[Asterisk-Users] Call Transfer

[Sip Rtp]

Hi
I would like to further ask if it is possible to
transfer a call from
openphone to pstn. i.e. i use openphone and asterisk
-oh323 channel driver
to make a call to a PSTN number through zap channel
connected on that
end.Then i wanna transfer that PSTN number to some
other openphone
extension/alias
May i have a look at your extension to conf, as i am
not clear with how to
implement this.

Rgds
Manoj k Gupta




----- Original Message -----
From: "Chee Foong" <cheefoong at inovas.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 2:00 PM
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
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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