[asterisk-users] h323 - SIP conversion

[Michelle Dupuis]

I'm looking at setting up an asterisk box dedicated to SIP<->H323 conversion
(a 3rd party is currently converting the protocols for us).
 
1. Is it worthwhile to split this functionality onto a second server?  Or
should we let the ast pbx handle the conversion?  (we have a couple hundred
active channels to convert)
2. Is it better to go direct from SIP to AIX?
2. Can Asterisk handle H323 natively with problem?
 
Thanks,
MD
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