similar to: Error working with X101P and S400P cards (fwd)

Could someone please assist me soonest. when i run asterisk with one X101P card and a S400P card asterisk will run but the card will not work.this is the message i get: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_mysql.so] =>

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

hello * users, i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium card, RH8.0. soon after running * i get this warning "file chan_zap.c,line 4155 (ss_thread):CallewrID returned with error on channel 'zap/7-1' then executes the other applications itself. --executing Wait("Zap/7-1","1") in new stack. -- executing..... it gives the same warning

I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of

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 producing a writeup on all the things one run into building an * box. Which after a couple of boxes is showing up aplenty.) So this uses the working h/w and config's from another box. All I do is build in a new computer. Loading the modules I got: __________________________________________________ Zaptel Configuration ====================== Channel map: Channel 01: FXS

I am about to try our TDM400P E model from the developer kit (not the "Lite") we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as "Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)" in dmesg and no apparent errors...yeah. Is "S400P

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.

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

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

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 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 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

Hi, I'm new to *. I just installed my X101P card with * from the source on Mandrake 10.0 and I test it. Everything seems to work fine. When I call at my home office all the demo ivr seem to work. But I have one question regarding * using /dev/dsp0. I only have one sound card on my system and it has to be use by my personnal PVR called MythTV. I though that * did not need a sound card to work.

Hi, Does anyone have any experience connecting Asterisk to a Meridian system using an ATA2 and x101p? The basics work -- I can make outbound calls, receive inbound, and use flash to transfer calls, but certain things do not work, specifically with calls from internal extensions. - Does the Meridian/ATA2 pass any kind of callerid info? We do not have external callerid, but I'm not even

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 want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (AT&W) to be permanent)

HI, I just installed an additional module to a TDM400P card. Everything worked fine for a while, but after rebooting the server the new channel fails to be recognized by asterisk. Here is the output == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling -- Registered channel 2, FXO Kewlstart signalling -- Registered channel 3, FXO

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it