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. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
