try dtmfmode=inband I noticed that I have issues with that when I was using the phone system and a FXS(O?) card when going over analog phone lines. dtmfmode=inband fixed the issue.
using dial tones strictly over an asterisk system will work just about any mode. However, when crossing into the PSTN, they have zero idea or do not have the ability to see the additional tone telemetry data that is not in the voice/audio streams. ~Benjamin On Fri, 2014-08-29 at 13:12 +0200, nedi wrote: > Hi, > I have problem since one months ago, I can't remember exactly the datum I use > this way to make phone not so frequently. > After retour from Holiday I can't make phone through GSM Gateway SC-385. I > don't change anything > only I made one or two update to newest Astlinux. 1.1.7 > > If I connect trough VPN and Use Sip client there is no problem DTMF are > recognized. > I tried another Gateway and Another provider SIM Card. So I think the problem > coming from Asterisk??? > I use since about 2 years dtmfmode=info in my Astlinux, yesterday i changed > for testing on GSM Gateway and in my sip.conf to Inbound and rfc2833 but > without success. > > I have now Astlinux. 1.1.7 > > I can see this message if I debugging this extension > > on-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 > (nothing), combined - 0x0 (nothing) > > has anyone idea what can be the problem > > is there was new in asterisk and DTMF on last 2 Updates? > > best regards > nedi > > > > > > ------------------------------------------------------------------------------ > Slashdot TV. > Video for Nerds. Stuff that matters. > http://tv.slashdot.org/ > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org.
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