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
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 <darcy at 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:darcy at Vex.Net > VoIP: sip:darcy at Vex.Net-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3577 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161123/6380b723/attachment.bin>
On 23/11/16 13:49, 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. > > 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 <darcy at 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.What are the problems with DTMF and VoIP? http://www.voipmechanic.com/dtmf-issues.htm In some VoIP routes a switch may be configured to detect in-band DTMF which is sent by the VoIP ATA, but then switches to an out of band RFC2833 DTMF required for an upstream provider. This upstream carrier then terminates the call to the PSTN, possibly to a voice mail system, which will require regeneration of the audible inband DTMF tones. The switch has to detect and remove the tone sent by the ATA from the audio stream because the upstream provider specified RFC2833 DTMF. At times the switch can't always completely remove the in-band DTMF tone which is a problem, because by the time it has detected the DTMF tone, it has already passed a short amount of it. This small amount of in-band tone along with the RFC2833 tone sent are both received by the far end voice mail system which will then register an error (problem), possibly an invalid mailbox or invalid password. If this is happening You can set asterisk to use receive the tones inband which if this is occurring might help Trial and error probably, good luck>> >> 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 >> >>-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161123/8f955000/attachment.html>
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/voiceopsI have subscribed. Thanks. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net