[Asterisk-Dev] SIP DTMF question

[asterisk at thehowertons.net]

I've been lurking for quite a while looking at all the SIP/DTMF
information passing through this forum.  My problem is after dialing a
number with the SIP client (XLite or SJPhone) no tones are recognize
(sent?) to the far end.  For instance, I call the demo at digium and try
to select 2 for support and it does not select anything.  What do I need
to have in my setup?  Here is my current sip.conf setup for the phone.

[phone1]
type=friend
host=dynamic
username=ryan at internal.thehowertons.net
secret=password
dtmfmode=inband; Choices are inband, rfc2833, or info
mailbox=9725 ; Mailbox for message waiting indicator
context=home
callerid="Ryan V. Howerton" <9725>

Thanks for the help.

Ryan