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

[Greg Hill]

On Fri, 17 Dec 2004, Brent Goran wrote:

> 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?

Depends on whether you're using dtmfmode= inband, rfc2833, or info.
Inband is sent, of course, "in-band" (as audio). g711 is the first step
toward success for inband DTMF. Evidently the other codecs work okay for
voice, but really make a mess of DTMF to the point where it's not likely
to even work. I believe the other two dtmfmode's actually send DTMF as SIP
notify messages, so the codec would be irrelevant and the DTMF should
always arrive fine.. so long as the endpoints do a good job of detecting
and regenerating the tones.

One thing which some have found helps with picky IVRs is to set the
endpoints to use dtmfmode=inband and set Asterisk to use dtmfmode=rfc2833
(or use the sipsetdtmfmode app, or whatever its name is, in the exten.conf
to change the setting on a per-call basis). This way Asterisk doesn't
notice the DTMF tones being passed inband and doesn't try to intercept and
regenerate them.

Greg