[Asterisk-Dev] DTMF when transferring calls - not handled well?

[Olle E. Johansson]

Bryan Field-Elliot wrote:
> I am using the current release of Asterisk (1.0.7).
> 
> When either a SIP user or a IAX2 user is transferred to a SIP extension,
> DTMF doesn't seem to be handled very well. Specifically, when the caller
> presses DTMF digits (even if they hold them down), the recipient of the
> call (SIP) only hears the briefest DTMF pulse imaginable (e.g. 50
> milliseconds or so).
> 
> Is this a bug in Asterisk? When the caller holds down a DTMF digit, I
> need the callee to hear the full duration of the tone.
> 
> We are using RFC2833 everywhere.
> 
I don't think Asterisk handles the duration of the DTMF very well.
In chan_sip the duration is hard coded in the source code for INFO DTMF.

/O