search for: loudspeakers

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs server in room A. the client from room A records from microphone and sends the recorded voice to server in room B. the server in room B plays it on loudspeakers. in room A there is also anoth...

Hi, I plan to use AEC for a live performance, storytelling for very young children (and their parents!) in a mongolian yourte . Actually the storyteller can make vocal loops, there is an omnidirectional microphone in the center of the yourte, 5 loudspeakers in a circle along the yourte's wall and Pure Data in a linux box. And now she wants to make vocal loops over music and loops over loops... Maybe aec will help her? I did some testing on a small setup, 2 desktop loudspeakers and have very good results (20 to 30dB rejection) with mono music and...

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, S...

...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 still could be louder. Maybe a light-up option would be better. The old phone system here had some huge loudspeakers that someone had wired right into the speakers of the old digital phones. I haven't figured out yet if they need a different voltage, or even if they still work; they were not responding when I replaced the attached phones. Justin C. Sherrill - American Rock Salt p: 585-991-6825 f: 585-991-69...

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being used, I only know that it is patented. Since it...

Hallo, an alle Wine-Freaks, das Programm - bis jetzt Freeware-, ist High-End unter den Hornsimus: http://www.users.bigpond.com/dmcbean/Setup.exe , 400 kB Ich habe wine in etlichen Versionen bis zur 0.9.26 probiert, auf Debian und Mepis (ubuntu), auch Crossover von 4.2 bis zum Beta 6.3a. Es geht nicht, alle Dinge wie native dll,s andere Winvers bringen keine Besserung, IE, Dcom--hilft nicht. Alle

Not sure yet what or where the problems are but having just done a jum update on my HP laptop (nw8240) and my IBM desktop from 2.6.9.42.0.2.EL the laptop is locking up during boot sequence and the desktop when running VMware Workstation seems to take all CPU and makes strange noises from the loudspeaker. Using grub to fall back to last kernel all is ok again !!. Ian -------------- next part

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 signal content, t...

Hi everyone, I've just released Speex 1.2rc1. This adds support for acoustic echo cancellation with multiple microphones and multiple loudspeakers. It also adds an API to decorrelate loudspeaker signals to improve multi-channel performance. In the bugfix department, there are fixes for a few bugs in the echo canceller, jitter buffer and preprocessor. At this point, the API for 1.2 should be stable and only a few very minor additions are plan...

Hallo, I'd like to get all channels of an ac3 5.1 stream coded into vorbis. With sox and oggenc it doesn't seem to work. Does anyone know a tool that can do this under linux? The ac3 stream is also avialabe as wav6 file converted by the cvs version of a52dec. This wav6 file contains one stereo stream and 3 mono streams. Is there a convention which stream number in an ogg file means

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or

...eex-dev@xiph.org Oggetto: Re: [Speex-dev] echo cancellation > echo cancellation: > speex_echo_cancel (st, ptr, echo1[0], e_buf, noise); speex_preprocess > (den, e_buf, noise); > > where ptr is a buffer that was recorded from microphone, echo1[0] is a > buffer that was played on loudspeakers. the delay between ptr and > echo1 is 2 buffers (512 samples). > with this configuration I can still hear the echo signal (althought > the signal is a little more silent. when using the delay 3 buffers the > echo is still not cancelled and after few seconds I don't hear anything....

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call spee...

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but