similar to: Preflight check / lint

I had a RH7.3 box that provided some web services, but the hard drive recently started to throw errors. So I got another drive and built a new system using CentOS4. Unfortunately, there is no one to one as far as Apache since COS4 uses Apache 2 and there is a dramatic difference in the way the config files are done. I have managed to get it to recognize the directory on my other drive after

Puppeteers, I''ve seen things in the documentation to make me think this is possible but I''m not sure how to implement it. Here''s what I''d like to do: puppetd pulls down a config file, say httpd.conf BEFORE puppetd moves it to $path, I''d like to have it check it''s integrity with apachectl configtest IF configtest passes, then puppetd should

i just went through the tutorial on http://wiki.rubyonrails.com/rails/pages/HowtoInstallAndRunRubyOnRailsOnCpanel but i''m having some problems. when i add: <IfModule mod_fastcgi.c> FastCgiIpcDir /tmp/fcgi_ipc/ AddHandler fastcgi-script .fcgi </IfModule> under the line that says: LoadModule fastcgi_module libexec/mod_fastcgi.so apache fails to restart. i went on

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

How do I include a file (not a context) in AEL? #include "filename" returns an error. Thanks, Jay

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

The facter 1.6.18 tarball has many issues. First, the ext/osx/createpackage.sh script still uses packagemaker, which is deprecated. And even if it used that, there are a few references to files in the now-deleted conf/osx/ directory. So it''s pretty difficult to create a Mac pkg from this, but after some tweaking of the createpackage.sh script, I managed to do it! I ran the package it

Good evening, I have not able able to run the http://ip-addr/src/configtest.php script, nor access Squirrelmail. Looking for suggestions on what I might have missed. When I try either http://ip-... or https://ip-..., I get the following reply Forbidden You don't have permission to access /webmail/src/configtest.php on this server. I got squirrelmail installed in /usr/share/squirrelmail and

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

> On Thu, 18 Jun 2020 16:11:01 -0400 > Jay Hart wrote: > >> Any suggestions???? > > selinux issue? I turned the firewall off and tried to access /webmail and the configtest file, Turning off the firewall resulted in same issue. > > I have this in my squirrelmail notes for Centos 6, maybe this is a similar issue for you on Centos 8: > > After configuring

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That