similar to: asterisk command line problem

Hi all, I have installed Trixbox 2.8.0.3 with TDM400p digium cards (2 fxs - 2 fxo). Before making security settings I need to backup all system. When I click on Backup on System - Backup menu in admin panel anything come up on screen. I tried to observe another way some users suggested "mondo" for backup. With mondo I had had some problems before installation. Can anyone tell me how

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [root@asterisk asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl'

Hi Everyone, I recently upgraded to Asterisk 1.4 using the RPMS at ATrpms.net on CentOS 4.4, Asterisk starts up but when I start the console it reports this error and drops out. "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist)?" I have checked to see that the file asterisk.ctl actually exists. Any suggestions? -- "I never look back darling, it

I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does

Hi everyone, I have a problem at my Trixbox that is version Asterisk "1.2.26.1 svn rev 79171" and 2.6.9-34.0.2.ELsmp kernel version. Two Digium 4fxs+4fxo card has been installed and everything was working before made yum update and at this server. (Centos 4.0). After update I faced with "zaptel not loading" problems. I have solved these problems too but now when I try to call

I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir => /var/run/asterisk runuser = asterisk rungroup = asterisk The directory,

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

Hi Please help me understand about the below issue ? [root at asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could

Hi, I've followed instructions of the book "AsteriskFutureOf TelephonySecEdit" on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1:

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [root@asterisk1 latest]# tar xvf libsupertone-0.0.2.tar libsupertone-0.0.2/ libsupertone-0.0.2/AUTHORS libsupertone-0.0.2/Makefile.am libsupertone-0.0.2/COPYING libsupertone-0.0.2/config/ libsupertone-0.0.2/config/ltmain.sh libsupertone-0.0.2/config/missing libsupertone-0.0.2/config/install-sh

Hi guys I am currently getting the following in my log asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found asterisk1*CLI> == Connect attempt from '127.0.0.1' unable to authenticate Can anyone tell me why asterisk would not be able to authenticate it's self?

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten =>