[Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper

[Bruno Hertz]

"Joe S" <printingfoot at hotmail.com> writes:

> Hi,
>
> I am new with asterisk. I was wondering if there is a way to call a
> OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
> default protocol without having a gatekeeper.
>
> I can make a call from SIP to OH323 by specifying it in the
> extensions.conf file, like:
>
> exten=>1001, 1, Dial(OH323/10.10.10.1)
>
> so I was wondering if there was a way to call from OH323 to SIP or OH323.

Sure. Just specify in oh323.conf the context where incoming calls
should go. That context then can include dial statements for any
protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to
setup dial plans.

Finally, instruct your H323 phone to use asterisk as a gateway
resp. proxy, not a gatekeeper. Any calls will then go through
asterisk, and to the context you specified.

I'm doing that with Gnomemeeting all the time, and it works without
problems.

Regards, Bruno.