[asterisk-dev] DTMF-initiated transfers
Kai Hoerner
kai at ciphron.de
Thu Nov 26 08:13:47 CST 2009
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)
Regards,
Kaii
More information about the asterisk-dev
mailing list