Is this a joke? It's not fixed.

Dashboard says "There are 30 bad destinations". Underneath that, it lists all of the destinations...

DEST STATUS: ORPHAN
Announcement: AATIGreeting
Blacklist a number <*30>
Remove a number from the blacklist <*31>

etc, etc...
Running FreePBX 13.01RC1.3 upgraded from 12.

Any ideas? Thanks.

hello：
I reinstalled everything, but it shows such errors:

installing files to /var/lib/asterisk/bin..error copying files:
'cp -rf' from src: '/var/www/html/admin/modules/framework/amp_conf/bin/*' to dst: '/var/lib/asterisk/bin'...details follow
cp: cannot stat '/var/www/html/admin/modules/framework/amp_conf/bin/*': No such file or directory
done, see errors below
installing files to /var/lib/asterisk/agi-bin..error copying files:
'cp -rf' from src: '/var/www/html/admin/modules/framework/amp_conf/agi-bin/*' to dst: '/var/lib/asterisk/agi-bin'...details follow
cp: cannot stat '/var/www/html/admin/modules/framework/amp_conf/agi-bin/*': No such file or directory
done, see errors below
PHP Warning:  Uncaught exception 'Whoops\Exception\ErrorException' with message 'require_once(/var/www/html/admin/modules/framework/installlib/installer.class.php): failed to open stream: No such file or directory' in /var/www/html/admin/modules/framework/install.php:125
Stack trace:
#0 /var/www/html/admin/modules/framework/install.php(125): Whoops\Run->handleError(2, 'require_once(/v...', '/var/www/html/a...', 125, Array)
#1 /var/www/html/admin/modules/framework/install.php(125): require_once()
#2 /var/www/html/admin/libraries/modulefunctions.class.php(2363): include_once('/var/www/html/a...')
#3 /var/www/html/admin/libraries/modulefunctions.class.php(2315): module_functions->_doinclude('/var/www/html/a...', 'framework')
#4 /var/www/html/admin/libraries/modulefunctions.class.php(1878): module_functions->_runscripts('framework', 'install', Array)
#5 /var/www/html/admin/libraries/Console/Moduleadmin.class.php(196): module_functions->install('framework', true)
#6 /var/www/html/admin/libraries/Console/Moduleadmin.class.php(86 in /var/www/html/admin/modules/framework/install.php on line 125
PHP Fatal error:  main(): Failed opening required '/var/www/html/admin/modules/framework/installlib/installer.class.php' (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/admin/modules/framework/install.php on line 125
Whoops\Exception\ErrorException: main(): Failed opening required '/var/www/html/admin/modules/framework/installlib/installer.class.php' (include_path='.:/usr/share/pear:/usr/share/php') in file /var/www/html/admin/modules/framework/install.php on line 125
Stack trace:
  1. () /var/www/html/admin/modules/framework/install.php:125
Done
---------------------------------------------------------------------------------------------------------------------------------------------

I copy the installlib files into to the required dir and can installed without problem. this is the final steps:

[root@localhost freepbx]# ./install -n
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
Reading /etc/asterisk/asterisk.conf...Done
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Done!
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...No (/etc/amportal.conf file detected)
Initializing FreePBX Settings
Finished initalizing settings
Copying files (this may take a bit)....
  900/5415 [====>-----------------------]  16%/var/www/html/admin/i18n/de_DE/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/de_DE/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/ja_JP/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/ja_JP/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/fr_FR/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/fr_FR/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/bg_BG/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/bg_BG/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/he_IL/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/he_IL/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/nl_NL/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/nl_NL/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/hu_HU/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/hu_HU/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/it_IT/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/it_IT/LC_MESSAGES/amp.mo has been changed from the original version.
/var/www/html/admin/i18n/amp.pot has been changed from the original version.
/var/www/html/admin/i18n/es_ES/LC_MESSAGES/amp.po has been changed from the original version.
/var/www/html/admin/i18n/es_ES/LC_MESSAGES/amp.mo has been changed from the original version.
 1000/5415 [=====>----------------------]  18%/var/www/html/admin/assets/js/module_admin.js has been changed from the original version.
/var/www/html/admin/assets/js/search.js has been changed from the original version.
/var/www/html/admin/assets/js/pbxlib.js has been changed from the original version.
/var/www/html/admin/assets/js/script.legacy.js has been changed from the original version.
/var/www/html/admin/assets/js/modernizr.js has been changed from the original version.
/var/www/html/admin/assets/js/jquery-ui-1.11.4.custom.min.js has been changed from the original version.
/var/www/html/admin/assets/less/freepbx/freepbx.less has been changed from the original version.
/var/www/html/admin/assets/less/freepbx/chosen.less has been changed from the original version.
/var/www/html/admin/assets/less/freepbx/jqueryui-overrides.less has been changed from the original version.
/var/www/html/admin/assets/less/freepbx/menu.less has been changed from the original version.
/var/www/html/admin/assets/less/freepbx/buttons.less has been changed from the original version.
/var/www/html/admin/functions.inc.php has been changed from the original version.
/var/www/html/admin/bootstrap.php has been changed from the original version.
/var/lib/asterisk/bin/retrieve_conf has been changed from the original version.
/etc/asterisk/modules.conf has been changed from the original version.
/etc/asterisk/cdr_adaptive_odbc.conf has been changed from the original version.
 5415/5415 [============================] 100%
Done
Finishing up directory processes...Done!
Creating missing #include files...Done
Running variable replacement...Done
Setting up Asterisk Manager Connection...Done
Running through upgrades...
Checking for upgrades..
No further upgrades necessary
Finished upgrades
Setting FreePBX version to 13.0.1RC1.3...Done
Writing out /etc/amportal.conf...Done
Setting Permissions...
 8711/8711 [============================] 100%
Finished setting permissions
Installing all modules...Checking if field did is present in cdr table..
did field already present.
Checking if field recordingfile is present in cdr table..
recordingfile field already present.
Checking if field cnum is present in cdr table..
cnum field already present.
Checking if field cnam is present in cdr table..
cnam field already present.
Checking if field outbound_cnum is present in cdr table..
outbound_cnum field already present.
Checking if field outbound_cnam is present in cdr table..
outbound_cnam field already present.
Checking if field dst_cnam is present in cdr table..
dst_cnam field already present.
Generating CSS...Done
Module cdr successfully installed
Updating Hooks...Done
Generating CSS...Done
Module music successfully installed
Updating Hooks...Done
Generating CSS...Done
Module infoservices successfully installed
Updating Hooks...Done
Checking if directdids need migrating..already done
updating zap callgroup, pickupgroup..not needed
checking for delay_answer field ..already exists
checking for reversal field ..already exists
checking for pricid field ..already exists
Checking if trunk table migration required..not needed
Checking if privacy manager options exists..already exists
Checking for noanswer_cid field..already exists
Checking for busy_cid field..already exists
Checking for chanunavail_cid field..already exists
Checking for noanswer_dest field..already exists
Checking for busy_dest field..already exists
Checking for chanunavail_dest field..already exists
Unable to add index to extensions field in users
Checking for General Setting migrations..not needed
Deleting unused globals..done
Converting IAX notransfer to transfer if needed..updated 0000 records
deleting obsoleted record_in and record_out entries..ok
checking for dest field in outbound_routes..already exists
checking for continue field in trunks..already exists
upgrading any zap trunks to dahdi if foundok
Generating CSS...Done
Module core successfully installed
Updating Hooks...Done
Generating CSS...Done
Module featurecodeadmin successfully installed
Updating Hooks...Done
Refreshing all UCP Assets, this could take a while...
Generating Module Scripts...Done
Generating Module CSS...Done
Generating Main Scripts...Done
Generating Main CSS...Done
Done!
Generating CSS...Done
Module ucp successfully installed
Updating Hooks...Done
Generating CSS...Done
Module logfiles successfully installed
Updating Hooks...Done
Checking for General Setting migrations..not needed
checking if Voicemail Admin (vmailadmin) is installed..not installed, ok
Generating CSS...Done
Module voicemail successfully installed
Updating Hooks...Done
Creating cel if needed..OK
checking for extra field..already exists
checking for userfield field..already deleted
Generating CSS...Done
Module cel successfully installed
Updating Hooks...Done
Generating CSS...Done
Module customappsreg successfully installed
Updating Hooks...Done
Generating CSS...Done
Module dashboard successfully installed
Updating Hooks...Done
checking for sipsettings table..already exists
Migrate rtp.conf values if needed and initialize
Generating CSS...Done
Module sipsettings successfully installed
Updating Hooks...Done
Generating CSS...Done
Module callrecording successfully installed
Updating Hooks...Done
Generating CSS...Done
Module userman successfully installed
Updating Hooks...Done
Done installing modules
Installing framework...
installing files to /var/www/html..done
installing files to /var/lib/asterisk/bin..done
installing files to /var/lib/asterisk/agi-bin..done
Checking for upgrades..
No further upgrades necessary
framework file install done, removing packages from module
file/directory: /var/www/html/admin/modules/framework/amp_conf removed successfully
file/directory: /var/www/html/admin/modules/framework/upgrades removed successfully
file/directory: /var/www/html/admin/modules/framework/start_asterisk removed successfully
file/directory: /var/www/html/admin/modules/framework/install removed successfully
file/directory: /var/www/html/admin/modules/framework/installlib removed successfully
Generating CSS...Done
Module framework successfully installed
Updating Hooks...Done
Done
Generating default configurations...
Checking for PEAR Console::Getopt..OK
Skipping extension and destination registry checks
Please update your modules and reload Asterisk by browsing to your server.
Finished generating default configurations
Trusting FreePBX...Trusted
Setting Permissions...
 8200/8200 [============================] 100%
Finished setting permissions
You have successfully installed FreePBX
[root@localhost freepbx]# reboot


--------------------after installed everything, but i still can not access the GUI from broswer/empty------------  
it looks ok from asterisk -r command, but GUI dead:
[root@localhost admin]#  systemctl status -l freepbx.service
● freepbx.service
   Loaded: not-found (Reason: No such file or directory)
   Active: inactive (dead)
[root@localhost admin]#  systemctl start  -l freepbx.service
Failed to start freepbx.service: Unit freepbx.service failed to load: No such file or directory.
[root@localhost admin]# /usr/sbin/fwconsole restart
Running FreePBX shutdown...

Checking Asterisk Status...
Run Pre-Asterisk Shutdown Hooks

Shutting down Asterisk Gracefully...
Press C to Cancel
Press N to shut down NOW
Stopping Asterisk...
 120/120 [============================] 100%
Asterisk Stopped Successfuly

Running Post-Asterisk Stop Scripts
Running FreePBX startup...

Checking Asterisk Status...
Run Pre-Asterisk Hooks

Starting Asterisk...
 100/100 [============================] 100%
Asterisk Started on  3835

Running Post-Asterisk Scripts
[root@localhost admin]# ps
  PID TTY          TIME CMD
 1646 pts/0    00:00:00 bash
 3833 pts/0    00:00:00 safe_asterisk
 4342 pts/0    00:00:00 ps
[root@localhost admin]# asterisk -r
Asterisk 13.5.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.5.0 currently running on localhost (pid = 3835)
localhost*CLI>
-------------------------------------------------------------

Is it still true that there are "lots of little issues" with CentOS 7? I only ask because I noted on the manual install page (http://wiki.freepbx.org/display/HTGS/Version+13.0+Installation) that CentOS 7 is quoted as "The recommended distribution" to build from.

Resolved.

Thinking back to my last changes. I remember I accidently clicked on Download all or Upgrade all in the Module Admin. All the commercial modules installed, and so I am sure there was something that was not only not configured properly for one of the modules, but I know they were not registered either.

Had me scratching my head for a moment.

yum update fail2ban fixes it

In my experience, if your provider is solid, faxing isn't that tough these days. I have hundreds of faxes per day going through FreePBX straight into email using the built in features. Lots of faxes everyday sent/received using cheapy Grandstream ATA's with default settings out of the box on the ATA. Can't remember my exact setup but, it's always worked really well, at least for the last few years. These customers are mostly hooked up directly to Level3 SIP though and the connection is not being bounced through six resellers of resellers of resellers degrading the connection etc. Level3 does support T.38 on their trunks but, even before we had T.38 on it was working almost flawlessly on both the ATAs and Fax to Email. I have also had good luck with our Vitelity trunks. I use Anveo but can't say that I have ever tried to push faxes through them.

By the way, this is all on a fairly old version of FreePBX running I think Asterisk 1.8 as the base.

I think the idea of receiving faxes using a fax machine these days is just bizarre, why would anyone want that. If you can convince your customer or users that receiving their faxes by email is a better way, then you only have to deal with outbound faxes using an ATA. Better yet, you can have them setup a Vitelity account to do the outbound faxing if you like, I have done this with a few clients for just outbound and they have been happy campers. Vitelity's faxing solutions "Just Work", I have been using them for my own inbound faxing for years and never remember someone telling me they couldn't send us a fax.

Hope that helps.

Please don't paste full logs on here. There is no way we can make out whatever you are trying to tell us between all of the noise you just posted

It really seems like you aren't following documentation listed on our wiki for manual installs as "installlib" is only removed after framework is finished installing, the fact you had to drag it back leads me to believe you are trying to reinstall an already installed system which is very bad. Additionally it looks like you ran "install.php" instead of "install" (maybe? perhaps? I am not sure). The biggest glaring issue is that libinstall was removed after install which wouldn't happen if you are following our wiki install guides. (Its removed from the web path NOT the install path)

Additionally your first error is even more baffling as you had an asterisk.conf file but it was missing vital run directories. Seems like you were trying to install freepbx on a machine that already had freepbx installed.

Are you not using the FreePBX distro? If you are doing this manually please start over and follow our documentation: http://wiki.freepbx.org/display/HTGS/Version+13.0+Installation

Here is an example asterisk.conf file for you to review (note that yours is missing the [directories] section):

[root@freepbxdev1 ~]# cat /etc/asterisk/asterisk.conf
[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
transmit_silence_during_record = yes
languageprefix=yes
execincludes=yes

Now you just need email2fax, relatively simple with postfix and hylafax as a back end. Still no machines involved. Works like a champ.

So my fax woes continue..... But thank you all for your input.

Like a few of you have mentioned, G711 and T.38 faxing has been very inconsistent. The other day, it seems 90% of faxes were going through. Today I was testing and got less 50%. Of course, this is for a number of factors.

The doctors offices we're doing VoIP for really prefer using their giant all-in-one machines to send and receive faxes, so it seems an ATA is what we're going to have to use.

I think I'm going to look into a Level 3 trunk and some different ATA's. We can choose to only use Level 3 on Anveo, which is a route marked at T.38 capable, but I'm not sure that is the same as getting one directly from Level 3 quality wise. That being said, I have read about a lot of people having bad luck with the Cisco SPA112's we're using, and then having much better luck with some other brands/models, using their same SIP trunks.

Funny thing is, a handful of our customers use the same ATA, the same PBX (FreePBX, current distro), and the same trunks we're providing via Anveo, and never complain.

Oh, how I wish I could get more logical answers in my troubleshooting!

I'm gonna check out Vitelity's faxing too.

As an optimist, my favorite post is AdHominem's: "Get rid of the Cisco ATAs and replace them with an Obi 200. I use them all the time for Fax Over IP using G711 with ZERO problems." I'm definitely buying a couple of these bad boys right now.

Dicko, with hylafax and/or sandsp, this is a purely efax solution correct?

tonyclewis, this may be a good solution too. Thanks.

Thanks again guys for all your input.

luckman212,
check if you've got the System Admin module installed, not necessarily activated/registered but just present.

I run a VM for testing and installed FreePBX 13 and it also show that OpenVPN server is not running and the only reference to VPN anything I can find in FreePBX is in the System Admin module, which I have not activated or registered the PBX as it's just a test VM.

My live PBX does have OpenVPN running but I don't have the System Admin module installed so it does not show the "OpenVPN server" on the dashboard.

It would be great to have options somewhere that you could select what shows in that list on the dashboard, whether you use it or not...

Absolutely not! you can use real fax machines plain and simple through an ATA , you can add "virtual" iaxmodem or t38modem to virtualize , you can print all incoming faxes to any networked printer routed by DID and send any file (converted by hylafax and it's extensions) you can use scanners as input devices and there even some hylafax clients that work in winbloze all you need is software that reads and write G4 tif files and that basically is hylafax/libreoffice/imagemagick/ghostscript with all the bells and whistles. Furthermore you have email2fax and fax2email abilities available.

Very cool. Thanks. I definitely need to look further into it then.

CentOS is recommended. Try not to look too deeply into versions at this time

Woah. Please try to be more kind in your responses. Perhaps you need to clear your cache or use a different browser

Right; should I add the context to the custom file as well, then, and it will use that instead of this one?
If I overwrite anything in that file, it will trigger the warnings...that's where my confusion is coming from.

I've already written some test code into the extensions.conf works great!

Hello everybody !!!

I setting up a code on a specific trunk in outbound route, this trunk is a landline for a specific extension,
so now i want to allow this extension to dial out using this trunk without putting code while other extensions will have to put code to dial out.

Look forward to your email .

Thanks !

Add these lines to extensions_custom.conf

[macro-dialout-trunk-predial-hook]
exten => s,1,Noop(This line will appear in the log for all outbound calls)

In its simplest, that is all you do.

Okay, thanks again Lorne! I'll play with this some more

Sent the link to you a short while ago.. please confirm..