[asterisk-dev] DTMF-initiated transfers

[Benny Amorsen]

One of the really ancient features of Asterisk is that you can do
transfers with DTMF -- as long as t or T is set in Dial() or Queue() as
appropriate. Quite nice, as far as it goes. However, this model doesn't
really fit usage at IP Vision, and the impedance mismatch is getting worse.

Our customers have 3 different kinds of needs:

1) SIP phones, where DTMF-initiated transfers should not be allowed
(those phones have perfectly good buttons, no need to fake it)

2) Mobile phones and some DECT phones where DTMF-initiated transfers are
necessary, because they can't do something smarter.

3) Outgoing lines where t or T must never be set.

Right now we're detecting whether the caller is type 1, 2 or 3 and
setting T as appropriate, and then doing the same for the callee. This
is quite a bit of code (involving fun with Local channels), and it still
breaks in this case:

If a Pickup is done using the *8 feature, the phone who did the pickup
will be able to transfer the call using DTMF if the ringing phone
happened to be a mobile phone (which is wrong if the phone doing the
Pickup is a SIP phone), and the opposite problem happens if a mobile
phone does a Pickup from a ringing SIP phone.

This would be a lot easier if the tT options were replaced with an
option linked to the SIP peer/user: allowdtmftransfer=yes/no. Better
names for the option are more than welcome, of course. We could set
allowdtmftransfer=no for peers connecting to the outside and as long as
we made sure to never use tT option (could be checked statically), we'd
know that no error in the dial plan could result in outside users being
able to transfer calls.

For backwards compatibility and for those with more complicated needs,
the tT options could be kept around.


/Benny