[Asterisk-Users] SIP-H323 translation

[voipses at gmail.com]

Hello,

I would like to find an appropriate solution for SIP to H323 translation
(vice versa would be great too!), in an environment where there's going to
be 100+ concurrent calls: has anyone succesfully implemented such a
translator/gateway, e.g. using Opal+OpenH323/Asterisk or anything else?

Any idea of the requisites or issues that could be faced?

Thank you!

Tim
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