9 thoughts on - DTMF Issue

• From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Rees Sent: Monday, July 06, 2015 5:54 PM
  To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF issue

  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

  This doesn’t help, but It DOES sound familiar. I’ve not seen this for a long time. If I can remember I’ll write back. Just thought I’d let you know you’re not crazy. 🙂

• Yes, I have had this annoyance happen to me before. It is very frustrating. In order to rule out the SIP Provider, I suggest you record the call. If the beep is not heard in the recording but only by the end user on the Cisco Phone, then its a phone issue. The phone is confusing audio with the specific frequencies of DTMF. There is little you can do to fix this except for firmware upgrades (and I remember there were some that addressed this specific issue, at least on Cisco ATAs).

• 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@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk@mcts.org

  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

• 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—

• 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@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk@mcts.org

  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—

• 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—

• You probably have to reload asrerisk after making the change.

  Thomas M. Peters | Systems Administrator | tpeters@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk@mcts.org

  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—

• Indeed, thanks. I’ll let you know how it goes. Thanks, Jamie
  —–Original Message—

• 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—