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 To :  yate@v...
 From :  Paul Chitescu <paulc@v...>
 Subject :  Re: [yate] H.323 Endpoint to SIP Registrar
 Date :  Wed, 15 Mar 2006 19:39:02 +0200 (EET)
Hi, Timothy!

I have no idea if your phone needs to register to a gatekeeper or you just
needs to have the Yate machine configured as its gateway. You have to
check the documentation.

A simple regexroute.conf like this will do sip <-> h.323 routing:


[default]
; the H.323 handset trying to call out
${module}^h323$=sip/${called};line=asterisk_account;caller=handset_number
; we may check the number but let's just assume it's for the handset
${module}^sip$=h323/user@h...
; oops! who's calling?
.*=;error=forbidden;reason=Where are you calling from?


Paul Chitescu


On Wed, 15 Mar 2006, Timothy Creswick wrote:
> Hi,
> 
> I've spent about 2 hours looking through the FAQ and Archive but I could
> still use a little help.
> 
> I presently have 3 SIP phones on my LAN connected to an account with a
> remote SIP registrar on the WAN. These are all traversing NAT, and working
> fine (not using Yate).
> 
> I also now have a H.323 protocol hard-phone, which I would like to enable on
> my system. The SIP registrar is running asterisk, but they don't (and wont)
> support H.323. I gather I need to run a proxy such as Yate on my LAN as a
> signalling gateway.
> 
> Do I also need to configure Yate as a gatekeeper in order to connect the
> H.323 handset?
> 
> I have managed to get Yate to connect to the SIP account successfully, but
> how to I configure it to forward all calls from the SIP side to the H323 and
> vice-versa?
> 
> Many thanks,
> 
> Tim