similar to: Zaptel Hung Up on x101p and cisco analog line

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

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

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 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 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

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 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

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.

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] =>

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've bought a Wildcard X101P card to play a bit with asterisk. I'm just trying to place a call to a number, wait until the receiver takes the call, and then play one of the .gsm sound files. I'm in France, so first I tried modprobing wcfxo with opermode=1, but the DAA mode remained FCC, so I changed the opermode variable in wcfxo.c to 1 by hand and recompiled & reinstalled

Any other FreeBSD users experiencing this particular problem? I have an X101P board plugged into a PC running 5.2.1 and using the latest (0.6) zaptel port right from the port tree. zttool shows a RED alarm on the card but the phone line appears to be okay (plugging an analog phone into the "Phone" port produces a dialtone on the phone). I've even tried switching the line polarity and

Pause between successive incoming rings on my phone line is 4 sec, so when x101p do not receive next ring signal after 4.5 sec, call should be consider as ended. But, if caller hang-up, call is ended (Hungup 'Zap/1-1) exactly 8 sec after last ring, and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change

Hello I'm using an older PC from 2000 that we have lying around at the office to learn about Asterisk using Pound Key 1.0.1 (http://www.rpath.org/rbuilder/project/asterisk/). It's using an AMD Duron 600MHz, a Gigabyte GA-7ZX motherboard, and two FXO Astralis X101P clones that I bought through eBay. Here are the three issues I have. Sorry, forgot the floppy with dmesg, lspci, and

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