Hello everyone! We use Asterisk for various services like voicemail. Our SIP clients usually use rtp events (rfc2833) for DTMF, which works just fine and independent from the codec (g711 vs. g726 etc.).
Now we noticed there are some SIP clients that announce telephone-event in their SDP, but send their DTMF inband. The problem with that is, that Asterisk obviously does not try to detect inband DTMF after seeing the telephone-event payload type in the SDP. So we are in a kind of dilemma: - dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for the described ones. - dtmfmode=inband would also work for most, but of course not for the ones using g726 et al. Is there any Asterisk setting to force inband DTMF detection (with non-compressing codecs only, of course)? I browsed the code without result. Does anybody have a hint how to handle this? Or if the SIP clients behaviour is even RFC compliant? Regards and TIA, Jakob -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users