Hi, you could try switching the DTMF mode of the ATA's SIP peer (and also in the ATA itself) to INBAND transmission. In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk can interpret it. For this to work, the ATA needs to use a G.711 codec. Inband DTMF needs an uncompressed codec to work properly.
Another way is (if the ATA supports it) to switch DMTF mode to SIP INFO. In this mode, DTMF is not interpreted out of the audio stream. For external peers which are not supporting this mode Asterisk then generates the proper RTP messages or tones. With SIP INFO mode I made my best results with all devices, sadly it's not very common used. Max Am 23.11.2016 um 20:02 schrieb D'Arcy Cain: > On 2016-11-22 07:49 PM, Pete Mundy wrote: >> >> One direction that may be worth exploring further is his ATA's config (or >> perhaps swapping it for a different model). Eg adjusting echo cancellation >> or line impedance settings. > > I have to be careful here as I auto-provison these devices and changes would > propogate to every user. Echo cancellation is off. Do you think it should > be on? > >> Is the ATA he is using the same as the ATA you use? > > No but it is the same as other users who do not have the problem. I use a > SIP phone and a Cisco ATA. > >> Failure to correctly recognise and decode DTMF is just one of many reasons >> why I never use them (ATAs). Like faxing over VoIP, they're just too much >> trouble :( > > I understand but some use cases just need it. > >> Genuine IP phones are pretty good value these days. Could you drop one of >> those on-site as a temporary measure to prove that it's phone and/or ATA >> related? > > He does want to have an extension so that won't work. > >> Ps, you might also want to consider joining VoiceOps (if you're not already >> subscribed) and posting there. >> https://puck.nether.net/mailman/listinfo/voiceops > > I have subscribed. Thanks. >
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