Modify DTMF Length for outgoing calls


I want to modify the default DMTF length when asterisk send a call to the PSTN network, there is a bug of asterisk identified at: and

I have used the value toneduration in chan_dahdi.conf without success, so I want to modify the DEFAULT_DTMF_LENGTH parameter but I can’t find it in this version of asterisk 1.8 and FREEPBX 2.8.1

I must say that the call origination is from SIP phones or ATA’s, Asterisk receives the entire phone number in one go, so there is no need of a tone duration or a delay between digits when dialing.

If someone can help me…