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

[Joe S]

Hi Bruno,
Thanks for the input, one question. Let's say I define context=default in my 
oh323.conf.

Then, in my extensiions.conf I have:
[default]

exten=>1002, 1, Dial(SIP/1002)            ; 1001 is an Xlite SIP UA

so how do I call a sip user like from NetMeeting, is it like 
1002@<ip_address_of_gateway>??

Thanks,

Joe


>From: "Bruno Hertz" <brrhtz at yahoo.de>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] From OH323  to SIP or OH323 without 
>gatekeeper
>Date: Mon, 11 Apr 2005 17:03:32 +0200
>
>"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.
>
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