[Asterisk-Users] Call Transfer

[Chee Foong]

Hi Sip,

I achieve that by adding the following extension into extension.conf:

exten => _9XXXXXXXX,1,Dial(H323/{EXTEN:1})

foong

----- Original Message ----- 
From: "Sip Rtp" <vovida2001 at yahoo.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 5:45 PM
Subject: Re: [Asterisk-Users] Call Transfer


> 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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