[Asterisk-Users] Call Transfer

[Chee Foong]

Dan,

Asterisk is suppose to trigger the transfer when it successfully call both
extensions
Do you mean I have to create conference room for every call? that would not
be practicle.

Or do you have a example dialplan to to illustrate you suggestion?

Actually, we have a client that is too lazy to do all the dialing, he want a
system that will call him and also the person he wanted to call, just like
some receptionists do theese days. The different is that asterisk is taking
over the receptionist's job

thanks


Foong


> 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" <cheefoong at inovas.com>
> To: <asterisk-users at lists.digium.com>
> 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" <dtoma at fx.ro>
> > To: <asterisk-users at lists.digium.com>
> > 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" <cheefoong at inovas.com>
> > > To: <asterisk-users at lists.digium.com>
> > > 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
> > >
> > >
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> > >
> >
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> >
> >
>
>
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