and canreinvite=yes ?<br><br><br><div class="gmail_quote">2009/1/17 Lenz Emilitri <span dir="ltr"><<a href="mailto:lenz.loway@gmail.com">lenz.loway@gmail.com</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;">
<p>Are you sure that the TRANSFER is supported by the other side at all? see <a href="http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267" target="_blank">http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267</a></p>
<div><div></div><div class="Wj3C7c">
<p>Thanks<br></p><p>l.</p><p></p><br><div class="gmail_quote">2009/1/16 Paul <span dir="ltr"><<a href="mailto:bulkmail@monafamily.com" target="_blank">bulkmail@monafamily.com</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>
<div dir="ltr" align="left"><span><font color="#0000ff" size="2" face="Arial">Yes, this is the first method I tried. The transfer
only works if it is done before a media path is set up to the first box (not
answered by the IVR). If it is answered then transferred, I get a 500
internal server error back from the ITSP and the call dies. I never see
anything hit the second box.</font></span></div>
<div dir="ltr" align="left"> </div>
<div dir="ltr" align="left"> </div><br>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><div><b>From:</b> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Lenz
Emilitri<br></div><b>Sent:</b> Friday, January 16, 2009 10:09 AM<div><div><br><b>To:</b>
Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re:
[asterisk-users] How to transfer a call from one AsteriskServerto
another<br></div></div></font><br></div><div><div>
<p>I guess you already tried this? </p>
<p><a href="http://www.voip-info.org/wiki-Asterisk+cmd+Transfer" target="_blank">http://www.voip-info.org/wiki-Asterisk+cmd+Transfer</a></p>
<p>Thanks</p>
<p>l.</p>
<p><br></p><br>
<div class="gmail_quote">2009/1/16 Paul <span dir="ltr"><<a href="mailto:bulkmail@monafamily.com" target="_blank">bulkmail@monafamily.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>
<div dir="ltr" align="left"><span><font color="#0000ff" size="2" face="Arial">I do have
it functioning with Dial(). I was looking for a way to completely
move the call from the first box though. When using Dial() media moves,
but the call is still tied to the first box. In looking at captures when
the call is ended, the first box invites out to the ITSP again, then after
receiving a 200ok sends a bye.</font></span></div>
<div dir="ltr" align="left"> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" size="2" face="Arial">Also while
testing, once the call was up on the second box, I stopped Asterisk on the
first box which kills the call.</font></span></div>
<div dir="ltr" align="left"> </div>
<div dir="ltr" align="left"> </div><br>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><b>From:</b> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of
</b>Lenz Emilitri<br><b>Sent:</b> Friday, January 16, 2009 12:17
AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion<br><b>Subject:</b> Re: [asterisk-users] How to transfer a call from
one AsteriskServer to another<br></font><br></div>
<div>
<div>
<p>Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?</p>
<p>l.<br></p><br>
<div class="gmail_quote">2009/1/16 Paul <span dir="ltr"><<a href="mailto:bulkmail@monafamily.com" target="_blank">bulkmail@monafamily.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>
<div><font size="2" face="Arial">
<div><span><font size="2" face="Arial">Can anyone tell me how I can completely
move an established call off of one Asterisk server to
another?</font></span></div>
<div> </div>
<div><span><font size="2" face="Arial">In our case we have a server with our
IVR. Depending upon digits entered, the call can be transferred to any
of our other servers depending where the extension or queue
reside.</font></span></div>
<div><span><font size="2" face="Arial">We would like to completely move the call
off of the first box so we don't tie up resources on it.</font></span></div>
<div> </div>
<div><span><font size="2" face="Arial">In our lab we are testing with
1.4.22.1</font></span></div>
<div> </div>
<div><span><font size="2" face="Arial">Our provider which delivers inbound calls
to us uses a Sonus gateway. So far, testing has shown that if we
transfer the inbound call prior to any media playback, it works. But,
if the IVR plays media, then it is failing, with a 500 internal server error
being returned.</font></span></div>
<div> </div>
<div><span><font size="2" face="Arial">Thanks for any
help</font></span></div></font></div>
<div> </div>
<div align="left"> </div>
<div> </div></div><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></blockquote>
</div><br><br clear="all"><br>-- <br>Loway - home of QueueMetrics - <a href="http://queuemetrics.com" target="_blank">http://queuemetrics.com</a><br><br></div></div></div><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></blockquote></div><br>
<br clear="all"><br>-- <br>Loway - home of QueueMetrics - <a href="http://queuemetrics.com" target="_blank">http://queuemetrics.com</a><br><br></div></div></div>
<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></blockquote></div><br><br clear="all"><br>-- <br>Loway - home of QueueMetrics - <a href="http://queuemetrics.com" target="_blank">http://queuemetrics.com</a><br>
<br>
</div></div><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></blockquote></div><br><br clear="all"><br>-- <br> (\__/) <br>(='.'=)This is Bunny. Copy and paste bunny into your <br>
(")_(")signature to help him gain world domination. <br><br>