On Wed, Sep 26, 2018, at 10:25 AM, Floimair Florian wrote: > Hey all! > > I recently tried the dtmf_mode "auto_info" on my setup to support > endpoints that only understand SIP INFO as a fallback. > > My setup is the following: > > Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) > > Both are configured with "auto_info" dtmf_mode in pjsip.conf. > What I ran into is, that DTMF sent from endpoint A to endpoint B is > additionally sent via inband audio on the RTP stream from Asterisk to > endpoint B, as one can clearly hear the DTMF tone in the audio stream, > when a capture is played back on Wireshark. On the leg from endpoint A > to Asterisk there is no inband DTMF signal in the RTP audio stream. > > Can someone confirm this behavior? If yes than this is clearly a bug. > I had a look in the code which introduced this feature and couldn't find > anything obvious why this is happening.
Have you bumped up the core debug to see what's going on underneath? There will be information about whether it is really generating the DTMF in the core, and if so then it'd be a result of the digit_begin function of chan_pjsip returning a value it shouldn't. -- Joshua 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 -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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
