similar to: txgain / rxgain no effect

Hi, Do you have some experience with the "best" values for those parameters in youyr particular case? I mean the best raport between sound level in both direction and echo cancellation. For me, the best result I can get is with: rxgain=10 txgain=15 ... the sound level is good, but the echo is a little bit to strong for my taste. Something interesting is that if I put a txgain value

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP

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

I am looking for docs on how to diagnose and adjust the rx/tx gain in zapata.conf. The wiki has a link to this article but it no longer exists on the server. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so

Hi, I know that this issue has been discused guite a lot, but I haven't managed to get a definite answer. Is those two values supposed to be floats (e.g. 3.5) or integers with the percent symbol (e.g. 20%)? Thanks, Yiannis.

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

I've got a rather strange callerid problem and don't know how to debug it. I've got two X100P cards installed in a RH9 box and both are connected to individual pstn pots lines. Both are equipped with callerid and both display callerid info on regular analog phones (verified 100%). Incoming and outgoing calls are handled just fine. When an incoming call arrives on one X100P card, *

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi All, I've got an MD3200 modem which is working as a Zaptel FXO interface for Asterisk (X100P clone I believe). It seems to work, but on incoming or outgoing calls I can hear the other party ok but when I speak, I hear my voice echo back at me (quite quietly but it's distracting!) on everything I say. The other party doesn't seem to hear the echo - they just say I sound a bit quiet.

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi, I've got an x100P and I'm able to dial out and make phone calls with it ok but I just want to set it up to answer the phone and be a simple answering machine but it doesn't seem to want to answer the phone. I keep getting this: on the console when the phone rings: -- Starting simple switch on 'Zap/1-1' Nov 28 08:55:09 NOTICE[29298]: chan_zap.c:5458 ss_thread: Got

Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have

I have a simple little asterisk setup: FXS is a PhoneJack PCI, FXO is an X100P. I have a regular old cordless phone plugged into the FXS port. I can dial anything and it's picked up properly by *. I can call FWD and it picks up the DTMF properly. I call out the FXO but nothing I call through there can hear the DTMF clearly. Bell Canada's Call Answer service, for instance, can't