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 <[EMAIL PROTECTED]> 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 <[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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