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

[Bruno Hertz]

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

> 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>??

Argh, this is really a netmeeting issue. Remember I said 'point your
phone to use asterisk as proxy/gateway'? Now, the question is whether
your client is smart enough to allow that, and if so, how it's done.

I.e. in GM I can set the proxy in the preferences dialog, and then
just dial 1002 with your above example.

Now, I don't use netmeeting myself (and have no Windows installed, for
that matter), but a colleague of mine tells me it should be
configurable via Tools->Options->General->Advanced Calling.

So try setting the gateway there, and if it's configurable simply
dialing 1002 should suffice. If not, I'm afraid Google resp. MS
support might be the only friends left in this matter.

Regards, Bruno.