[Asterisk-Users] SIP <-> PSTN DTMF

[John Goerzen]

Hi,

I have a SIP phone connecting to my asterisk server, using
dtmfmode=rfc2833.  When calling from the SIP phone to internal asterisk
services, such as voicemail, it works fine.

But when I call out to the PSTN, from the SIP phone, via my X100P, the
call will be connected fine.  After that, though, any numbers I dial on
the SIP phone are lost.  I hear them on the phone, but I don't hear them
on the remote end of the PSTN connection.

I know that rfc2833 is correct for the SIP phone since it is working
fine with internal asterisk services.

I have tried the wiki, searching the list, and google.  No luck.  Ideas
would be welcome!

-- John