[asterisk-dev] DTMF-initiated transfers

[Olle E. Johansson]

26 nov 2009 kl. 15.13 skrev Kai Hoerner:

> Hi,
> 
> Nick Lewis wrote:
>>> Backwards compatibility could be a bit
>>> fun though; the default when not set would have to be
>>> allow_transfer=refer. 
>>> 
>> 
>> I do not understand this comment of yours.
>> Wouldn't the default be allow_transfer=yes (tone&refer)?
>> 
>> 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. 
>> 
> 
> 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).
> 
> This has two benefits over the other way round:
> 
> 1. compatibility with old dialplans .. tT will work anywhere specified
> 2. you can still override this setting for special purposes in dialplan.
>   Example:  When you redirect a call to a mobile phone associated
>             with an extension of your PBX, you may wish to enable
>             transfers for that callee, but not for the whole PSTN
>             trunk.
> 
> IMHO a setting like "transfer=native,dtmf" makes more sense.
> (I find "yes" actually meaning "native & dtmf" is very intransparent)


Well, what do you do if you want to disable all peer configs in the dial() command.
dial(sip/olle,15,-t) 
?

/O