[asterisk-users] No DTMF on Sip Trunk?

[Jared Smith]

On Thu, 2008-04-24 at 12:02 -0400, Noah Miller wrote:
> For the first time, I'm setting up SIP trunking between two asterisk
> boxes.  The calls themselves work fine, but I'm not able to get DTMF
> working.  

If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it
appears that you are), you'll need to set "rfc2833compensate=yes" in the
peer or friend section of sip.conf on the Asterisk 1.4 box.  

This tells Asterisk to send RFC2833 DTMF the way that Asterisk 1.2
expects it, instead of the newer (read: more standards compliant) way
that Asterisk 1.4 now handles RFC2833 DTMF tones.

In a nutshell, try adding "rfc2833compensate=yes" to your section named
[129trunk551] on the box you're calling Asterisk2.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.