What is  the best way to debug DTMF on a PJSIP trunk?  I have cycled through all available options ('rfc4733','inband','info','auto','auto_info') but my IVR does not recognize any options from the remote end. I have also tried changing codecs from g729 to alaw or ulaw with the same result.  Outgoing calls do not seem to have this problem, just incoming.  This is with Asterisk 13.29.2 but the problem started with 13.21 before I decided to upgrade to the latest 13.x version.  Any pointers?

-- 
_____________________________________________________________________
-- 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

Reply via email to