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