similar to: combined AEC

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo

hello, the last 3 times I installed C5, the MBR was unchanged and nothing was written into /boot/grub except splash.xbm.gz there were no stage* files, nor a menu.lst I know how to fix that. Would I have better luck using a partition mounted as /boot? Anyone succeded with Grub that way? I prefer not having to do surgery to get C5 to boot :) -- Mark New Packages for C5 ---------

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

? hi all, i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" in my VOIP application. Is it that the frame length to be chosen should always be 20ms or can i have flexibility in chosing the frame lenght? on what parameters does the frame length choise depends? thank you all, Shri. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Any one know why the glibc-profile is obsoleted on centos 5 (glibc 2.5)? I've checked the glibc.spec file and find that the --disable-profile option is enabled in the configuration command. So how the original profiling functions are implemented on glibc 2.5? Is there a new way to do the same/similar thing? Thanks a lot.

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal

Dear all I am going to install asterisk on production now i m confused about version i dont know which version is good and best for my setup 1.2.x or 1.4.x can anyone tell me in detail which version whoud be best for my setup 1.4.x or 1.2.x if 1.4.x is good then which version whoud be better like 1.4.1 or 1.4.10 , 1.4.11 Regards satish patel

Hi: When should we use "unnumbered priorities"(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten => 333,1,Answer exten => 333,n,meetme(8000|cim) exten => 333,n,playback(vm-goodbye) exten => 333,n,hangup form2: [Conferencerooms] exten => 333,1,Answer exten =>

>Wine Is Not (an) Emulator. yup i know that :) >This means that it cannot run windows - it /is/ >windows as far as your >end-user applications are concerned. no issues with that >This means that you should not use your windows boot >as a 'fake >windows', but instead install applications properly >and use them. i usually run a setup of all my windows app in wine

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Hi all, > > I've got two problems compiling the current CVS FLAC sources on OSX. > > Firstly, the configure script can't find the OGG libraries which were > installed from MacPorts. I have tried: > > ./configure --with-ogg-includes=/opt/local/include \ >

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send

Hi, I've set up speex AEC in our application. The echo's seem to be canceling, but the captured voice is distorted somehow. It sounds to me like low and/or high frequencies are removed. This happens even if I set my playback(echo) data to only zero's. Is this "normal" for the AEC? As far as I understand, if I send 0's as playback/echo data, the resulting data should be

Hello This is a bit off-topic, but here goes: Try installing the "device-mapper-multipath" package and follow the documentation below - residing at the powerlink.emc.com site: "Native Multipath Failover Based on DM-MPIO for v2.6.x Linux Kernel and EMC(r) Storage Arrays" When configured this works just as good as PowerPath. This works great on RHEL / CentOS 5 Daniel On

> From: Chuck Kollars <ckollars9 <at> yahoo.com> > Subject: Samba & ACLs? > Date: 2006-08-19 02:46:45 GMT > How exactly do Samba 3.x and ACLs interrelate? ... > I started out naively assuming that the *nix > uidNumber/gidNumber Samba mapped the end user to > would behave exactly the same whether they were a > Samba user or were logged on locally. ...

Thank you for you quick reply, Jean-Marc. I have just used a human voice to replace the sine wave, but the result is like a broken voice after tens of seconds. Do you have some sample voice in .sw that I can test with? I want to have one set or a few sets of sample voice that work for my understanding and debugging. This is the output from echo_diagnostic.m in my human voice test: Far end to

many many thanks for ur help ok i m making my problem more clear... my LAN ip is :172.16.100.0/24 ip of eth1 is :202.51.78.122 and this is fixed and permanent address and it has no netmask and gateway(so, to provide internet to my LAN i have used the ppp0 for this connection ) ip of eth2 : 203.78.165.154; netmask:255.255.255.248; gateway:203.78.165.153 for eth0 my

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: comunip.gif Type: image/gif Size: 1663 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.gif

Hi,all I have tested AEC on files, it works well.I have some files,one is echo file, others are echo-added files(an origin file adding echo at different delay,such as 20ms,40ms...120ms,140ms).AEC do wonderfully on those files except echo added at 140ms-delay. But ,when i use AEC in my voip project, it does feebly. Who can give me some hints why caused this.How long can sound be picked up by