[asterisk-dev] DTMF-initiated transfers

[Nick Lewis]

>My understanding is that the Dial() options "t" and "T" should simply 
>override the "allow_transfer=no" (or "=refer") setting with 
>"allow_transfer=tone" (or "=yes", respectively).

I suggest that nothing in the dial plan override allow_transfer=no

For your requested functionality perhaps a sip.conf parameter as follows
would be clearer:
 force_dial_options=tTwWr

N_L
(Your suggested parameter enum of "dtmf" is better than "tone" since
rfc2833 is not really a tone.)

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