Thanks a lot guys. Exactly what I needed.<div><br></div><div>Best regards,</div><div>Örn<br><br><div class="gmail_quote">On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net">oej@edvina.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
26 jan 2010 kl. 16.48 skrev Örn Arnarson:<br>
<div><div></div><div class="h5"><br>
> Hi guys,<br>
><br>
> I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method.<br>
> Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt).<br>
><br>
> Using a timeout in the Dial command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent).<br>
><br>
> Can anyone think of a way to get the call back to the transferrer after this timeout?<br>
><br>
</div></div>THe transferred call is sent to a context set with the channel variable TRANSFER_CONTEXT before you call DIAL().<br>
<br>
In there, run DUMPCHAN to see which variables you have and then dial with a timeout. After the timeout, dial back.<br>
<br>
/O<br>
<font color="#888888">--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</font></blockquote></div><br></div>