similar to: Still unacceptable echo on X101P

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

> > 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

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, Has anyone in australia got asterisk running on FreeBSD? how would i pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload doesnt let you pass parameters to the module like modprobe in Linux. I tried to get the sysctl variable using sysctl -a it might use but nothing applicable was visible.... i need to set the impedences to 600 ohms, can someone please advice me on

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

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

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

Hi, I've just added an X101P FXO card to my asterisk machine. The card is detected ok but zttool always shows it in a RED alarm state, which I understand means that it doesn't detect the line. I've connected the card via the line socket to my analogue line using the cable that came with the card (and have also tried other cables too) but I always get a RED alarm. I'm located in

Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card

Since the X100P/X101P/Clone cards does not work in all countries that use DTMF based Caller-ID systems, I've developed a hardware that you connect to a serial port and the PSTN. You then run a perl script "cid_logger.pl" as a daemon, and modify extensions.conf to call an agi script whenever a call comes in, and if it's on the X100 card it will get the caller id information

I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? --

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and

When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time

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

X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver.... Just an indication of how many people have got this to work would be useful. Cheers Mark.

During some calls, and during the playback of some messages (that people leave), I have audible beeps... I don't know if the other party can hear them, but I sure can. What is that? Also, I currently have 2 x100p (single fxo card), I have read that they tend to have a lot of echo problems, but that the new 4 port fxo card does not. Can someone verify that for me? I am looking to upgrade

Hi I received the Asterisk Dev Kit Lite on Saturday and got a small PBX working fairly quickly using the config files supplied with the kit. This was on my desktop machine and the sound card stopped working. I turned the system off and moved the sound card to another slot to give it a different IRQ. On reboot, the modules zaptel, wcfxo, and wcusb all load OK and asterisk starts OK but there

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have worked well with 1 >> carrier but

Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case