> -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Vladimir Mikhelson > Sent: Saturday, 15 September 2012 5:56 p.m. > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] DTMF digits falsely detected > > > On 9/14/2012 11:04 PM, Matthew Jordan wrote: > > ----- Original Message ----- > >> From: "Vladimir Mikhelson" <v...@mikhelson.com> > >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >> <asterisk-users@lists.digium.com> > >> Sent: Friday, September 14, 2012 10:39:30 PM > >> Subject: Re: [asterisk-users] DTMF digits falsely detected > >> > >> > >> On 9/14/2012 10:11 PM, Matthew Jordan wrote: > >>> ----- Original Message ----- > >>>> From: "Vladimir Mikhelson" <v...@mikhelson.com> > >>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >>>> <asterisk-users@lists.digium.com> > >>>> Sent: Friday, September 14, 2012 9:24:41 PM > >>>> Subject: Re: [asterisk-users] DTMF digits falsely detected > >>>> > >>>> > >>>> On 9/14/2012 6:04 PM, Alec Davis wrote: > >>>>>> -----Original Message----- > >>>>>> From: asterisk-users-boun...@lists.digium.com > >>>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > >>>>>> Vieri > >>>>>> Sent: Saturday, 15 September 2012 8:45 a.m. > >>>>>> To: asterisk-users@lists.digium.com > >>>>>> Subject: [asterisk-users] DTMF digits falsely detected > >>>>>> > >>>> Can it be related to > >>>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ?? > >>>> > >>>> -Vladimir > >>> Most likely not. If the SIP peer is using rfc2833 DTMF, its most > >>> likely related to r370252. > >>> > >>> Please file an issue on the issue tracker, > >>> https://issues.asterisk.org/jira. > >>> Please include a pcap of the RTP stream and a DEBUG log with RTP > >>> debug enabled, using 'rtp set debug on'. > >>> > >>> Thanks, > >>> > >>> -- > >>> Matthew Jordan > >>> > >> Matt, > >> > >> I have created the issue. See > >> > https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedComment > >> Id=197108#comment-197108 > >> > >> Sorry, I will be unable to produce pcap and rtp debug as I > have fixed > >> the issue by uninstalling the Soft Phone I used for multiple years > >> with no issues. > >> > >> -Vladimir > > Well, it'd be appreciated if someone who is experiencing > this would be > > willing to reproduce it and attach a pcap and DEBUG log to > the issue. > > The bug fixed by that commit dealt with out of order DTMF; > I suspect > > that the problem is your soft phone is sending re-transmits > of the end > > event of the DTMF digit with an increasing timestamp. The previous > > behavior in Asterisk would most likely have been more > tolerant of this > > non-compliant scenario, but didn't handle the out of order > packets as > > well. > > > > Unfortunately, without evidence confirming that, there isn't much I > > can do. > > > > -- > > Matthew Jordan > > > Hopefully the initial poster still has the configuration to > produce the files for you. > > Are you saying the DTMF logs I attached do not provide enough > evidence to support the theory of the DTMF length being the > cause of this issue? > > -Vladimir >
Vladimir, What was the Softphone/Version you were using to get this to fail. I'm using an old version of X-Lite, V3.0 build 56125 and with asterisk 1.8.16.0 when in voicemail I was unable to get any errors. DTMF log below. [2012-09-15 22:36:39.974909] DTMF[1706] channel.c: DTMF begin '1' received on SIP/alec-00000009 [2012-09-15 22:36:39.974985] DTMF[1706] channel.c: DTMF begin ignored '1' on SIP/alec-00000009 [2012-09-15 22:36:40.514978] DTMF[1706] channel.c: DTMF end '1' received on SIP/alec-00000009, duration 560 ms [2012-09-15 22:36:40.515037] DTMF[1706] channel.c: DTMF end passthrough '1' on SIP/alec-00000009 [2012-09-15 22:36:41.014955] DTMF[1706] channel.c: DTMF begin '2' received on SIP/alec-00000009 [2012-09-15 22:36:41.015009] DTMF[1706] channel.c: DTMF begin ignored '2' on SIP/alec-00000009 [2012-09-15 22:36:41.459045] DTMF[1706] channel.c: DTMF end '2' received on SIP/alec-00000009, duration 460 ms [2012-09-15 22:36:41.459089] DTMF[1706] channel.c: DTMF end passthrough '2' on SIP/alec-00000009 [2012-09-15 22:36:41.909042] DTMF[1706] channel.c: DTMF begin '3' received on SIP/alec-00000009 [2012-09-15 22:36:41.909093] DTMF[1706] channel.c: DTMF begin ignored '3' on SIP/alec-00000009 [2012-09-15 22:36:42.429177] DTMF[1706] channel.c: DTMF end '3' received on SIP/alec-00000009, duration 540 ms [2012-09-15 22:36:42.429236] DTMF[1706] channel.c: DTMF end passthrough '3' on SIP/alec-00000009 [2012-09-15 22:36:42.849091] DTMF[1706] channel.c: DTMF begin '4' received on SIP/alec-00000009 [2012-09-15 22:36:42.849185] DTMF[1706] channel.c: DTMF begin ignored '4' on SIP/alec-00000009 [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end '4' received on SIP/alec-00000009, duration 1660 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users