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

[Joe S]

Hi Bruno,
Thanks I appreciate your help its really working, I just dial 1002 for NM, 
and Xlite is ringing.
Joe.

>"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.
>