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

[Gary Richardson]

Nope, asterisk does the bridging. Asterisk can talk to SIP phones and H323
gateways/phones. It can also cross connect them.

Since I have SIP users plugged into asterisk, I have a dial plan that looks
something like:

exten => 100,1,Macro(local_sip_user,SIP/bill)
exten => 101,1,Macro(local_sip_user,SIP/bob)
exten => 102,1,Macro(local_sip_user,SIP/steve)
exten => _XXX,1,Macro(call_ccm,${EXTEN})
exten => _8XXX,1,Macro(call_ccm,${EXTEN:1})

So, if you dial 100-102, you get a sip call, but if you dial 103, it would
try to dial my CCM. If you dial 8100, it would call CCM anyway.

>From the cisco side, I have some similar logic. That's pretty much it.

On 6/15/06, Cesc <cesc.santa at gmail.com> wrote:
>
> 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 <gary.richardson at gmail.com> 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 <cesc.santa at 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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