[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation

[Brian Wilkins]

>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