similar to: VERY bad sound on S100U -> X100P calls, and caller id problems ...

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

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

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

Hello, I just bought a X100P from digitnetworks. It is supposed to be a FXO card, but there are 2 rj-11 plug on the card. One is labelled "phone" and the other "pstn". When i plug the "pstn" on the wall and the "phone" on my analog phone, everything (incoming and outgoing calls) works like before (without asterisk). AFAIU, i should have an FXS card in my

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

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

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.

I have * configured with 2 X100P cards (fxs_ks). The lines from the telco are 'analogue both way ddi trunks'. This means that every inbound call contains digits that represent an extension on the PBX. I can make outbound calls from * with no problem however I cannot receive inbound calls on these trunks. Some investigation has shown that when the line is idle there is 50 volts present and

Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric

Hi all! I am in the same situation here in Sweden, but I just got the X100 card and don t get it to work with DTMF caller ID. Is the only way to solve this to buy a TDM400 card w FXO modules? Is it just a driver question or a HW issue with the X100 card? Where to find dok about the new keywords in conf files like "cidstart=polarity" in zapata.conf? rgds Gunnar >>>

I have asterisk successfully dialing out using a X100P over a normal analogue PSTN line. But when I try to dial out over an analogue line that goes via a hardware PBX the call asterisk does not dial. Is there a configuration change I should make ? I am thinking of something like not wating for dial tone.

Hi folks. I am using a X100P card and after some random amount of time of correct operation, say 8-20 hours, the card starts acting up and producng horrid sound quality which is all broken up. All other channels appear to work fine. One thing I noticed, is that "zap show channel 1" always shows the "Actual Hookstate: Offhook" as soon as the telco line is plugged in. Is this

-----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

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

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.