similar to: "Couldn't load variables.txt?aldope=xxxxx "

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value

Hi there, We did a clean install the AsteriskNOW 1.7.0 64 bits ISO and configured it. On the main screen (Crtl-ALT-F1) we keep seeing the following lines when making a call Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in

When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the called party gets transferred rather than the calling party. This is controlled by the reverse_transfer parameter in op_server.cfg but the behavior is exactly the same whether the parameter is set to 0 or 1. This is after the call is picked up by

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but they do not tell you were to put this file. There is something wrong with the

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg "Im-sorry&an-error-has-occured" and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some

Hi, I don't explain very well what my problem, but i can't make calls. i analise my log full and i found two errors Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-FAILED' priority 1 to macro-vm Jul 3 19:02:08

Well thanks I have changed the ip address now as you said the opposite is true however is there a command in asterisk that we can make both visible in var/www/html/panel/op_server.cfg ?

I have a fresh install of AMP. In the AMPortal, Setup, Devices or Users, I get: Cannot connect to Asterisk Manager with user/password (set respectively) This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available. I checked /etc/amportal.conf, /var/www/html/panel/op_server.cfg, and most of the conf files in /etc/asterisk/ Am

Hi, I know this is not a 100% Asterisk question, but is there anyone who has the Flash Operator Panel working with Asterisk 1.6?? In asternic.org there is a version that show call status but you cant make transfers or originate a call. Has anyone fixed the op_server.pl file to fully work with Asterisk 1.6??? Thank you very much, Pablo Bernasconi -------------- next part -------------- An HTML

Hello I have couple issues with AAH. 1.5 1. Flash panel doesn't show proper status. Sometime accessing with IP seems to work and it shows current line status etc. Sometimes accessing with hostname of the asterisk server seems to show lines, but it doesn't show off hook etc when we pickup a extension and talk. In /var/www/html/panel/op_server.cfg I have tried setting manager_host to all

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would

We have 1 server that after a few hours operating has multiple process of asterisk running. Here is the pstree output: # pstree init-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl |-crond |-cwASTcall.pl |-dbus-daemon |-events/0 |-hald-+-hald-addon-acpi | `-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd

I am participating in a project in the university, and appeared a doubt: if can I call one function from R as "solve" from C code. The intention of the research is to create a DLL. Is it possible and where can I find an example? Sorry for bad english! ___________________________________________________________________________________ Acesse nosso portal www.click21.com.br Porque

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

O que ? CentOS? CentOS Enterprise Linux ? uma distribui??o baseada no Red Hat Enterprise Linux. CentOS Cada vers?o ? suportada por 7 anos (por meio de atualiza??es de seguran?a). CentOS Uma nova vers?o ? liberada a cada 2 anos, e cada vers?o CentOS ? actualizado regularmente (a cada 6 meses) para suportar os hardware mais novos. Isso resulta em uma rede segura, de baixa manuten??o, confi?vel e