[Asterisk-Users] Call Transfer Announcement

[Steven Critchfield]

On Fri, 2003-02-28 at 11:03, Matthew S. Hill wrote:
> Steven Critchfield wrote:
> 
> >On Fri, 2003-02-28 at 09:52, Matthew S. Hill wrote:
> >  
> >
> >>Does anybody know how to announce a transfer with asterisk?
> >>
> >>Example:
> >>
> >>    Call is established between A and B. A then wants to transfer B to C.
> >>    On many switches when A transfers B to C, first B is placed on hold,
> >>    then A and C are bridged, and A announces to C that B would like to
> >>    talk to them. A then Hangs up and B and C are brigded.
> >>    
> >>
> >
> >Call established between A and B. A wants to transfer B to C. A flash
> >hooks the phone, dials extension, or number to get connected to C and
> >announces intent to transfer. A then flash hooks again, and proceeds to
> >introduce C and B, then bows out of the conversation by hanging up. This
> >method uses the 3 way calling function.
> >
> >Call established between A and B. A wants to transfer B to C. A flah
> >hooks the phone, dials extension 700 and listens for number to be
> >repeated. Then A hangs up phone, then places call to C, and announces
> >call. At this point A and C have handed off the call. C then can dial
> >the number that was read to A and be connected to B. This uses the call
> >parking function of asterisk. It is beneficial if C needs a moment to
> >compose before picking up the line and A doesn't need to be tied up.
> >There is also a variation of this where A parks the call, then goes to a
> >new location where A dials the parked location of the call and is able
> >to retrieve the call at a new location. Beneficial if you are not at
> >your desk, but still was routed the call.
> >
> >  
> >
> Ok, that will work on a 1 to 1 ratio with analog phones. How do you do 
> it with sip
> phones? The # transfer is nice, but it has no call announcement feature. 
> I can park
> calls, but the users, don't want to park and dial again.

If you are using SIP phones, then it is a phone feature, not a asterisk
feature to get the job done. Even if the origination of the audio is the
asterisk server, and the other destination user is accessed via the
asterisk menu, you are doing a SIP to SIP transfer.
-- 
Steven Critchfield  <critch at basesys.com>