I have a sip trunk between two asterisk boxes. I can call into the first box, hit 499 for example and the call goes to the second box and answers as expected plays me audio message just fine etc... My issue is that DTMF does not seem to be working.
Both sides are set for: dtmfmode=RFC2833 What might I look at as to why DTMF digits are not transferred? Thanks, Jerry
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
