[Asterisk-Users] sip to h323 gateway ...

[Cesc]

Hi,

I am familiar with asterisk, though never actually tinkered with one
myself ... so i don't know the full extent of its capabilities.

I am facing a request to bridge a sip network and an h323 network.
I would like to operate the sip with ser as the proxy and some
gatekeeper on the h323 side (not required though).
Actually, i have a few more points that may make it simpler
- i do not need codec negotiation: both sides are configured use
the same (g711 alaw) by default.
- I have just a few "phones" on each side, so even "static routing"
can work, if that is of any help.
- it is not a production environment, for now. It is a demo/lab

The question is ... can asterisk do the job?

Ideally, the bridge would be only signalling-wise (rtp to be direct
end-to-end). But, if someone had bad experience with this and would
recommend to use a B2BUA approach, please, tell me.

I don't know if it makes a difference, but most of the calls would go
from the H323 side to the SIP side ... but i don't really want to
restrict SIP->H323.

Thanks a lot!

Cesc