[Asterisk-Users] Getting in to h323

[Sathya Weerasooriya]

Greetings,

I am progressing well with this great product, the *.  SIP to SIP calling,
Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also,
purchased couple of FXO cards and did zaptel as well. It's time to get to
h323 now. Read the mailing list for H323 and OH323 etc. need some help to
where to start.

Requirement is very simple, SIP calls need to be routed to a third party
gatekeeper for PSTN termination. I do not want to start a debate on h323 vs
oh323, just want to know the limitations and capabilities.

1) How many simultaneous calls can be made through a h323 driver (I know it
is a limitation of the linux box but I need to know whether these chan
drivers have any such limitation, capacity or something) ?
2) Support of CODEC - Which driver support which codec ?
3) Documentation - I know this is open source, but for a starter which
driver provides better doc support
4) Configuration - Which one is simpler and easier, again any pointers other
than readmes., Wiki and John Todd was my reference points for all the above,
but cant find any h323 stuff there.

Thanks in advance for your help.

Cheers

Sathya