[asterisk-users] Attended Transfer

[Eric "ManxPower" Wieling]

Arlen Nascimento wrote:
> Henry, according with voip-info.org, attended transfer is
> "While on conversation with another party, you dial the atxfer key
> sequence. Asterisk says "Transfer" then gives you a dial tone, while
> putting the other party on hold. You dial the transferee number and
> talk with the transferee to introduce the call, then you can hang up
> and the other party will be connected with the transferee. In case the
> transferee does not want to answer the call, he/she simply hangs up
> and you will be back to your original conversation."
> The callee is put on hold "automatically"
> 
> Eric, attended transfer is only possible with an ATA??

Attended transfer is supported by every decent SIP device out there.  It 
is a basic phone feature.  There are a few SIP devices out there that do 
NOT support attended transfer but I would not call them "decent."  The 
GS BT101 and the FREE version of X-Ten's phone are both devices that do 
not support attended transfer.

There are a couple of reasons to want to do "DTMF Transfers" (configured 
in Asterisk via /etc/asterisk/features.conf.  One reason might be that 
you are stuck, for some reason, with a phone that does not support 
attended transfer.  Another reason would be if you have several 
different types of phones and ATAs around and do not want to make users 
learn different ways to do a transfer, depending on the phone the person 
is using at the moment.  Another reason, and one I think is the most 
common, is that you simply don't know any better.