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

Lenz Emilitri lenz.loway at gmail.com
Fri Jan 16 12:09:00 CST 2009


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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090116/07d70192/attachment.htm 


More information about the asterisk-users mailing list