similar to: Background Noise Elimination

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding

Hi All; I installed one digium card of 2 fxo and 2 fxs, but the following problems related to the voice are happening: 1) Sometimes when I call to the PBX, I hear like modem sound and after little it disapear. 2) There is a disturbance in the background (like the channel radio disturbance that might happen if the frequency was not captured well), and that disturbance appear much more when

Not I. -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Tuesday, June 01, 2004 7:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sipura-SPA2000 background noise I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise that is noticeable, especially

Hi, I have had this issues for ages but have ignored it, but my handsets get faint background crackle on the FXS port. When I connect the handset directly into my PSTN, it goes away complete, so I am generating this from somewhere in my asterisk box. I get it immediately when the phone goes off-hook and it stays through the dialing and call progress. I can't think of any other devices

The background noise is unacceptable. I'll have to stick with 32kbps mp3s eventhough the files are 70% larger.

I'm using 22khz wavs because 16khz doesn't provide the quality i need in the original wavs. I can't get 32khz wavs because that part of the project is finished and not redoable. The .spx's don't have to be as high quality, so i guess i could convert them to 16khz. I was trying 17kbps, ultra-wideband, and VBR.

Hi, I am developing a peer-to-peer video conference application which uses speex as a codec for the voice. I am new to speex, so please bear with me if I asked the obvious. After I added the encode and decode function to my MFC app, I heard a constant background noise, even when no one is speaking into the microphone. #define FRAME_SIZE 160 The application is coded in MFC C++. The record

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Hi Guys, i have recently connected my (working) asterisk 1.2 server, with two 1.4 asterisk servers (one using SIP the other using IAX), since then (i believe) people starts complaining about a high background noise when using the handset on Polycom phones (but when using the speaker it's fine, and i noticed that my self), my question is, can anybody tell me any step to begin diagnosing the

I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and

Hi, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer

Hi everyone, Our team is working on a realtime voice communication application. We are using openAL 1.1 to capture the samples and then encode them using speex. On the remote machine we decode the samples and play them using openAL again. Capture and play formats for the samples is AL_FORMAT_MONO16 and frequency is 22050. we are using wide band encoding/decoding. The encoding sample

I''ve gotten quite a bit out of Pat Maddox''s screencast http://evang.eli.st/blog/2007/10/8/story-runner-top-to-bottom-screencast One thing I''m not sure of is the feature where he writes things like: When "I POST to", "/articles", :post => {:title => "Title", :body => "Body") do | path, params| post_via_redirect

On 12/24/2016 5:24 PM, geo.inbox.ignored wrote: > temperature regulated fan speed can reduce noise. > > use of a baffle box with styrofoam or thin foam rubber lining. > > -----+-----+ > fan \/ | | > /\ | | > --+-----+--- > ^ ^ > baffles > > spacing between upper and lower baffles must be equal to or

On 12/24/2016 10:25 PM, John R Pierce wrote: <> > he said 'server rack'. thats a probably 6' tall cabinet filled with > servers, each of which has likely dozens of high speed fans. > }} true. i was thinking 'old school' slide drawer racks. because i doubt that 'it' or 'heads of authority' will allow changes to server racks, H's best bet

If the server decision makers had not gone for a server designed by accountants but a server designed by engineers then you would not have this problem. Regards, Mark Woolfson MW Consultancy Ltd Leeds United Kingdom Tel: +44 113 259 0759 Mob: +44 786 065 2778 -----Original Message----- From: geo.inbox.ignored Sent: Sunday, December 25, 2016 7:49 AM To: centos at centos.org Subject: Re:

I have a system that is using Intel HDA as its output, and a Plantronics USB headset as the microphone input. I would like to have my wine configuration emulate this setup, I currently have been unable to confirm that it isnt... I just cant get Steam to choose the plantronics headset as the default voice device (this is the goal). Would appreciate any insight!

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions?

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc