Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information? Thanks! -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/7cfe4298/attachment.htm
Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information? Thanks! -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/c42f8629/attachment.htm
Why not SCP some of the sample MOH files to a PC then open it up in Winamp or what have you, determine the bitrate then do your MP3's the same? hth -----Original Message----- From: Justin Selleck [mailto:jselleck@smoothfusion.com] Sent: Wednesday, September 28, 2005 12:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music on Hold Quality Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information? Thanks! -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/a83f9f07/attachment.htm
I was on hold at Digium and noticed they had that EXACT problem. It was REAL bad on the one occasion I was on hold with them a couple weeks ago. If Digium has it then there must be some inherent issues with Asterisk that need to be worked out. I personally think this should be given a HIGH priority. Just my opinion as I am not a coder and cannot contribute in that way. _____ From: Justin Selleck [mailto:jselleck@smoothfusion.com] Sent: Wednesday, September 28, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music on Hold Quality Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information? Thanks! -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/b3cedb99/attachment.htm
Music on hold audio should be resampled to 8 bits 16 Khz mono and preencoded, so audio distortion is minimized in Asgerisk encoding. This theory has worked form me on other commercial platforms, but not yet on Asterisk, because MP3s cannot be resampled that way. If anyone figures it out, please advice. Regards, Jorge Alay?n -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de Matt Enviado el: Mi?rcoles, 28 de Septiembre de 2005 06:42 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Music on Hold Quality I have heard this issue when on hold with Cisco and Vonage... I don't think it's an asterisk problem I htink it's a G711 "problem"... or gsm "problem". Basically they are made for voice, and I think the music goes outside their encoding ranges... sound logical? On 9/28/05, canuck15 <canuck15@hotmail.com> wrote:> > > I was on hold at Digium and noticed they had that EXACT problem. It was > REAL bad on the one occasion I was on hold with them a couple weeks ago. If > Digium has it then there must be some inherent issues with Asterisk that > need to be worked out. I personally think this should be given a HIGH > priority. > > Just my opinion as I am not a coder and cannot contribute in that way. > > > ________________________________ > From: Justin Selleck [mailto:jselleck@smoothfusion.com] > Sent: Wednesday, September 28, 2005 11:19 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Music on Hold Quality > > > > > > Does anyone know how to maximize music on hold quality on calls inbound from > PSTN? I know that it is common to have choppy and static sounding music on > hold when connecting via PSTN but how can that be minimized? I assume that > the bitrates, type of music, etc can minimize the effects. Does anyone have > any experience in this area? Do you know where I should look for more > information? > > > > Thanks! > > > > -Justin > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
My IVR prompts I create by mixing together the voiceover with an imported music MP3 in Audacity. I use the tools in Audacity to remove the hiss, normalize the audio, and bump up the low end (that really helps) I save it as CD-quality stereo then downsample it to 16 bit, 8Khz mono with good old sndrec32.exe in Windows. Copy it over to Asterisk and sox it as GSM files, and the sound is perfect, no skips, hiss or anything like that. The system is under light load (maybe 5-15 calls concurrent at any one point) , however, so I can't comment on if it's an Asterisk thing for it to skip under heavy load or what. Works for me, though. -----Original Message----- From: Jorge Alayon [mailto:j.alayon@ses.com.ar] Sent: Wednesday, September 28, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music on Hold Quality Music on hold audio should be resampled to 8 bits 16 Khz mono and preencoded, so audio distortion is minimized in Asgerisk encoding. This theory has worked form me on other commercial platforms, but not yet on Asterisk, because MP3s cannot be resampled that way. If anyone figures it out, please advice. Regards, Jorge Alay?n -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de Matt Enviado el: Mi?rcoles, 28 de Septiembre de 2005 06:42 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Music on Hold Quality I have heard this issue when on hold with Cisco and Vonage... I don't think it's an asterisk problem I htink it's a G711 "problem"... or gsm "problem". Basically they are made for voice, and I think the music goes outside their encoding ranges... sound logical? On 9/28/05, canuck15 <canuck15@hotmail.com> wrote:> > > I was on hold at Digium and noticed they had that EXACT problem. It was > REAL bad on the one occasion I was on hold with them a couple weeks ago.If> Digium has it then there must be some inherent issues with Asterisk that > need to be worked out. I personally think this should be given a HIGH > priority. > > Just my opinion as I am not a coder and cannot contribute in that way. > > > ________________________________ > From: Justin Selleck [mailto:jselleck@smoothfusion.com] > Sent: Wednesday, September 28, 2005 11:19 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Music on Hold Quality > > > > > > Does anyone know how to maximize music on hold quality on calls inboundfrom> PSTN? I know that it is common to have choppy and static sounding musicon> hold when connecting via PSTN but how can that be minimized? I assumethat> the bitrates, type of music, etc can minimize the effects. Does anyonehave> any experience in this area? Do you know where I should look for more > information? > > > > Thanks! > > > > -Justin > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Wed, September 28, 2005 5:41 pm, Matt said:> I have heard this issue when on hold with Cisco and Vonage... Idon't> think it's an asterisk problem I htink it's a G711 "problem"... orgsm> "problem". Basically they are made for voice, and I think > the music goes outside their encoding ranges... sound logical?Rolling off the high-end of the audio range above 4 KHz helps. Try madplay instead of mpg123 and also have the playback gain reduced ~12 db with this musiconhold.conf line: default => custom:/usr/local/lib/asterisk/mohmp3/,/usr/local/bin/madplay --mono --sample-rate=8000 --attenuate=-12 --output=raw:- -kim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050929/e3276327/attachment.htm