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.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 16, 2009 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul <bulkm...@monafamily.com> 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. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 16, 2009 12:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul <bulkm...@monafamily.com> Can anyone tell me how I can completely move an established call off of one Asterisk server to another? 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. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 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. Thanks for any help _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com
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