similar to: Random Intermittent Noise for SIP to FX0 calls plus echo

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

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

Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. I have the following components in my call routing schema: - PBX with E1 port. - asterisk router with TE405P card(32bit/4 E1 ports). - Teles server with PRI interface card(3 E1 ports) and VTM

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

I am running speex 1.2beta1 on WM2003 using FIXED_POINT option, I got big clicking noise during encoding or decoding randomly, I got more noise if I spoke louder or made the MIC sensitivity higher, for example, when I encode the wav file to spx file, the original saved wav file does not have such noise, the noise is added only after it's converted to spx format. The floating point version of

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

Would it help if I scale the input wav data before encoding to spx? One interesting thing is if I encode the same wav file to spx(or decode the same spx file to wav ) on Windows platform, then there is no such big clicking noise, samething is true if I use floating_point speex library to encode/decode on WM2003. Problem happens only on WM2003 using FIXED_POINT library. Thanks Yanxin -----

Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard

I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Hi John, I noticed the values after decode is very much different from the values of the original PCM samples. For cases when the PCM sample values are constant (because no one is speaking), the decoded values fluctuate too. These fluatuations are causing the background noise. Is my observation expected? Thanks, Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To:

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles

Hi All, I have been wrestling with this problem for months now and at my wits end. The sound is always very low and if I try to bring the sound louder by using the sound references icon on the top panel, the sound is choppy with lots of static. Output of lspci for the sound device is as follows: 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1)

We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

Has anyone had problems accessing random websites since going up to 6.4? Since about the day after I got partly upgraded, if I try to access nytimes.com, or orbitz.com, I get server not found. With a lot of work, I, my manager, and the other admin, found that setting options edns0 in /etc/resolv.conf fixed it - I suspect that the network folks updated their internal nameservers (which are M$)

Am 2016-03-22 um 10:45 schrieb Garming Sam: > Hi, > > As you should know, 3.x is out of support. Assuming this is related to > the KB2992611 MS update, basically the bar was raised for clients in > response to a security issue, and caused havoc for people on Windows as > well. In order to fix 3.x, a good chunk of the infrastructure written > for Samba 4 (AD) would likely have

Hello all. I am reading the mailinglist for a long time now, and there was a thread i believe called webgui or something. In this thread there was a company i believe german, that was working on a dovecot appliance with a web based gui to administrate a dovecot mail server. The problem is i did not save the URL of that company. Does anyone know which company make the appliance. Or does