similar to: problem with SU100

Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap....... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using

Hello Everyone. I have noticed a number of errors on the Digium FAQ page. I dont know if this is the right place to mention this, but apparently # is not a comment indicator. Many samples would not work, but the following changes made them work for me. Doesn't Work: d) Edit configuration files and make sure they have at least these lines: /etc/zaptel.conf fxsks=1 #X100P fxoks=2 #S100U

I am running * in a development environment, adding functionality as I go. The * box has a X100P card in it which ztcfg enabled as channel 1 with fxsks signalling (fxsks=1). Everything worked fine and I was able to make inbound and outbound calls to/from the PSTN, the only issue being that some exchanges wouldn't handle the dtmf signalling, but I put that down to a peculiarity with some AU

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

I'm attempting to configure a machine with a DevKit Lite (X100P & S100U). After I modprobe wcfxo, the machine goes into some kind of loop after about 10 seconds, where it spits out what appears to be 32-bit addresses, ad infinitum. At this point, the machine becomes completely unresponsive to keyboard input (i.e. Ctrl-C, Ctrl-Alt-Del, even CapsLock and NumLock states [lights] cannot be

Thanks in advance root@asterisk01:~# modprobe wcfxo /lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device /lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcfxo.o: insmod /lib/modules/2.4.20/misc/wcfxo.o

Hi, I have an IBM server running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to

I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal. 1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event and displays the message on the screen. I tried dialing but nothing happens. I hangup and * shows the hangup event.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've tested the S100U unit in two different computers both with different USB subsystems. The one it worked the best in used the usb-uhci USB driver. The other system uses the usb-ohci USB driver. To make sure it was not the installation, I used the same hard drive that had asterisk in both systems. And I moved the X101P and S100U between both

Hi, I'm trying to get an S100U that came as part of the dev kit light working. I have tried compiled EHCI as modules or directly into the kernel but the same problem happens both times: I connect the S100U and run modprobe wcusb - initially nothing happens (great - no errors, etc). Fairly soon the server will die. It seem to be related to communications with the S100U. For example if I

Hi Everyone, I've been using Asterisk now for a few months for my small office (which is mostly just me while other guys are always on the road so we rely heavily on telephones) - I'm very excited with Asterisk as it can do everything I've ever wanted to do with a PBX. I'm having a problem with an S100U USB --> Telephone interface. I haven't actually made it work yet

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

Hi folks! I'm using an X100P (connected to my phone line) and an S100U, and when I calls out from the phone connected to the S100U it is a very bad sound quality, it "pops" and "jitters" a lot. But internal calls from for example a SIP client to the phone on the S100U sounds good. Calls from an SIP client to the outside world using the X100P also works good! My second

Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Thanks. MCH -------------- next part

Hello- Sorry if you see this again, I am new to this list and accidentally put this message in another thread. What I want to know is, is there any way to send a hook-flash signal from a cell phone (and then have Asterisk pass it up to the PSTN?) I suspect we need an example. I have an X100P to the PSTN and an S100U to an Analog Phone (ext. 3100) Let's say my PSTN number is 555-1296.

We have a S100U which we were using as our sole internal phone extension. Today it died. Picking up the phone now results in a brief dialtone followed by bursts of random static and crackling noises and then a second or two of fast-busy followed by silence and more crackling noises. I've tried the hardware attached to a couple of different machines as well as a couple of different phones.

I have an ATA box on my local LAN, talking to asterisk. On a remote LAN I have another instance of asterisk; that box has an S100U as its primary instrument. I set up an extension to dial between the two, sort of a "private line," and with the exception noted below, things appear to work just fine. That exception is echo. The "ATA" side of the conversation has a

Hi everyone I have a Digium card TE110P and after plug it, turn on the computer and configure, the LEDs don't light up in spite of what Digium FAQ says about the LEDs: When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP? ... The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install