On Tue, Jul 23, 2019, at 1:47 PM, Jerry Geis wrote: > 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,
"rtp set debug on" will show the RTP traffic flowing, and thus the DTMF. The "dtmf" option to the logger can also be used to provide a log message when DTMF is received. This can be used to narrow it down. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
