[asterisk-users] Restricting transfers between SIP phones

[Noah Miller]

>> > So, does anyone know of a way to detect whether a call from a SIP phone
>> > is the first step of an attended transfer or an original call?
>
> It could probably work if you put a SIP proxy in between (ref. Kamilio).

Another way might be to set up a special transfer extension that all
users use to perform transfers.  To do a transfer, all users would
first transfer to that special transfer extension.  The transfer
extension could then read the intended destination and compare the
source and destination in a series of GotoIf statements.  The GotoIf
statements would check the source and destination of the transfer, and
if it's ok, use the transfer() app.  If not, playback a message that
the transfer is not allowed.

It means a lot of very specific dialplan logic, and a change of
procedures for the users, but it's one way to do it.


- Noah