and canreinvite=yes ?
2009/1/17 Lenz Emilitri <lenz.lo...@gmail.com> > 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 > > > _______________________________________________ > -- 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 > -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination.
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