similar to: Echo with Digium TDM02B

Hello all, I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Hi, I got a card from Digium TDM with 2 FXO modules (red ones). There is a problem that has me quite upset and is that asterisk always detect tones repeated two, three or more times. i mean, if i press 123 on my phone. asterisk detects somethin like: 1112233333 or 1111112333 or things like that. How can I fix it? I tried to change the volume level but this happens even if is set to 0. Why this

I am a little confused about the FXO ports on the Carrier Accesss Bank 1. The wiki, and people on the mailing list have stated that they do not have impedance matching, but Carrier Access states in their documentation that they do have impednace matching. Did Carrier Access make a typo, or does it depend on the version of the FXO card? I current have a CAC AB1, and I am having echo problems

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

> > First, ensure the TDM card has been defined to match the telephony > standards for each country where the card is installed. Bad echo on these outside North America is likely to be because of impedance mismatch. Echo cancel isn't likely to properly fix it. I was surprised at the lack of info about setting up the TDM cards for various countries. I eventually worked it out from a

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine ? 1 x AMD A64/3500+ CPU: Desktop Athlon64? Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon? 64 S939 NVIDIA nForce(r)4 SLI? PCI Express Req: 24pin ATX 1 x

What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. ?120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two

I have Asterisk 1.2 and a generic Wildcard single FXO card (a cheapo from eBay). I have read about many people who have used these cards without an issue and I'm just testing to work up a new system. The problem I have is that if I call the telephone number of the line attached to that card and have configured my incoming calls to, say, forward to an extension, then it will ring that

Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are "far less" however we're still getting them. While I have

Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of them have a ADSL splitter. The location are Spain, i load the tdm driver with opermode

Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now "incompatible" with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset