[Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not getting?

[Martin Joseph]

Hi again,

I am trying to get my DTMF to use RFC 2833 rather then inband, so that 
I can utilize lower bandwidth codecs through my FXO.

After much tinkering I was able to get my gateway (wellgate 3701A) 
configured to a point where I have some success,  but no real joy.

I have configured the RTP Payload type (or RFC2833 Payload type) to 
101.  I don't have a clue what this means,  but I took the 101 from my 
AG168V ATA's configuration screen, as I know that device seemed to work 
fine through the old HT-488 fxo(via rfc2833).

I then changed my asterisk extensions for both the FXS and FXO on the 
wellgate to include dtmfmode=rfc2833.

This has brought me to a point where both my hardphones (ATA's) and my 
softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail.

To me this means that asterisk is properly getting the RFC2833 events 
from the user agents.

BUT, if I try to dial out the FXO, none of my phones (hard or soft) 
produce working touchtones for a PSTN based voicemail system.

Even stranger to me, is the fact that from the phone connected to the 
FXS on the wellgate I can hear tones(listening on a called line), but 
they sound kind "rough" at the edges.  From the AG168V  I hear no 
tones,  but what seems to be "blown out" tones (ie overdriven volume).  
 From the IAX softphones I hear no tones at all just clicks!

Now I would have guessed that the FXO would be doing the conversion of 
the RFC2833 to inband, so that I thought all the tones should sound the 
same from any phone?  Apparently this isn't the case at all.

Thanks to all of you for any help understanding and or debugging this 
mess.

Marty

PS I spent a good deal of time adjusting the DTMF volume for the 
wellgate FXS/FXO hoping this might help before I discovered the variety 
of non working DTMF being generated.