[Asterisk-Dev] chan_h323 dtmf

[Kelvin Chua]

in reference to bug# 1414, 

what would you achieve by setting the f.samples in chan_h323.c to 800?
is this the dtmf on-time? off-time? in milliseconds? 

there's something weird about dtmf in our scenario (doesn't exist in sip
only in chan_h323), after pressing # to transfer a call, it only accepts
2 or less digits depending on the speed on which you press the digits.
the thing is, dtmf functions in voicemail is all ok.