similar to: Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set "quietmp3" but what about when using native?

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with

Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

does anyone know of or would be interesting in writing a program that is an alarm clock for your computer that plays ogg vorbis files? i used to have one that played mp3z before i heard about vorbis, but it got trashed somewhere and i can't find it again because i forget what it was called. but i would be interested to have one that plays ogg vorbis files, as right now the only alarm clock i

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

I have tried both linear and non-linear (residual echo) parts of SPEEX AEC on MS Windows platform. Great job. Most echo can be cancelled. However, the voice volume becomes too small. Does any one have suggestions that makes voice volume louder but echo still can be cancelled? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Problem: When talking to someone (from pstn) and this person is not talking loud, the voice is cut by Asterisk. I tried increase RxGain but it changed nothing (was talking louder but voice still cut.) I use XLite as soft phone. I think this is probably a codec setting... but how do I check that on server side? I just don't know what to do. All works fine (asteriskathome) but I always

hi all I just setup a new site, perhaps soon a wiki, to collect what's out there of useful backports from Trunk/1.4 beta back to 1.2. Take a look at http://http://www.asterisk-backports.org/ and judge for yourself ;) roy -- Roy Sigurd Karlsbakk roy@karlsbakk.net (+47) 98013356 --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way.

hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even louder

We've had a few people complain that the "beep" before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing

I just noticed that quality of .gsm files for using with asterisk is not that good.. is there any way to make then sound better? asterisks sample voices sound way better than theones recorded using applications like wavepad or with asterisk like unavailable messages... any tips? Do you know the command line for sox to adjust the volumen levels or gsm files (make the louder)? Also, do they have

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain

I just tried this with Goldwave.. it decoded the file, i edited out some stuff, and saved the file (flac format) and the tags appear to be intact. Greg M. --- Tomas Valusek <tvalusek@seznam.cz> wrote: > Hello, > > and what if I want to preserve FLAC tags while I'm > editing its contents? > Is there a way e.g. to cut silent block from FLAC > and preserve tags? By

> Any idea or suggestion to avoid having two sound cards to do what I need to > do ? > > Or how to fade in / fade out without using the card' mixer, while playing > i.e. using mpg123 ? Depends on how much you want to program. Basically, the easiest would probably be to use OSS software mixing; if you buy the commercial OSS drivers, you can have several programs writing to