[asterisk-users] DTMF issue

Tom Peters TPeters at mcts.org
Tue Jul 7 14:44:56 CDT 2015 

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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