[asterisk-dev] DTMF-initiated transfers

[Benny Amorsen]

"Nick Lewis" <Nick.Lewis at atltelecom.com> writes:

> I must be misunderstanding the proposal. I assumed tone 
> transfers would be enabled only if the call had tT 
> options set and sip.conf had allow_transfer=tone. If 
> the default is allow_transfer=yes then tone transfers 
> would be controlled by the tT options as currently. 

Most of the point of my proposal is to avoid the tT option on calls
completely. It is of course possible to do it the other way around; I
can simply make a search/replace on all Dial()'s to add the tT option.
To me it seems like the wrong way around; when allow_transfer=tone I
would expect tone transfers to be allowed without having to do anything
more.


/Benny