diff options
author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2000-01-07 10:54:40 +0000 |
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committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2000-01-07 10:54:40 +0000 |
commit | e4b9f6a3431bc99c454ddda3a845629e832b7226 (patch) | |
tree | ed26e276bfaf05fc50c295d18015f65e3a1af2ad /CHANGES | |
parent | 075980cea1ff2f36ae0372bb6db0de5c8168b090 (diff) |
Version 0.1.2 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rwxr-xr-x | CHANGES | 15 |
1 files changed, 15 insertions, 0 deletions
@@ -1,3 +1,18 @@ +* 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 |