add initial support for SCCP via chan_skinny
+ -- Better support for Future Group B signaling
 Asterisk 0.5.0
  -- Retain IAX2 and SIP registrations past shutdown/crash and restart
  -- True data mode bridging when possible