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 <cesc.santa@gmail.com> 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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