2c65582b66
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
44 lines
1.4 KiB
Text
44 lines
1.4 KiB
Text
Build
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-- Hold lock when creating new H.323 channel to sync the audio channels
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-- Decrement usage counter when appropriate
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-- Actually unregister everything in unload_module
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-- Add IP based authentication using 'host'in type=user's
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0.1.0
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-- Intergration into the mainline Asterisk codebase
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-- Remove reduandant debug info
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-- Add Caller*id support
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-- Inband DTMF
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-- Retool port usage (to avoid possible seg fault condition)
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0.0.6
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-- Configurable support for user-input (DTMF)
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-- Reworked Gatekeeper support
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-- Native bridging (but is still broken, help!)
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-- Locally implement a non-broken G.723.1 Capability
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-- Utilize the cleaner RTP method implemented by Mark
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-- AllowGkRouted, thanks to Panny from http://hotlinks.co.uk
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-- Clened up inbound call flow
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-- Prefix, E.164 and Gateway support
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-- Multi-homed support
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-- Killed more seg's
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0.0.5
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-- Added H.323 Alias support
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-- Clened up inbound call flow
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-- Fixed RTP port logic
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-- Stomped on possible seg fault conditions thanks to Iain Stevenson
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0.0.4
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-- Fixed one-way audio on inbound calls. Found
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race condition in monitor thread.
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0.0.3
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-- Changed name to chan_h323
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-- Also renamed file names to futher avoid confusion
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0.0.2
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-- First public offering
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-- removed most hardcoded values
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-- lots of changes to alias/exension operation
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0.0.1
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-- initial build, lots of hardcoded crap
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-- Proof of concept for External RTP
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