Re: [Asterisk-Users] feature - VM gain adjust?
Hello I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Think about a dynamic sound compressor that would possibly auto-adjust. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Hello That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Think about a dynamic sound compressor that would possibly auto-adjust. Are you suggesting such a thing exists, or that that would be a proposed future application? Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. -Seth On Mon, 2004-07-12 at 08:46, Rich Adamson wrote: Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Hmmm... I don't know if playing with the * code would really be the best here... Although if it was a plug-in app like app_volume or something I guess it couldn't hurt... It really sounds like you have a line issue here. You said that adjusting the gain on your card introduced echo issues. It sounds like you have an impedance mismatch/imbalance. Like your telco is trying to cut corners going from a 4-pair to 2-pair or doing some creative splitting... Do you possibly know where the source of the echo might be coming from? Maybe somewhere under your control? If not it can be a pain getting the telco to acknowledge/fix the problem. Most proprietary PBXs even would have this problem, although they usually don't introduce so much attenuation as your FXO card seems to be doing... I know I know * is way better than a PBX and it should be more flexible. I'm just saying that normally there's no way short of getting the damn telco to fix the problem or getting your own ISDN (T1 if you're in the Telco-Logically backward USA like me) with channel bank... Even then they don't always work... Just my $0.2 ... - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 5:46 AM Subject: Re: [Asterisk-Users] feature - VM gain adjust? Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote: I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich Rich - I'll say that this would be very useful. Regardless of where the loss is being inserted, it still exists. I like the idea of associating the voicemail db adjustment on a per-channel basis. I don't want to have to dink around with yelling at the telco to fix something that just works otherwise. Their answer will be Well, turn up the volume on your phone! which is exactly what your proposed patch will do. A simple trial-and-error process should be able to sort out the proper adjustment on any typical system that doesn't have radical db changes across time. I'm heartily in favor of this idea; I'll even throw a donation towards it, if you have a PayPal account. Another cool feature would be app_volume, which would turn up/turn down tx/rx levels dynamically, but that's left for a different day, and after we have an enhanced app_dial that lets single-digit dtmf sequences jump to dialplan routines and then can reconnect bridged calls. See my various rantings about this in months (years!) past. When I get some spare time (ha ha ha) I should really learn how to code this stuff... JT The above feature request has been entered as bug #2023. It also appears that VM has an issue (by itself) with recording/playing volume. Transmitting a 1004hz tone at 0db through a ata186 (set for -1db fxs loss), and then retreiving the same VM results in that tone measured at ~ -10db. Doing the same from a pstn location (via TDM FXO) suggests the same -10db loss (in addition to the pstn loss). Zapata.conf rxgain and txgain set to 0. Using CVS-HEAD-07/12/04, but same result with CVS-HEAD-07/1/04. Entered as bug #2022. Add comments to either if you'd like. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
On Mon, 2004-07-12 at 09:31, Seth Remington wrote: What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. While I have tried to stay out of the comments here for a while, I would suggest not going post processing. While it might get the problem fixed for now, it isn't a good long term solution. I have experienced similar trouble with recordings from AGI. We have some recordings that where dead on sound wise, and others that ended up being so soft as to be useless. Would it be something people would like to be able to add filters to a line? Consider normalization as a filter. Monitor could then be moved to a filter as well. Echo cancel could be a filter. Set it up so multiple filters could be added and chained together. This could help those with echo chain a couple of filters together and see if that helps. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Normalize for Linux can tell you the levels of a wav and can be used to adjust it according. Been toying with using it for some of my streaming media clients since it sucks to go from too low and having to up the volume to very loud. On Mon, 12 Jul 2004 10:31:08 -0400, Seth Remington [EMAIL PROTECTED] wrote: What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. -Seth On Mon, 2004-07-12 at 08:46, Rich Adamson wrote: Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] feature - VM gain adjust?
I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Let's see you have 11.2 db of loss from the phone you are using to call in on and the FXO interface on Asterisk. Retreiving voice mail would add another 11.2 or a total of 22.4 db. But your measured tone level was 36db. In other words the FXO interface and Asterisk introduced about 14 db of loss. I would find this amount of loss to be unacceptable. Rather than hack at the code, why not find this additional loss? A PBX or telco switch should not introduce this much loss, IMHO. Lyle - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 6:00 PM Subject: [Asterisk-Users] feature - VM gain adjust? I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
Good catch. I've got two pages full of test results for various items and copied values from the wrong page. Regardless, it's still an issue of very low volumes when voicemail involves creation and access from a pstn location. Let's see you have 11.2 db of loss from the phone you are using to call in on and the FXO interface on Asterisk. Retreiving voice mail would add another 11.2 or a total of 22.4 db. But your measured tone level was 36db. In other words the FXO interface and Asterisk introduced about 14 db of loss. I would find this amount of loss to be unacceptable. Rather than hack at the code, why not find this additional loss? A PBX or telco switch should not introduce this much loss, IMHO. Lyle - Original Message - I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] feature - VM gain adjust?
At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote: I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich Rich - I'll say that this would be very useful. Regardless of where the loss is being inserted, it still exists. I like the idea of associating the voicemail db adjustment on a per-channel basis. I don't want to have to dink around with yelling at the telco to fix something that just works otherwise. Their answer will be Well, turn up the volume on your phone! which is exactly what your proposed patch will do. A simple trial-and-error process should be able to sort out the proper adjustment on any typical system that doesn't have radical db changes across time. I'm heartily in favor of this idea; I'll even throw a donation towards it, if you have a PayPal account. Another cool feature would be app_volume, which would turn up/turn down tx/rx levels dynamically, but that's left for a different day, and after we have an enhanced app_dial that lets single-digit dtmf sequences jump to dialplan routines and then can reconnect bridged calls. See my various rantings about this in months (years!) past. When I get some spare time (ha ha ha) I should really learn how to code this stuff... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
