search for: chrisi

The machine in question is a DL360 G3 R03 Xeon DP 3060-512/533 (part number: 322471-421). It's the first HP-branded model we've had so far (previously all Compaq branded). It has 1Gb of memory and a couple of drives on the on-board SCSI controller (setup for mirroring in the SCSI bios). No SMP is setup right now (doesn't get far enough to do anything with it!). In any case, this

Hi Community, I''m currently trying to create a blog software in rails, but I''ve got a problem: I generated scaffolding for my articles and only want administrators to write, edit or delete articles. So I wanted to move this parts to another, secured controller. The one controller to display articles is called articles :), the other is also called articles, but is located in a

Hi community, as I''m a previous PHP and especially symfony programmer, I often try to find features of symfony in rails. Now I''m looking for a method which forwards to a 404 error page (but with stack trace and so on in development mode). I tried to manually render public/404.html, but if I do so, I miss the stack trace. Is there a method like this or are rails programmers using

Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew

Hello all, Downloaded, compiled and installed Asterisk CVS-04/15/04-17:54:5. Everything looks fine except I see a pseudo channel in the 'zap show channels'. xxxx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo default 1 default The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore: exten =>

Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour

Hi I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * -> (H323) -> CCM -> IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s

Hi all ! Whenever I call form_for a div with margin:0 and padding:0 is created. Not only that is the default value in my application.css, I have no chance to override this attribute. In my application I display a list of items, using <ul> and <li>. Each li tag contains an form, which should be displayed in one line (as inline element). But I can''t set the div to

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the

I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an "Error 1" after typing "make" but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly "Error 1" Anybody successfully compile Rate Engine? The least cost routing module for Asterisk? Thanks in advance.

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.

The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route!

Is it possible to get a few 7960's and asterisk to allow all of the 7960 phones to use one extentsion and can only be used by one person at a time, have it indicate on the other 7960's when one of the others has the line engaged. Basicly so like I can setup a rule when an incoming call comes from IAX to divert to this extension, it will ring the extension (thus all phones), and allow me to

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,

Hi all ! I''ve written a Character model, each character has many Attributes. My schema.rb looks like this: create_table "characters", :force => true do |t| t.string "first_name" t.string "last_name" t.integer "user_id" t.datetime "created_at" t.datetime "updated_at" end create_table "attributes",