similar to: X100P - FO card

Hello all. I want to make pagination URLs pretty instead of the ?page=2. I have written a route like this: map.connect ''restaurant/list_dishes/:page'', :controller => ''restaurant'', :action => ''list_dishes'', :requirements => { :page => /\d+/}, :page => nil Which appears to work. However the following only

Wednesday''s Beer & Pizza SIG was attended by about a dozen Rubyists, plus one extremely well-dressed gentleman (no hacker he!) who was on the prowl for Rubyists to hire. Beer and Pizza (among other things) were consumed, Ruby (among other things) was discussed, and everyone appeared to have a good time. The upcoming SIG will, as announced previously, be held just up the street at

Hello all. I have the following Code in a view: <% @dishes.each do |dish| -%> <tr> <td><%= link_to dish.name, :controller => "restaurant", :action => "show_dish", :id =>"#{dish.id}" -%> <% if dish.vegetarian? -%> <%= image_tag("/images/vegetarian.gif", :class => "bevel", :size => "8x8")

Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (<100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything >300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

> Date: Sun, 19 Sep 2004 12:59:52 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: [Asterisk-Users] Re: [Asterisk-Dev] X100p on VIA EPIA-V > problems > To: asterisk-users@lists.digium.com > Message-ID: <Chameleon.1095617288.adar0@vegas> > Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > > > I'm running Asterisk on my (new) VIA EPIA-V

I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

PROBLEM: My x100p only does bad things. When I plug the line from the wall into the card, other phones in my house go nuts, no dialtone, crazy clicking, random tones, etc. Following the steps below, the line is screwed up any time I test after step 2. Just plugging the line from the wall into the x100p takes my other phones off line. External calls to my line hear busy. Obviously, can't

Hi We are trying to get better sound quality out of the prompts on our Asterisk system. We had some new ones made by thevoice.digium.com and they are in WAV format instead of the default GSM format on the Asterisk server. The problem is, when you dial in to the server using the FXO card (X100P) you really can't tell the difference between the WAV prompt and the GSM prompt, however, if you

Anyone tested this card in Greece? I would like to buy several of these but I don't know if it works well with the PSTN in Greece. The Quicknet Linejack card seems to be unable to understand the telephone network's tones (busy, ringing,...). TIA, Michael.

Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul.

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best, Michael --------------------------------- Do

I'm trying to get the X100P card working in AU. So far I have managed to get it to handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db

Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I

I have a friend I am building a phone system for, I would like to use * He owns a "Take Out" restaurant with 4 Phone lines and 4 Phones. I'm going to use Cisco 7960's for the phones; is there a better phone I should be using? I need to know if this is possible... On each phone program the appearances of 4 "Extensions" that are really the 4 phone lines? If Phone