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

[Cesc]

Ok,

I will trust you guys :) But it is puzzling that on another list (SER)
someone (actually 2 people) told me that asterisk would not do the
job. One suggested Yate ... any opinion? (I know ... not nice to ask
in the asterisk list ... but it is good to compare, i think).

Now, just to make sure that if it works for you it does for me ...
what version of asterisk you are running?  :D

Cesc

On 6/16/06, Tigran Kocharyan <tigran_k at nerdshack.com> wrote:
> 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* <cesc.santa at gmail.com
> > <mailto: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
> >     <mailto: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
> >     <mailto: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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