similar to: 123 Basic configuration files

Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/ I get notices when I try to run asterisk like: parking.conf is depreciated in favour of features.conf Please rename it. (I'm not getting rename parking .conf to features.conf ???) I am also missing: mgcp.conf enum.conf cdr.conf I using dev-lite Digium board that has 1 fxo and 1fxs port. So far I was

Oh geez nobody respond, I guess I have to be more descriptive. I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled. I'm using a Dell Optix (whatever). Everything seems to be in order. I connect an analogue telephone to the FXO port? and try to dial. Dial tones work however not dial tone.

I'm a newbie, after installing asterisk software and the Digium Lite card I do not get a dial tone. The telephone push buttons makes tones but no dial tone. Is this a hardware problem ? Can some one point me in the right direction. Sincerely, Don Moskaluk don@moskaluk.com www.moskaluk.com --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).

I currently working on a project using Locustworld's wireless mesh network and running SIP through the network. I want to install Asterisk as server running sip and Digium FXO FXS dev lite card. Unfortunately I having the time of my life to install Asterisk. The astinstaller does everything perfectly but when I try to configure an analogue telephone to setup voice mail I do not get any dial

as i am trying to use asterisk and install my newly purchased ( got it yesterday) digium cards. i am following the very detail steps of http://www.automated.it/guidetoasterisk.htm. but one thing did not seems right so i wanted to let enveyone know the page says: Once compiled make sure there is a copy in /usr/bin/mpg123 i think the location is /usr/local/bin/mpg123

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hi Experts, I am writing a guestfish script . I want to implement conditional logic inside guestfish script. For example , if state.tgz exists in image do something else if onetime.tgz exists in the image do someotherthing . Is it possible to do this within guestfish script. I tried to search a solution for this in guestfish doucmentation but couldn't find anything useful. Thanks in advance

Hello, I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I

hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up

Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent,

Hi, I'm trying to setup Asterisk on a Linux (redhat 8) machine. There are no analogue phones to be used & use is purely for internet traffic (SIPs). I've followed the setup guide from : http://www.automated.it/guidetoasterisk.htm But cannot get asterisk to run. If I type asterisk at the command prompt I get invalid instruction If I type ./asterisk start I get [FAILED] Can

Hello My friends, Today my asterisk stop working and i could see the following messags in /var/log/asterisk/messages at the time that asterisk stop working: [Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms / 2000ms) [Feb 16 13:24:41] NOTICE[8230] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds [Feb 16 13:25:54]

Where can I get some asterisk books.. or tutorials..? I?ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic

Hi, I am working at adding namespace to my packages, carefully following the doc "Writing R extensions" and some threads on the web. However I cannot find clear explanation about how to best deal with the import or importFrom functions in the name space. To make it short: To declare dependencies in the description file either after Depends: (packages including functions that are

Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom

Hi, before you start throwing stones to me let me tell you that I am a bit new to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July 2004, as described in Andy Powell's "Getting Started with Asterisk" (http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read about the Asterisk 1.0 RC1, and I would like to download it and install it. Could someone tell

Hi all, iam a student trying to build an asterisk pbx as a simple configuration only two extention (using Xlite)without outsite telephone line. i already follow the instruction and seem the asterisk work fine because there is no error message. when i configure SIP.conf and extention.conf i hope the phone will ring each other as an extention. but it doesn't work. i follow the instruction at

Dear all, I have been using the contrast library in my teaching for the last couple of years and am right in the middle of this year's round. In the last week R has been updated to version 2.14.0 on our computers. This has had the unfortunate effect of meaning the contrasts library no longer works, as the Design library is no longer available. I wonder if anyone has a fix for this...or

Given an HTML file which looks like this: --------- begin snippet --------- <HTML ><HEAD ><TITLE >We've Lied to You&#8230;</TITLE ><META NAME="GENERATOR" CONTENT="Modular DocBook HTML Stylesheet Version 1.79"><LINK REL="HOME" TITLE="Maximum RPM" HREF="index.html"><LINK REL="UP"

Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050725/14265bb5/attachment.htm