[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation
Peter Brown
peterabrown at froggy.com.au
Thu Sep 30 15:22:59 MST 2004
Hi guys,
Is there a repository of documentation from presentations at Astracon
2004.
I would like to read about the Performance and Scalability study.
TIA,
Peter
On Thu, 2004-09-30 at 19:33, Brian Wilkins wrote:
> >From my experience, I wouldn't recommend sending h323 calls through Asterisk
> because the channel drivers appear broke. You can also reference the
> Performance and Scalability study presented at AstriCon 04.
>
> > Hello,
> >
> > --- UTRINI at embratel.com.br wrote:
> > <snip>
> >
> > > Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator?
> > > I want to implement PC-to-Phone calls in the following topology (for
> > > signalling):
> > > SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 --->
> > > PSTN
> > > The RTP audio packets would go direct through Softphone to gateway.
> >
> > You can use Asterisk as a SIP-H323 translator. It is not a SIP proxy,
> > but a PBX having a SIP channel. It also is a SIP UAS/Registrar. I
> > dont think when it is used as a translator, RTP packets will go
> > directly from softphone to gateway, since there are 2 different
> > protocols involved. Asterisk will force the RTP packets to go
> > through it.
> >
> > >
> > > Helaine
> > >
> >
> > Regards, Girish
> >
> >
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>
> --
> Brian Wilkins
> brian at hcc.net
> Heritage Communications Corporation
> Melbourne, FL USA 32935
>
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