similar to: ztmonitor

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

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?

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! 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'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between

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

I'm still relatively a novice with asterisk and am having issues with echo. The calling party that calls a PSTN number doesnt hear the echo, but the answered side via sip or forwarded to another PSTN number over voip hears excessive echo that makes it difficult to communicate. I've been playing with the zapata.conf settings for echocancel, echotraining, rxgain, txgain, etc and am

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

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 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.

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module 2) Is it best to use a TDM card with an TFX and FXO module or is the the FXO = X100p and FXS on TDM a good method? 3) How do I

Hello everyone, I am having big problems trying to detect dtmf tones while a IVR prompt is playing on zap channels. Sometimes the detection only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems. And most times it also does not detect all digits, eg when 123 is keyed in, it may detect 23 or 2 or 3, but never the complete

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,

For those that have had X100P echo problems, it seems that somewhere just prior to Asterisk CVS-12/04/03-14:24:40 it has been fixed to where its hardly detectable for just a couple of seconds at the start of a call. (I use to have it rather bad for the first 15 seconds or so.) I'm using DECHO_CAN_MARK2 with KFLAGS+=-DAGGRESSIVE_SUPPRESSOR and echotraining=yes echocancel=yes

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 an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Hi All, Channels 25-28 on a customers PBX are regular Zaptel FXO cards that are hooked into 4 incomming phone lines. They are all in a group to do automatic rollover for outgoing calls (if channel 25 is being used, dial on channel 26, etc.). Sometimes when a user is dialing a number, instead of bridging to one of the FXO cards it goes and rings to Zap/1-1. This doesnt occur all the time but some

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the