Yes, I second to that idea.
I think thats only available option to put them in a
local conference.
Rgds
Manoj K Gupta

----- Original Message -----
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 2:04 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hi Foong,
>
> But then... who and when will trigger the transfer
between the two remote
> extensions?
>
> I think to something like that.
> One of the extension calls a special number,
entering a password (or check
> after the Caller ID).
> Asterisk close the call, wait for answer
> Call the second extension, wait for answer
> Then, in some way (eventually through a conference
mode using local
CONSOLE
> as master) bridge the two calls.
> What do you think about that?
>
> Dan
>
>
> ----- Original Message -----
> From: "Chee Foong" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 11:30 AM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Hello Dan,
> >
> > Thanks for you reply.
> >
> > Base on you recomendation using the 'T' argument.
I manage to do call
> > transfer an it works really well.
> >
> > My problem comes when my boss comes out with a
superb idea where the
> > transfering process is automated without involving
a human :(
> >
> > Say asterisk get 2 numbers (from database, text
file, etc), one belongs
> > party A and the other belongs to party B. Asterisk
will calls both
parties
> > and do the tranfer automatically. In another
words, asterisk is
resposible
> > to 'press' the '#' to do the transfer. I don't
this can be achieve in
the
> > extension.conf not matter how you structure you
dial plan.
> >
> > Perhaps, the only way is to write a apps and plug
it into asterisk like
> all
> > the asterisk modules such as Meetme.
> >
> > Any ideas?
> >
> >
> > Foong
> >
> > ----- Original Message -----
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 3:42 PM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hi,
> > >
> > > It works if you put the 'T' switch in the dial
line.
> > >
> > > You can then transfer the call from the caller.
> > > I have tested it in the folllowing configuration
and it works:
> > > Call from a Cisco 7960 to an ATA 186.
> > > Select 'Transfer" on 7960
> > > Call another extension (X-Lite)
> > > Select again transfer on 7960.
> > > The call remain between ATA and X-Lite.
> > >
> > > This is what you need?
> > >
> > > BR,
> > > Dan
> > >
> > > ----- Original Message -----
> > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > Subject: [Asterisk-Users] Call Transfer
> > >
> > >
> > > Hello all,
> > >
> > > I am in a situation where I need to use asterisk
to call someone say
> Party
> > > A. After the call to Party A got through,
asterisk will put Party A on
> > hold,
> > > then asterisk will call Party B. If call to
Party B got through,
> asterisk
> > > will transfer Party A to Party B.
> > >
> > > I wonder if this features is implemented into
asterisk. I have found a
> > post
> > > in asterisk mailing list:
> > >
http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> > >
> > > but that doesn't help much.
> > >
> > > If this features is not implemented, can anyone
give me some point on
> how
> > to
> > > implement this in asterisk? Do I need to write
an app like the Dial
apps
> > for
> > > asterisk to load at start up?
> > >
> > >
> > > thanks
> > >
> > > Foong
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
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