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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Mark Spencer 2004-11-01 02:43:53 +00:00
parent 668001f9c8
commit fbc2051442
1 changed files with 12 additions and 0 deletions
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@ -1,8 +1,20 @@
-- Pass redirecting number on PRI calls
-- Add RTP debug support
-- Misc Debugging improvements
-- Add TALK_DETECTED variable
-- Adding Q.SIG switchtype option to chan_zap
-- Added pbx_builtin_serialize_variables
-- Update to new iLBC codec
-- Add CLI for realtime stuff
-- Add DUNDi.... (http://www.dundi.com)
-- Misc Memory fixes
-- Voicemail improvements from the bug tracker
-- Major revamp of PBX core including 'n' and 's' priorities and labels
-- Deprecate pbx_wilcalu and app_qcall in favor of pbx_spool
-- Remove old chan_iax and chan_vofr
-- Major Caller*ID Restructuring
-- Realtime API (IAX, SIP and Voicemail)
-- codecs.conf to tune various codec options (ie Speex)
Asterisk 1.0.1
-- Added AGI over TCP support
-- Add ability to purge callers from queue if no agents are logged in