[asterisk-users] Simple Meetme Question

[Grygoriy Dobrovolskyy]

2009/3/8 Sven Geggus <usenet at fuchsschwanzdomain.de>

> 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
>
> You need to transfer option enabled in dial()   (tT)

CLI > core show application Dial
And you need to press a transfer button ;)
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