2 -- Hold lock when creating new H.323 channel to sync the audio channels
3 -- Decrement usage counter when appropriate
4 -- Actually unregister everything in unload_module
5 -- Add IP based authentication using 'host'in type=user's
7 -- Intergration into the mainline Asterisk codebase
8 -- Remove reduandant debug info
9 -- Add Caller*id support
11 -- Retool port usage (to avoid possible seg fault condition)
13 -- Configurable support for user-input (DTMF)
14 -- Reworked Gatekeeper support
15 -- Native bridging (but is still broken, help!)
16 -- Locally implement a non-broken G.723.1 Capability
17 -- Utilize the cleaner RTP method implemented by Mark
18 -- AllowGkRouted, thanks to Panny from http://hotlinks.co.uk
19 -- Clened up inbound call flow
20 -- Prefix, E.164 and Gateway support
21 -- Multi-homed support
24 -- Added H.323 Alias support
25 -- Clened up inbound call flow
26 -- Fixed RTP port logic
27 -- Stomped on possible seg fault conditions thanks to Iain Stevenson
29 -- Fixed one-way audio on inbound calls. Found
30 race condition in monitor thread.
33 -- Changed name to chan_h323
34 -- Also renamed file names to futher avoid confusion
37 -- First public offering
38 -- removed most hardcoded values
39 -- lots of changes to alias/exension operation
42 -- initial build, lots of hardcoded crap
43 -- Proof of concept for External RTP