Have a digium TDM400P card which is converting 4 analog lines into the asterisk server. When you call in from another phone line, and try to dial an extension, it doesn't seem to recognize the inputted numbers while the "welcome" message is playing. If you wait until the welcome message has stopped playing, and enter the extension in it works. I believe this is the DTMF?
I have seen this issue with SIP providers before, but not with analog lines.
In your case, because it works in the absence od transmitter audio, I believe you simply have a levels issue.
Have you balanced the hybrid (2 wire to 4 wire circuits on card) the procedure uses a utility called FXO tune and should have been covered in the installation manual of your card.
I believe the FXO tune was run, but since then the server was moved to another location, and has had an echo cancellation module added. I will try re-running the FXO utility and review the manual for the card. Thanks!
After running all of the Digium commands again, including Fxotune the IVR still does not seem to acknowledge numbers pressed on inbound analog calls. (Unless they are pressed during a silent period in the IVR).
Is the echo canceller enabled? Just for a test disable it.
Also have you tried relax DTMF?
Thanks SkykingOH. Adding relaxdtmf=yes to /etc/asterisk/chan_dahdi.conf has resolved the issue. For some reason I only thought this setting was possible for SIP trunks.