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<div>Who sends the 500 failure code? Asterisk or the VOIP supplier through which you got the call? Note that Asterisk has the basic mechanism for call trasnsfer, just as you transfer a call, so the problem is either in using Transfer() inside IVR context, or the provider.</div>
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<div>As David noted - use canreinvite=yes</div>
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<div>And last word: If you get the error from the local Asterisk then raise the verbosity level - probably you'll find some hint there.</div>
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<div> Good luck, __Yehavi:</div>
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<div>2009/1/17 David fire <span dir="ltr"><<a href="mailto:ddfire@gmail.com">ddfire@gmail.com</a>></span></div>
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<div>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" target="_blank">lenz.loway@gmail.com</a>></span>
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<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>
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<p>Thanks<br></p>
<p>l.</p>
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<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>
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<div align="left"><span><font face="Arial" color="#0000ff" size="2">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>
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<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
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<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>
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<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>
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<div 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>
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<div 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>
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<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>
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<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>
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<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>
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<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>
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<div><span><font face="Arial" size="2">In our lab we are testing with 1.4.22.1</font></span></div>
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<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>
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<div><span><font face="Arial" size="2">Thanks for any help</font></span></div></font></div>
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