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" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]>
To: [email protected]
Subject: Re: [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper
Date: Mon, 11 Apr 2005 17:03:32 +0200
"Joe S" <[EMAIL PROTECTED]> 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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