It should do the job!
In my setup, I call from an IAX phone to an h323 Gateway, and all is
fine. The opposite direction also works fine.
Though this is an IAX setup, SIP should perform likewise.
REgards,
Hohenzolern
Gary Richardson wrote:
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* <[EMAIL PROTECTED]
<mailto:[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]
<mailto:[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]
<mailto:[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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