similar to: cli commands missing

Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files.

I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules

Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8So I have installed Asterisk 11 following the procedure in

I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've

Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the " == Parsing '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

This might sound lika a "small" issu, but here it goes: I'm a long time unix user and my shell history usage and editing is configured to use vi keybindings; it's something that's already built into my fingers and using different bindings, like the arrow keys to fetch previous lines, really blows me !... :-( Is there any way to setup the asterisk cli to use such

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

Any way to initiate a call and execute a playback of an audio file from the cli? My only chance to debug or make changes is usually when no one's at the office including me! Thanks! jlc

What is wrong when I can not execute any command that starts with sip ??? > freepbx*CLI> help sip show > No such command 'sip show'. > freepbx*CLI> help sip > No such command 'sip'. IAX works fine : > freepbx*CLI> help iax > iax2 provision Provision an IAX device > iax2 prune realtime Prune a cached realtime lookup >

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Thanks -- - Eric Smith

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. -------------- next part -------------- An

hi for asterisk users, please any asterisk distribution (or Trixbox) for windows ? (except for the Asterisk Win32) bicose asterisk for Win32 have a Free (limited) PBX Manager and i have a problem with it: 1. the asterisk Windows Service is not installed by default 2. unable to connect to it (no responce from it) help me please! thanks -------------- next part -------------- An HTML attachment was

When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI

Hi, I configured Asterisk to run as "asterisk" user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. -------------- next part -------------- An HTML attachment was

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

Hello dear list. I am currently working on a Automatic attendant, and the core things work, but I think the loop function isn't working as expected. I am testing this environment: a sip internal call from 301 to 501. The setup here is when 301 calls 501, and 301 doesn't enter an extension, it will go in loop, 3 times, and then hangup...Can't get that working. Could someone please

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am