[Asterisk-Users] Optimizing Sipura/Asterisk for DTMF?

[Brent Goran]

We have an application which is primarily DTMF driven (automated on both
sides), which we are trying to deploy over VOIP and Asterisk (using some
Sipuras and some IAXY's).

We are finding that in around half the cases, the Asterisk server can't
decode the DTMF digits from the field office (or at least some of them).
Though, when we place voice calls for testing, we can hear eachother
quite well.

I was wondering if there are any settings in Asterisk and/or in SIP
clients such as the Sipuras, which will optimize the connections for
DTMF rather than voice?

Thank you in advance,

Brent