Phantom DTMF issue

Asterisk Asterisk Support
1

Hi guys, i have an issue regarding phantom dtmf. I have two system with same asterisk versione (1.8.10.1) that show the same issue.
users says to me that sometimes calls are dropped, so i have enabled debug logging. Analizying the log file i saw that frequently i have:

[Sep 28 09:43:00] DTMF[24168] channel.c: DTMF end '0' received on SIP/5202-000031e7, duration 150 ms
[Sep 28 09:43:00] DTMF[24168] channel.c: DTMF end accepted with begin '0' on SIP/5202-000031e7
[Sep 28 09:43:00] DTMF[24168] channel.c: DTMF end passthrough '0' on SIP/5202-000031e7
[Sep 28 09:43:01] DTMF[24168] channel.c: DTMF begin '0' received on SIP/5202-000031e7
[Sep 28 09:43:01] DTMF[24168] channel.c: DTMF begin passthrough '0' on SIP/5202-000031e7
[Sep 28 09:43:01] DTMF[24168] channel.c: DTMF end '0' received on SIP/5202-000031e7, duration 150 ms
[Sep 28 09:43:01] DTMF[24168] channel.c: DTMF end accepted with begin '0' on SIP/5202-000031e7
[Sep 28 09:43:01] DTMF[24168] channel.c: DTMF end passthrough '0' on SIP/5202-000031e7
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF begin 'A' received on IAX2/local-device-1346
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF begin passthrough 'A' on IAX2/local-device-1346
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF end 'A' received on IAX2/local-device-1346, duration 0 ms
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF end accepted with begin 'A' on IAX2/local-device-1346
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF end 'A' detected to have actual duration 51 on the wire, emulation will be triggered on IAX2/local-device-1346
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF end 'A' has duration 51 but want minimum 80, emulating on IAX2/local-device-1346
[Sep 28 10:11:36] DTMF[24212] channel.c: DTMF end emulation of 'A' queued on IAX2/local-device-1346
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF begin '*' received on SIP/trunk-pri1-00003246
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF begin passthrough '*' on SIP/trunk-pri1-00003246
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF end '*' received on SIP/trunk-pri1-00003246, duration 40 ms
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF end accepted with begin '*' on SIP/trunk-pri1-00003246
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF end '*' detected to have actual duration 39 on the wire, emulation will be triggered on SIP/trunk-pri1-00003246
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF end '*' has duration 39 but want minimum 80, emulating on SIP/trunk-pri1-00003246
[Sep 28 10:23:18] DTMF[24260] channel.c: DTMF end emulation of '*' queued on SIP/trunk-pri1-00003246

I am really sure that users didn’t press ‘A’ or ‘*’ and also most duration says ‘0 ms’ that is quite impossibile.

i do some test and i hav verified that in same case, when the log talk about ‘letter’ dtmf (A,B,C,D) the call is dropped.

I googled a lot and i have found some issues regarding DTMF in 1.8.10 but only talk about duplication of dtmf.

2

Is there an A to D conversion in the DTMF path? If so, this looks like talk off, i.e. the decoder is misrecognizing the voice as a digit. If so, you need to attack it at the point where the decode is happening.

There is not enough information to identify that in your initial report.

3

Hi david55, thank you for your reply… i didn’t understund very well your answer… i try to better explain with the simplest test case that show the issue: a sip user with hard phone (in my case i have one system with all yealink phones, on the other system all cisco 3911 phones), the asterisk box and a patton pri gateway.

On several calls in the debug log there are strange dtmf detection that i am really sure that users didn’t use at all.

I made several wireshark capture and there are any dtmf from sip user towards the asterisk box.

i hope that you or anyone other than you can give me at least some suggestion on where look for.

the asterisk box is configured with dtmfmode=rfc2833 and if i try voicemail login or other services that need dtmf all works well.

Thank you in advance.

4

teleflow.org/wiki/index.php5?tit … es_or_DTMF

5

Hi

I have seen the 4 a-d dtmf digits show up in calls , this has always been when the users are using handsfree, either the Asterisk user or the remote party

As to capturing in wire shark if using RFC2833 I think they are displayed , If you also capture the rtp stream you may be able to hear something as well