[asterisk-users] how to transfer calls when analog phone hasnotransfer button

[Erick Perez]

On 1/5/07, Doug Crompton <doug at crompton.com> wrote:
> Well it would be interesting to know what FXS device you are using to
> connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could
> bypass Asterisk and connect the FXO to FXS or dial directly if it were so
> configured, so reinvite would work but wwould probably not be desired but
> that is not the question...

Right, I forgot to mention that.
Plain an simple analog phones will be connected to audiocodes
fxs-to-sip and then the audiocodes talk to asterisk.
im planning *not* to use transcoding and go full g711 ulaw on this one.

>
> I am using the SPA-3000 as both an FXO (connection to telco) and FXS
> (connection to my house analog phones) with Asterisk in between. I have
> said this before on here but I will say it again. With the SPA-3000 you
> cannot have analog phone feature keys, transfer etc. AND still be able to
> use DTMF for control outside of the dialplan.
>
> If you want feature key control then you would use rfc2833 DTMF, if you
> want to be able to use DTMF incoming or outgoing for control then you must
> use inband DTMF. It is either/or.
>
> My choice was to use inband and not have features selected for the analog
> phones. To often I would use these phines with banking or on incoming to
> control voicemail functions so I wanted that capability.
>
> In that case a hook flash works fine. If you have never done it just flash
> the hook for a second (or use the flash key on the phone) and you will get
> another dialtone. Then you can call another party, tell them you have a
> call to transfer and hangup or click again and bring them in as a
> conference.
>
> Doug
>
>
> On Fri, 5 Jan 2007, Don Pobanz wrote:
>
> > > Erick Perez
> > >
> > > Don, I suppose that in order for this to work i need
> > > canreinvite=no, right?
> > >
> >
> > No! It doesn't matter what you have for 'canreinvite' since
> > 'canreinvite' is a SIP attribute, not an analog phone attribute.
> > For analog phones, Asterisk will always be in the call path. :-)
> >
> > --
> > Don Pobanz
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>
>
> "Those that sacrifice essential liberty to obtain a little temporary safety
>  deserve neither liberty nor safety."  -- Ben Franklin (1759)
>
> ****************************
> *  Doug Crompton           *
> *  Richboro, PA 18954      *
> *  215-431-6307            *
> *                          *
> * doug at crompton.com        *
> * http://www.crompton.com  *
> ****************************
>
>
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-- 
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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