similar to: Fun with Echo

I figured I'd answer my own thread and document what it took to get rid of the echo at my location. For those of you trying to get rid of echo, let me tell you, "what worked for that guy, probably won't work for you". I think we've all heard that before, and it's true. Let me assure you that echo can be removed from your phone lines. At 20 hours into my 40

I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time

Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation.

Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it as-is with the dahdi one ; to no

Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone --

I searched the list archives, and looked around on Digium's web site, so I appologize if this is answered somewhere. In SVN what is the difference between the Zaptel drivers in "branches" and that in "trunk"? If I download the stable version 1.2.6 linked from asterisk.org do I have the "trunk" code? Thanks! Brian

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call

We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of "office-style" phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> I recently upgraded my Asterisk system to Dahdi and now I have an echo <br> problem. <br> <br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium <br> TE121B PCI express

Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I would like to double check what Echo Cancellation my Digium Card uses. I thought I bought the little more expensive card that integrates EchoCancellation. How can I check? root at sn1:~# zaptel_hardware pci:0000:0b:08.0 wcte12xp+ d161:8000 Wildcard TE121 root at sn1:~# ztcfg -v Zaptel Version: SVN-branch-1.4-r4309 Echo Canceller: MG2 Is MG2 the correct one that I am supposed

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But