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

[Bryan Field-Elliot]

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.

Thank you,

Bryan

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