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Hi there, i called one asterisk server from another asterisk server. The calling server played back a audio data und the answering server recorded the audio sample using record() function. I tried both ISDN, VoIP connections. Camparing with the original audio data, the recorded samples from both connections were amplified by asterisk, so that the recording were much louder. But I didn't

Hello there, recently installed CentOS6 on a (quite old) 64-bit system, and from the GNOME's sound preference app, I can see that the Output volume range is said "amplified" from 63% to 100%. Below 63% it's unamplified. No idea what's implied behind this amplification (I don't see this on other desktops), but I notice that when "amplified" the sound is pretty

Hello! I have a cabinet full of Wilcom Enhanced Line Powered Amplifiers with Manual Balance, model ELPA-421V. I *believe* these were used for a bank of analog modems back in the mid-90's. They were removed from a suite when the old company moved out. Here's a URL: http://www.wilcominc.com/elpa421v.htm Does anyone have any interest in these? If so, ple...

I have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset everytime you finish a call. I have even purchased the Aastra 9120 which sais it has on-hook...

> Just curious, is it possible to amplify Ogg frames without a decode/encode > cycle? I usually normalise before encoding all recordings for my radio > station, but I obtain some music from 3rd parties and it often need > normalising. I find this ability very useful for MP3s and it would be a big > handicap to me if Ogg can not. Add a comment into the ogg file :

...the SB Live! 5.1 series. That is 6 analog outputs. A Live! Player 5.1 is 40 USD here at my place. > Most people don't have them. Doing real 5.1 output from the PC without a > receiver that locks you into Dolby Digital is either expensive (full pre/pro > combo) There was a surround amplifier available for 1000 ATS in Austria recently. That is cca. 64 USD. It has 4(6) analog inputs and can drive four speakers ( it has 4 amplifiers ). You need another normal 2 channel amplifier to get an 5.1 setup. I believe it also has a Dolby ProLogic decoder, but that is not of interest to us. Is has...

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

...o lists the H-91 and H-101 as certified to work, however these are both over-the-head type models. I was looking for an over-the-ear model, as I would like to be able to provide a variety of headsets depending on the individuals taste. I am not looking for a headset that requires an external amplifier, but rather a headset that can make use of the headset jack on the phone itself.

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

Hello all. We have an asterisk working perfectly but we need a sollution for the PA system. Before Asterisk PBX we had an expensive analog PBX which plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the amplifier, into the amplifier. I've configured a pjsua with auto answer but the audio is very po...

Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device for all calls? Thanks for any ideas! Alvin

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

Hi, I've had a small case with noise suppression and AGC. I have a fairly noisy environment here, and with the default parameters, noise suppression works fairly well while I talk. However, when I shut up, AGC starts slowly increasing the gain until it has amplified whatever noise is left to levels about equal to having no filtering at all. As soon as I talk, AGC backs down fairly quick

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the original signal present in a live recording merely degrades the playback system's ability to reproduce the audible signal. Because a perceptual coder removes inaudible...

...quot;hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help disguise this I've been implementing various 'amplifier' style sample tricks including: * A pre-amplifier. * A self adjusting pre-amp that aimed for a target RMS (did an astounding job of listening to the roadworks outside). * A compressor. * A compressor that "expanded" at the bottom end of the range. * A noise floor implemented on a p...

...the Living room. I also have a pair of speakers in the basement and two more sets of Labtec PC speakers in the kitchen and the Guestroom/office. It's definitely not a HI/FI setup, but it works! So what do you use to Listen to your Oggs and would you have any recommendations on sound cards, amplifiers and speakers if I wanted to upgrade my system. Thanks Evert ________________________________________________________________ GET INTERNET ACCESS FROM JUNO! Juno offers FREE or PREMIUM Internet access for less! Join Juno today! For your FREE software, visit: http://dl.www.juno.com/get/web/. ---...

> Then I pulled the microphone out. Our system still records noise. To > isolate the problem, I wrote a small app just to open the device and > record raw samples, calls speex_denoise() and outputs both sample > sets. > The noise is still there, with level fluctuating with gain level, > unless > "All mute" is chosen. > In the case when NO microphone is

Hi, The leakage problem you describe is very, very common and you will need to do something to address it. I modified the version of Speex I use to implement an adjustable max gain. If you look at speex_compute_agc in preprocess.c, you will see: if (agc_gain>200) agc_gain = 200; This max of 200 is usually more than enough to amplify leakage which occurs either in the sound

...han 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to help you. But lots of effects are not linear--simulating a tube guitar amplifier, for example. Even something as straightforward as resampling a signal to 44.1KHz is going to benefit from starting at 192KHz rather than 48KHz. There may not be more signal information but there will be less noise. -a -------------- next part -------------- A non-text attachment was scrubbed.....