<br><br><div class="gmail_quote">2009/3/8 Sven Geggus <span dir="ltr"><<a href="mailto:usenet@fuchsschwanzdomain.de">usenet@fuchsschwanzdomain.de</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">Gavin Henry <<a href="mailto:gavin.henry@gmail.com">gavin.henry@gmail.com</a>> wrote:<br>
<br>
> Just transfer them to your meetme extension after you've called them.<br>
<br>
</div>Hm, how would I do this? Until now call switching usually ended for me when<br>
the call has been established.<br>
<br>
I'm using a SIP phone connected to an asterisk box which is connected to the<br>
world via various ways (ISDN, SIP, IAX2).<br>
<br>
So what would I do on the my SIP phone after the call has been<br>
established and what needs to be changed in the dialplan to actually<br>
reconnect the current call to the MeetMe Conference then?<br>
<br>
Sven<br>
<font color="#888888"><br>
</font></blockquote><div>You need to transfer option enabled in dial() (tT)<br><br>CLI > core show application Dial <br></div></div>And you need to press a transfer button ;)<br>