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Version 0.1.2 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Mark Spencer
2000-01-07 10:54:40 +00:00
parent 9440db13f4
commit 5800db6556
6 changed files with 208 additions and 957 deletions
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* Asterisk 0.1.2
-- Updated README file with a "Getting Started" section
-- Added sample sounds and configuration files.
-- Added LPC10 very low bandwidth (low quality) compression
-- Enhanced translation selection mechanism.
-- Enhanced IAX jitter buffer, improved reliability
-- Support echo cancelation on PhoneJack
-- Updated PhoneJack driver to std. Telephony interface
-- Added app_echo for evaluating VoIP latency
-- Added app_system to execute arbitrary programs
-- Updated sample configuration files
-- Added OSS channel driver (full duplex only)
-- Added IAX implementation
-- Fixed some deadlocks.
-- A whole bunch of bug fixes
* Asterisk 0.1.1
-- Revised translator, fixed some general race conditions throughout *
-- Made dialer somewhat more aware of incompatible voice channels