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.
Is the ATA he is using the same as the ATA you use? 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 :( 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? Pete 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 > On 23/11/2016, at 12:16 pm, D'Arcy Cain <da...@vex.net> wrote: > > I am hoping someone else has seen this and can offer a solution or at least a > direction to investigate. I am running 11.23. Most of my clients are fine > but one has a strange behaviour. He has a Grandstream HT701 like most of my > clients who use an ATA. He can make call and they are crystal clear. > However, when he tries to use phone menus ("dial 234 for John Doe" for > example) it doesn't work. At first I thought that the tones were not being > delivered but I had him play them to me and the issue is that each tone > stutters. As a result, entering "234" becomes "223344" which is not > understood as a valid input. > > He has a recent phone and, in fact, is almost the same model I have at home. > His is a Panasonix TX-TGD220 and mine is a TX-TGD-212. The difference is > mainly that his has a built in answering machine. > > As I said, no one else is having the issue. One person has a horrible > connection with voice drops all the time but the touch tones still work. > > I have made two files available. http://darcy.vex.net/Bishop.ogg is an OGG > file of the sound that it makes at the receiving end and the other at > http://darcy.vex.net/Bishop.png is a picture of the wave form. I had the > user think "one Mississippi" etc. and alternately press and release three > different buttons. I recorded off my SIP phone which is going through the > same Asterisk server as the user. > > The only thing I can see in my configs that might apply is in sip.conf > "dtmfmode=rfc2833" which I don't want to change unless I am absolutely sure. > No one else is having the problem so I don't want to risk it. Would I be safe > if I set it to "auto"? Is there any chance that it would help? Is there > some place else I should be looking? > > Thanks in advance for any help. > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:da...@vex.net > VoIP: sip:da...@vex.net
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