[asterisk-users] Simple Meetme Question

[Sven Geggus]

Gavin Henry <gavin.henry at gmail.com> wrote:

> Just transfer them to your meetme extension after you've called them.

Hm, how would I do this? Until now call switching usually ended for me when
the call has been established.

I'm using a SIP phone connected to an asterisk box which is connected to the
world via various ways (ISDN, SIP, IAX2).

So what would I do on the my SIP phone after the call has been
established and what needs to be changed in the dialplan to actually
reconnect the current call to the MeetMe Conference then?

Sven

-- 
The main thing to note is that when you choose open source you don't
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