similar to: Dial 0 to outbound

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

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any

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

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

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 ======================

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

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 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

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

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

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

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'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

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

Hi, I'm having some problems getting calls to go out via freshtel. There dosn't seem to be any specific information on how to get it working anywhere. The only information I've found is here: http://www.voip-info.org/wiki-Freshtel and that dosn't give you any idea of how to actually get it working. I've tried adapting information from other IAX2 provider examples but have

hi, i am having callerid problems with *. i have the callerid from my telco and it shows up in my normal phone when i connect it directly to the line but if i connect the same phone thru * server the callerid is not shown. i am using X101p and tdm400p. i have everything defined in my zapata.conf well and fine. i finally came to conclusion that this might have happened due to the registration of

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.

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not