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To all: I am researching the feasibility of replacing our current PBX (ATT Partner Plus) with an * PBX. I have purchased an X101P card and I have * running of a FC2 machine. The X101P is connected to an extension on our current PBX. Many archives exists regarding X101P cards but I just need some guidance. First the X101P did not detect a ring, so I found an email where someone

Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users

Hi, I have a X101P card and am running Asterisk SVN-trunk-r226890. The X101P card seems to be identified properly: debian:/usr/src/dahdi-tools# dahdi_hardware pci:0000:01:01.0 wcfxo+ e159:0001 Wildcard X101P debian:/usr/src/dahdi-tools# dahdi_cfg -vv DAHDI Tools Version - SVN-trunk-r7409 DAHDI Version: SVN-trunk-r7445 Echo Canceller(s): MG2 Configuration ======================

Hey all, I've been trying to get my X101P working again as of late (it used to work great) and before I decide to trash the card I thought I'd post up my symptoms to see if anyone has any ideas. My old working config was basically 1 channel running fxsks signalling. It was working great with no echo, busy detect worked well and I was very impressed considering this is all off and

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

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P.

Hi every one, i have the problem opening MS Powerpoint 97 from samba share. the ms ppt always shows: <file name> is not correct where the file name is manged. i tried to disable manged name in samba, but the problem still happens thank you for advice. -- To unsubscribe from this list go to the following URL and read the instructions: http://lists.samba.org/mailman/listinfo/samba

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

Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and

Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and

Let me start by saying no, this is not the normal stupid echo question :) I currently have a SIP device and an X101P, and have had the usual echo issues and have played around with the various solutions, none of which are quite perfect (understandably).. however, my girlfriend is a little more picky than I am when she uses the phone, so I'm considering a Digium TDMXXB card for the house. My

I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

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

I'm looking to get an x100p off ebay and am not particularly familar with the "life cycle" of the card.. An "Authentic X100P" listing has a buy it now of $29.95 and says There are 3 types of cards Asterisk would recognize: *Screenshots from the official, original driver install Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs

I just installed two new X100P's today in my RH7.3 box and was poking around /proc What does the RED refer to on board 2? /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 /proc/zaptel/2 Span 2: WCFXO/1 "Wildcard X101P Board 2" RED 2 WCFXO/1/0 Any clues on how to setup channels on the two interfaces?? I am up to here in

Has anyone been able to sucessfully use 2-4 X101P clones with Asterisk on a single system ? I'm using one Encore MD 3200 modem sucessfully, alongside a TDM400P (3 FXS, 1 FXO) and would like to be able to use up to 4 X101P on a single system. In most cases I'll have 2 or 3 instead. I understand all the issues with interrupt sharing and PC motherboard quality. Just need to know if any

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) My /etc/zaptel.conf is: fxsks=1 fxoks=2 loadzone = au defaultzone=au Channel 1 is a X101P card connected to the PSTN and channel 2 is a S100U box driving an analogue phone. The zaptel kernel module gets loaded OK as does the