So, asterisk does the bridging ... I asked on another list and the
answer was that asterisk could not do the job :O
The truth is that my setup should be fairly simple ... i do not need
any "cool" feature (voicemail and the like). I just need to call from
one side to the other, for a reduced amount of users (so name mapping
could even be manual ... no problem).

Cesc

On 6/15/06, Gary Richardson <[EMAIL PROTECTED]> wrote:
I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP
setup. It works. There are issues, but that has more to do with Unity
voicemail than the h323 implementations.


 On 6/15/06, Cesc <[EMAIL PROTECTED]> wrote:
>
 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
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