Probably because this isn't the way that a lot of other PBX systems work. It's not always easy to educate users about the difference between a blind and attended transfer when the systems that they've used in the past don't make this distinction.
Disconnecting the outside caller certainly doesn't sound like a desirable response if you hang up the transferring extension while the transfer destination is ringing. I would kind of expect it to either degenerate into a blind transfer or return the call to the transferring extension; I can't, offhand, think of any likely situation in which I'd want to summarily hang up on the caller at that point. p. On Sun, 2006-02-12 at 10:57 +0200, Rob Lith wrote: > Why don't you think it is correct behaviour? The purpose of attended > transfer is that you consult with the party before transferring with > hooking, otherwise it would be a blind transfer for which there is a > blind transfer option. > > Rob > > On 2/10/06, Moises Silva <[EMAIL PROTECTED]> wrote: > this is a Normal behaviour, nevertheless i dont think is a > correct behaviour. Several weeks ago other user asked the > same, i suggested him to open a feature request on > bugs.digium.com, check for that > > regards > > > On 2/9/06, Thomas Artner <[EMAIL PROTECTED]> wrote: > Hi! > > I am new with asterisk and I have my first problem > with the attended > call transfer feature. > > When a call comes in, i take the call and i would like > to transfer it. > So I press the * button (mapped for the attended > transfer in > features.conf) and the number for the receiving > extension. > > The receiving extension rings and the call can be > taken there. > So far so good. > > Now to my problem: > If I hook on the handset BEFORE the receiving > extension take the call, > the caller from outside will be disconnected and the > receiving extension > stops ringing. > Shouldn't the receiving extension keep on ringing > until the call is > taken? Independent of hooking on the handset or not! > (as it is with the blind transfer feature) > > The incoming line and all of the extensions are POTS, > connected on a > tdm400p card. > > I use asterisk 1.2.4 and zaptel 1.2.3 > > Hope someone could help me. > > Thx, > Tom > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org"; > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
