Jakob Hirsch wrote:
Hello everyone!
Hola,
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.
Generally DTMF is something that has to be configured on both sides, you can't just configure it on one and have the negotiation force it to be that.
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.
Unfortunately there isn't a way to force this as you describe out of the box, you would have to make changes to chan_sip or explicitly have the clients configured properly.
Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
