asterisk/channels/h323/ChangeLog

Build
	-- Hold lock when creating new H.323 channel to sync the audio channels
	-- Decrement usage counter when appropriate
	-- Actually unregister everything in unload_module
	-- Add IP based authentication using 'host'in type=user's
0.1.0
	-- Intergration into the mainline Asterisk codebase
	-- Remove reduandant debug info
	-- Add Caller*id support
	-- Inband DTMF
	-- Retool port usage (to avoid possible seg fault condition)
0.0.6
	-- Configurable support for user-input (DTMF)
	-- Reworked Gatekeeper support
	-- Native bridging (but is still broken, help!)
	-- Locally implement a non-broken G.723.1 Capability
	-- Utilize the cleaner RTP method implemented by Mark
	-- AllowGkRouted, thanks to Panny from http://hotlinks.co.uk
	-- Clened up inbound call flow
	-- Prefix, E.164 and Gateway support
	-- Multi-homed support
	-- Killed more seg's 
0.0.5
	-- Added H.323 Alias support
	-- Clened up inbound call flow
	-- Fixed RTP port logic
	-- Stomped on possible seg fault conditions thanks to Iain Stevenson 
0.0.4
        -- Fixed one-way audio on inbound calls. Found
           race condition in monitor thread.

0.0.3
        -- Changed name to chan_h323
        -- Also renamed file names to futher avoid confusion

0.0.2
        -- First public offering
        -- removed most hardcoded values
        -- lots of changes to alias/exension operation

0.0.1
        -- initial build, lots of hardcoded crap
        -- Proof of concept for External RTP