Are you sure that the TRANSFER is supported by the other side at all? see http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267
Thanks l. 2009/1/16 Paul <bulkm...@monafamily.com> > 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 > > > _______________________________________________ > -- 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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