similar to: Asterisk echo & fxotune

This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c

I noticed the following came into cvs head yesterday: > Update of /usr/cvsroot/zaptel > In directory mongoose.digium.com:/tmp/cvs-serv2118 > > Modified Files: > fxotune.c wctdm.c wctdm.h > Log Message: > More TDM card echo API modifications. Making the fxotune program automatically > find the correct coefficients for the module. Lots of neat stuff. I did a complete

I'm trying to learn to use fxotune and am getting the error: Could not fill input buffer - got -1 bytes, expected 4000 bytes Failure! when I run it. This is a new install of Asterisk 1.4 and zaptel 1.4. Also I can't really find an up to date how to on it. Am I supposed to place fxotune on my startup path and run fxotune -s once it is working correctly. Any nuggets of info are most

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 -vvvv Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252 calibtype=2 waveformtype=-1 delaytosilence=0 silencegoodfor=18

Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Any advise? Regards Bilal

Is it still needed to run fxotune on the TDM24XXE cards with hardware echo cancellation? Jerry

Hey all, First of all, some background: Fxotune is a utility that is used to tune the hybrid on FXO modules For all of you with FXO modules out there, fxotune can help you adjust the analog and digital hybrid that is on the FXO interface and tune it so that it maximizes echo return loss. This means that it will reduce your default echo which is received, and will help any echo cancellers

Dear Pals One customer has a Panasonic PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now, This customer has 10 wireless links to his branches, wireless working great now, no voice at the present MY IDEA : T1Card into the Panasonic (additional to the E1) connected to a T1 Digium Card into Asterisk as far as I know the T1 Card can be configured as E&M to act as

Hi, Well, I have a big problem with Asterisk, my problem is that when I'm in a conversation, using zap channels, in a moment the line has a interferce that produce a sound in the conversation, this sound is a electratical sound I think, I was reading about that and I found that the utility fxotune can help me to change some settings about the audio and the supression of interferences. My

What does fxotune need to do its job correctly - complete silence on the line, or just absence of the dial tone? For how long? I've been trying to get a silent termination number out of my telco all morning and have come up empty. When I just dial "1" as some online guides suggest to break the dial tone, it is silent for a few seconds, then plays a "Your call cannot be

I recently ran fxotune against our incoming PSTN lines to try and help with some echo problems. It produced the following fxotune.conf file: 2=8,253,2,244,255,10,244,3,253 3=4,0,0,0,0,0,0,0,0 4=4,0,0,0,0,0,0,0,0 I'm a bit surprised by all of the '0's for channels 3&4, esp. given that it's populated values for channel 2. Is this considered 'normal' behaviour, or

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

Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' =============================================== make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f

Chuck, Thank You I'm also going to try CentOS 3 The problem is I have SATA HDD and running in to trouble getting Linux installed. Will update after I test Ver 3 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List -

Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've

I know this subjeact has been beaten in here so much, but, here is my issue.<br /><br />I have a TDM400 with 2 ports tied into our telco lines. I have run fxotune on the lines, and have got the conf file in my /etc folder. I have fxotune loading in my startup right after the drivers load up, then asterisk loads up. <br /><br />Problem! When calling out, we get a horible

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there