[asterisk-users] Touch tone stutter
darcy at vex.net
Tue Nov 22 17:16:17 CST 2016
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:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net
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