Re: [asterisk-users] Random DTMF Tones Only on heard on ATA
Thanks Sherwood for all the info. The devices are using ulaw and rfc2833. There is no transcoding on my server, but not sure what my trunk providers are doing. I was thinking about the frequency detection issue as it seems to be primarily involving women so I'll try adjusting the input/output gain to see if that helps. I've tried all other combinations of DTMF settings on the ATAs so that's my last hope on the device end. I'm hoping to avoid the packet capture as that is never a fun road to go down but that is probably the next step. Any idea why Asterisk still ques/replays the tone considering it see's it's shorter than the 80s minimum? Travis Hoping to avoid On Wed, Jul 28, 2010 at 7:16 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Sorry, I came into this late...what codec is the device using, and is the audio being trascoded? Back at Voxitas, we had a couple of customers complain about random DTMF tones coming across their line, and Asterisk WAS actually hearing DTMF tones...want to know what it was?. In that particular case (just a place to start looking) it was G729 on customer ATAs (don't remember the models)Here's the freaky thingIt only happened with CERTAIN people talking on the phone...IIRC, we determined that the ATA's G729 processor was mistaking certain audio frequencies in the speaker's voice and believing it was a DTMF tone from the analog device and sending the appropriate DTMF signal to our servers... I'm sorry, I don't remember how we fixed it...I think we did some audio tweaking (advanced ATA config, input level, out level, etc..), be we may have just ended up having to tell that client to not use G729 on those ATAs This _MAY_ happen with other codecs, but I think it's mainly either G729..maybe primarily transcoding? NERDY FUn Crap below: capture SIP and RTP between your Asterisk and an offending device (writing to a file)then start doing everything you can to cause the DTMF issue to occur. NOW, open your capture in wireshark...dump the RTP payload to a file and open that file in an audio editor Now, go through the wireshark capture...see if you see any DTMF events (if rfc2833 it'll be an RTP EVENT, if SIP INFO, it'll be a sip info, and if you're using inband **SHUDDER** you can just listen to the audio).note the time in seconds from the beginning of the audio stream whenever a DTMF event occurs, and then go to that spot in the audio fileIf you're feeling REALLY frisky, do a frequency analysis...I'll bet you'll see that the voice that is speaking at the time of the DTMF event on your various captures will have a frequency range in common...a very close range...maybe look up DTMF tone definition and get the freqs(did itmore detail than even I feel like doing right now :D) Cheers, Sherwood McGowan On Wed, Jul 28, 2010 at 6:43 PM, Travis Langhals tra...@netitek.com wrote: SIP/5211 is a Grandstream device. Did not add relaxdtmf=no, but sip show settings verifies it's already set to no. Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 Travis On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Random DTMF Tones Only on heard on ATA
On 7/29/2010 6:51 PM, Travis Langhals wrote: Thanks Sherwood for all the info. The devices are using ulaw and rfc2833. There is no transcoding on my server, but not sure what my trunk providers are doing. I was thinking about the frequency detection issue as it seems to be primarily involving women so I'll try adjusting the input/output gain to see if that helps. I've tried all other combinations of DTMF settings on the ATAs so that's my last hope on the device end. I'm hoping to avoid the packet capture as that is never a fun road to go down but that is probably the next step. Any idea why Asterisk still ques/replays the tone considering it see's it's shorter than the 80s minimum? Travis Hoping to avoid Travis, Let me start off by saying if you're experiencing this primarily with women, this is DEFINTELY the talk off issue I was talking about... As far as why Asterisk plays the DTMF tones even though they're too short.. that's probably because you have relaxdtmf on? I don't know for sure, I'd try the devs for that :) Cheers! -- _ -- 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
Re: [asterisk-users] Random DTMF Tones Only on heard on ATA
Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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
Re: [asterisk-users] Random DTMF Tones Only on heard on ATA
SIP/5211 is a Grandstream device. Did not add relaxdtmf=no, but sip show settings verifies it's already set to no. Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 Travis On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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
Re: [asterisk-users] Random DTMF Tones Only on heard on ATA
Sorry, I came into this late...what codec is the device using, and is the audio being trascoded? Back at Voxitas, we had a couple of customers complain about random DTMF tones coming across their line, and Asterisk WAS actually hearing DTMF tones...want to know what it was?. In that particular case (just a place to start looking) it was G729 on customer ATAs (don't remember the models)Here's the freaky thingIt only happened with CERTAIN people talking on the phone...IIRC, we determined that the ATA's G729 processor was mistaking certain audio frequencies in the speaker's voice and believing it was a DTMF tone from the analog device and sending the appropriate DTMF signal to our servers... I'm sorry, I don't remember how we fixed it...I think we did some audio tweaking (advanced ATA config, input level, out level, etc..), be we may have just ended up having to tell that client to not use G729 on those ATAs This _MAY_ happen with other codecs, but I think it's mainly either G729..maybe primarily transcoding? NERDY FUn Crap below: capture SIP and RTP between your Asterisk and an offending device (writing to a file)then start doing everything you can to cause the DTMF issue to occur. NOW, open your capture in wireshark...dump the RTP payload to a file and open that file in an audio editor Now, go through the wireshark capture...see if you see any DTMF events (if rfc2833 it'll be an RTP EVENT, if SIP INFO, it'll be a sip info, and if you're using inband **SHUDDER** you can just listen to the audio).note the time in seconds from the beginning of the audio stream whenever a DTMF event occurs, and then go to that spot in the audio fileIf you're feeling REALLY frisky, do a frequency analysis...I'll bet you'll see that the voice that is speaking at the time of the DTMF event on your various captures will have a frequency range in common...a very close range...maybe look up DTMF tone definition and get the freqs(did itmore detail than even I feel like doing right now :D) Cheers, Sherwood McGowan On Wed, Jul 28, 2010 at 6:43 PM, Travis Langhals tra...@netitek.com wrote: SIP/5211 is a Grandstream device. Did not add relaxdtmf=no, but sip show settings verifies it's already set to no. Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 Travis On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen benny+use...@amorsen.dk wrote: Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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 -- _ -- 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
