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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
28 lines
1.1 KiB
Plaintext
Executable File
28 lines
1.1 KiB
Plaintext
Executable File
* Asterisk 0.1.2
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-- Updated README file with a "Getting Started" section
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-- Added sample sounds and configuration files.
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-- Added LPC10 very low bandwidth (low quality) compression
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-- Enhanced translation selection mechanism.
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-- Enhanced IAX jitter buffer, improved reliability
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-- Support echo cancelation on PhoneJack
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-- Updated PhoneJack driver to std. Telephony interface
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-- Added app_echo for evaluating VoIP latency
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-- Added app_system to execute arbitrary programs
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-- Updated sample configuration files
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-- Added OSS channel driver (full duplex only)
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-- Added IAX implementation
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-- Fixed some deadlocks.
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-- A whole bunch of bug fixes
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* Asterisk 0.1.1
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-- Revised translator, fixed some general race conditions throughout *
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-- Made dialer somewhat more aware of incompatible voice channels
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-- Added Voice Modem driver and A/Open Modem Driver stub
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-- Added MP3 decoder channel
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-- Added Microsoft WAV49 support
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-- Revised License -- Pure GPL, nothing else
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-- Modified Copyright statement since code is still currently owned by author
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-- Added RAW GSM headerless data format
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-- Innumerable bug fixes
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* Asterisk 0.1.0
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-- Initial Release
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