[Asterisk-Users] Call Transfer

[Dan]

There is no need to create a Meeting Room... just to initiate a conference
in three...

----- Original Message ----- 
From: "Chee Foong" <cheefoong at inovas.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 1:02 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hello
>
> But If i do that I have to create lots of conference room if I have lots
of
> caller.
>
> Foong
>
> ----- Original Message -----
> From: "Sip Rtp" <vovida2001 at yahoo.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, July 30, 2003 5:44 PM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > 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" <dtoma at fx.ro>
> > To: <asterisk-users at lists.digium.com>
> > 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" <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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