similar to: Huge Echo

Hello! I'm having an echo problem with a TDM card. The TDM card is being fed by a channel bank just 12 or so feet away. When you put an analog handset on the line, both the RX and TX volume seem to be just fine. However, when I use the TDM card, I have to have an rxgain of 13.5, and even then, the audio is relatively quiet. I'm also getting echo on these lines, so I have turned

I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production

Hi, I've seen some reports about ruuning Intel modem with 537 or MD3200 chipset running with Zaptel drivers as a FXO port. Did anybody managed to set up a PCI faxmodem based on Intel536ep chipset to work with * and Zaptel drivers? Modem seemd to work just fine with Linux, but the driver says no;) some more info: Linux 2.4.26 mazuchna:~# cat /proc/pci | grep 536 Communication

Thanks for the suggestion. I have busydetect=yes in zapata.conf. You refer to "Martin's algorithm". Can you provide more details please? ----- Original Message ----- From: "Soner Tari" <list@kulustur.org> Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing To: "Asterisk Users Mailing List - Non-Commercial Discussion"

I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm

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

> > That would be called a milliwatt generator. It likely exists in their > central office, but its typically used by their technicians to ensure > new installations meet specs and sometimes in troubleshooting. Call your > telco repair number and see if they will give you the telephone number > for it. I'll try that. 1st line technical support at Telstra are not known for

Hey Guys! We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make

Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

I have an x100p which suffers from echo (no surprise there apparantly :), and a few of the things I've read about tuning out echo say the first thing to get right is the tx and rx gain, and for that you need a few different types of test lines from the telco (Telstra, Optus, whatever). Does anyone have any numbers in Australia for such lines? I think they just need to send out a tone of a

Hi, i have installer wine from the Ubunu Dapper Repositories but i can't access the audio properties...when i run winecfg and click on audio, i come back to the console...its a problem with the alsa config, he cann't found a file... here is what i get : <CODE> ALSA lib seq_hw.c:456:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory Creating link

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

i have a box running debian sarge with asterisk installed from distribution packages: CLI> show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small office with just a few phones) I have one Zap (PSTN) line connected to it and one SIP to a local provider. After

Hi there, I have a new asterisk working in New Zeland and everything is working well except when an incoming call to the PSTN hangs up, asterisk wont hang up the zap trunk (X100P). I have found this information: http://bugs.digium.com/bug_view_page.php?bug_id=0001474 Which discusses my problem and i have made sure that i have the latest info in the indications.conf as follows: [general]

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

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'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around

Hello! Given the current discusison regarding ztmonitor, line testing, etc., I've been looking into purchasing a used transmission test set. From my research, it seems that there are two items that might fit the bill: the HP 3551A and the HP 4935A. I know nothing about these specific devices. I *do* have a good background in electronics, and I understand the concept what they're

I'm looking for some input from someone with real experience of telephony. I am having problems with the sound quality on our PSTN line calls. Our channel banks are Adtran 600 and 750 and I spent a lot of time on the phone with Adtran trying to work out the problem. We are getting hum and noise and very low volume on calls. I can increase the gain in zapata.conf but that increases the echo