[asterisk-users] How to transfer a call from one AsteriskServerto another

[Yehavi Bourvine]

 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.

As David noted - use canreinvite=yes

And last word: If you get the error from the local Asterisk then raise the
verbosity level - probably you'll find some hint there.

                                  Good luck, __Yehavi:


2009/1/17 David fire <ddfire at gmail.com>

> and canreinvite=yes ?
>
>
> 2009/1/17 Lenz Emilitri <lenz.loway at 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 <bulkmail at 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-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at 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 <bulkmail at 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-bounces at lists.digium.com [mailto:
>>>> asterisk-users-bounces at 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 <bulkmail at 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.
>
>
> _______________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090118/4b1801af/attachment.htm