[Asterisk-Users] SIP dtmf=rfc2833 not working

[Brent Goran]

We are testing some DTMF-driven applications over VOIP (legacy systems
which use fast pulses of standard DTMF tones).

The applications work fine when Digium IAXy's are used - no loss or
garbling of DTMF tones.

However, when we use SIP modems (such as Sipura 1000's), the DTMF tones
are frequently uninterpretable and our applications have to ask for
retries.

I am under the impression that the IAXy is digitizing DTMF tones and
sending just the "pure" data, rather than the audio representation, and
that this explains why the IAXY's work flawlessly in this application.

I am also under the impression that SIP modems should also support a
mode like this.. We have tried:

dtmfmode=rfc2833

in "sip.conf", and we have also tried turning on "DTMF Tx:" to "AVT" on
the Sipura, but this does not affect reliability at all.

So my question is:

1) Are we doing anything wrong, or is there something more we should be
doing, to enable DTMF translation (ala rfc2833) in Asterisk and/or our
SIP modems?

2) Is there any kind of debugging mode in Asterisk which we can turn on,
which will show once and for all whether or not we really have
successfully enabled rfc2833?

We are using Asterisk 1.0.3, by the way.

Thank you very much in advance!

Brent