[asterisk-users] DTMF issue
Jamie Rees
jrees at gmlnt.com
Fri Jul 24 05:16:59 CDT 2015
Hi all,
Just an update here....
I added the relaxdtmf=yes in sip_custom.conf, given according to the
documentation the default option is no. This has made a bit of difference,
I'm getting less reports of it now although one particular person seems to
still be affected (they talk to a particular person who has a distinctive
voice)
All phones are Cisco SPA512G, are set to G711u/ulaw codec, DTMF process
AVT=yes and DTMF TX Method: Auto. I tried InBand, amongst others and that
did nothing.
Where some DTMF bursts are lower than 80ms (which is the lowest Asterisk
expects), it's triggering emulation to bring it in line. I've read that
reducing the minimum DTMF tone length to 40ms can solve this issue, by
editing the #define AST_MIN_DTMF_DURATION variable.
Does anyone concur? If so, where can I find said variable in the config
files?
Thanks again,
Jamie
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jamie Rees
Sent: 08 July 2015 10:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
Indeed, thanks.
I'll let you know how it goes.
Thanks,
Jamie
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 22:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
You probably have to reload asrerisk after making the change.
Thomas M. Peters | Systems Administrator | tpeters at mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org
>>> "Jamie Rees" <jrees at gmlnt.com> 7/7/2015 3:53 PM >>>
Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.
I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.
Thanks,
Jamie
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport.
See this discussion, which I found after I posted my first response:
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence:
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."
The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that?
Thomas M. Peters | Systems Administrator | tpeters at mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org
>>> "Jamie Rees" <jrees at gmlnt.com> 7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).
Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.
My dahdi-channels.conf file looks stock:
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63
; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63
Thanks again,
Jamie
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] DTMF issue
It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off.
I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself.
We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting)
Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension.
So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.
-T
Thomas M. Peters | Systems Administrator | tpeters at mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org
>>> "Jamie Rees" <jrees at gmlnt.com> 7/6/2015 4:53 PM >>>
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance.
I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.
I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:
[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-0000c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-0000c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209-00021cac
Can someone please provide any tips?
Thanks,
Jamie
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