"Joe S" <[EMAIL PROTECTED]> 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. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
