<p>I guess you already tried this? </p><p><a href="http://www.voip-info.org/wiki-Asterisk+cmd+Transfer">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">bulkmail@monafamily.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">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 face="Arial" color="#0000ff" size="2">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 lang="en-us" dir="ltr" align="left">
<hr>
<font face="Tahoma" size="2"><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 class="Wj3C7c">
<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="padding-left:1ex;margin:0px 0px 0px 0.8ex;border-left:#ccc 1px solid">
<div>
<div><font face="Arial" size="2">
<div><span><font face="Arial" size="2">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 face="Arial" size="2">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 face="Arial" size="2">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 face="Arial" size="2">In our lab we are testing with
1.4.22.1</font></span></div>
<div> </div>
<div><span><font face="Arial" size="2">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 face="Arial" size="2">Thanks for any
help</font></span></div></font></div>
<div> </div>
<div align="left"> </div>
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