search for: ztmonitored

To start ztmonitor in quantitative mode you do the following. Assuming you are running Asterisk V1.0.9 you need to edit one line to ztmonitor.c in /usr/src/zaptel as per patch 2783 http://bugs.digium.com/bug_view_page.php?bug_id=0002783 Change line 261 fprintf(stderr, "Usage: ztmonitor <channel num> [-v] [-f FILE]\n"); to fprintf(stderr, "Usage: ztmonitor <channel

Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *********Wiki start******** If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor displays full bar (or whatever maximum allowed by rxgain) in RX. It only drops zero when the line picks up (and remote was silent). Is this something of concern? The zap channel seems to work despite echo. Additionally, what's the objective of tuning with ztmonitor? I mean, what would indicate an optimal level?

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

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, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

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

Err, stupid me. Seems like the phone is hungup, but when I start ztmonitor, the line is picked up. I guess the line in India is pretty loud b/c the RX bounces off the charts, changed it to (-4) and all my problems are solved. Regards, Min Chang On 4/14/06, Kyle Sexton <ks@mocker.org> wrote: > > Have you tried putting a Hangup in your extensions.conf? > > > > On 4/13/06,

Details: Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: India This is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the zaptel.conf, trying zaptel v1.2(with

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

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 wouldn't normally post this to the asterisk mailing list but I'm really stuck... I've been trying to get meetme working on and off for a few months now but I always hit a brick wall when trying to compile. I keep seeing this... make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

I have been trying to adjust the gain as per this document without any success: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html I have a PSTN and VoIP (SIP) connection via *. I disabled all echo cancel/training in zapata.conf and set tx/rxgain to 0. I then changed my extensions.conf so that when I call the VoIP line from the PSTN line, it plays the Milliwatt

I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical

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

Have anybody experience with the following error on a linux system. The system is for the rest running perfect without problems. The system was full installed with Suse 9.1 and updated. Following uname is my kernel 2.6.5-7.104-default Greatings Han Suse 9.1 professional AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk

Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see

I try to compile zaptel, without much success. I followed the guidelines in http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation making dependencie results in: asterisk:/usr/src/linux # make dep *** Warning: make dep is unnecessary now. and make tells me make[1]: *** No rule to make target `modules'. Stop. asterisk:/usr/src/zaptel # make clean rm -f torisatool makefw tor2fw.h rm -f