@Patte wrote:

Hello eveyone,

I installed the latest distro SNG7 64Bit and I want like to do a restore from our current system
FreePBX-Distro-10.13.66-18 / 64Bit to SNG7.

I read that someone did this and had some troubles. So a restore between major versions is not working

and I should do a migration:

https://wiki.freepbx.org/display/PPS/Elastix+and+PBXinaFlash+to+FreePBX+Distro+Conversion+Tool

I followed the instructions but I getting on your current FreePBX server (donor system) follow error message on the end:

[root@pbx ~]# curl -s https://convert.freepbx.org | bash
Checking that 'curl' exists ... OK!
Validating sha256 integrity ... OK!
Trying to download converter to /tmp/tmp.zFcCDLM6rP ... Complete!
Validating download ... OK!
Starting FreePBX Converter version release/20151217r1-12-g6f5db4b
Testing connectivity to Conversion server...Success!

FreePBX Conversion Wizard
-------------------------

The FreePBX Conversion Wizard needs to be run on two machines, firstly on the
NEW machine, which must be an ACTIVATED FreePBX Distro machine, and then it
must be run on the DONOR machine.
The DONOR machine is the machine that is currently processing calls, and is
the machine that will be migrated to the NEW machine. No changes will be made
to the DONOR machine, and this script will not stop or restart any services
that may cause an outage.
If this is the NEW machine, just push 'Enter' to prepare this machine

Enter Conversion ID (leave blank if this is NEW): 4f024dcc-3245-4022-8283-50cd6d5d6f96
Testing FreePBX functionality ... Success!
Checking Slot ID ... OK!
Getting modules to convert ... Complete! 106 modules
Dumping astdb ... Complete!
Do you want to copy /tftpboot to the new machine [Ny]? y
Creating encrypted backup for new machine. This may take some time.
Backing up the following:
983M    /var/spool/asterisk
51M     /var/lib/asterisk/moh
4.0K    /var/lib/asterisk/sounds/custom
330M    /tftpboot
Complete!
Sending backup details to new machine ... Complete!
Skipping module accountcodepreserve (Nothing to convert)
        Module announcement ... announcement
        Module areminder ... areminder areminder_calls areminder_settings areminder_updates
        Module arimanager ... arimanager
Skipping module asterisk-cli (Nothing to convert)
Skipping module asteriskinfo (Nothing to convert)
Skipping module backup (Nothing to convert)
Skipping module blacklist (Nothing to convert)
        Module broadcast ... broadcast_campaigns broadcast_campaign_groups broadcast_settings broadcast_groups broadcast_callees broadcast_log
Skipping module builtin (Nothing to convert)
Skipping module bulkhandler (Nothing to convert)
Skipping module calendar (Nothing to convert)
        Module callback ... callback
        Module callerid ... callerid_entries
Skipping module callforward (Nothing to convert)
        Module calllimit ... calllimit calllimit_usage
        Module callrecording ... callrecording callrecording_module
Skipping module callwaiting (Nothing to convert)
Skipping module campon (Nothing to convert)
Skipping module cdr (Nothing to convert)
Skipping module cel (Nothing to convert)
Skipping module certman (Nothing to convert)
        Module cidlookup ... cidlookup cidlookup_incoming
        Module conferences ... meetme
Skipping module conferencespro (Nothing to convert)
Skipping module configedit (Nothing to convert)
        Module contactmanager ... contactmanager_groups contactmanager_group_entries contactmanager_entry_numbers contactmanager_entry_images contactmanager_entry_userman_imagesERROR
Unable to continue.
Cleaning up...Done!
[root@pbx ~]#

Afterwards the test system showing up this message:

[/] Donor now sending module 'contactmanager', table 'contactmanager_entry_images' ...

After a whole night its still on this state.

I did some research, but I haven't found an answer. Maybe some of you can help me with this error:

contactmanager_entry_images contactmanager_entry_userman_imagesERROR

Kind regards

Patte

donor system (current system): FreePBX-Distro 13: 10.13.66-18
new test system: FreePBX-Distro 14.0.1.1: SNG7-FPBX-64bit - 12.7.3-1707-2.sng7

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@tonyg wrote:

hi, i am on FreePBX 14.0.1.1, i am trying to find where to defign the smtp server for email notifications. i found this wiki, but it seems to be referring to menus that are not in 14....I would appreciate anyone's help

thanks

https://wiki.freepbx.org/display/FPG/System+Admin+-+Email+Setup#SystemAdmin-EmailSetup-UseExternalSMTPServer

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@therobust wrote:

Hi,

I installed freepbx 14 on my debian 8 machine and GUI shows warnings like 'modules ... signed by invalid key'. When i try to upgrade modules it shows 'Warning: Cannot connect to online repository(s) (http://mirror1.freepbx.org,http://mirror2.freepbx.org).'

i saw from one discussion to try this command:

/var/lib/asterisk/bin/freepbx_setting MODULEADMIN_SKIP_CACHE 1 && amportal a ma upgrade framework && /var/lib/asterisk/bin/freepbx_setting MODULEADMIN_SKIP_CACHE 0

trying to run as user asterisk:

[MODULEADMIN_SKIP_CACHE] already set to [1]

Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
PHP Warning: Declaration of FreePBX\Console\Application\FWHelpCommand::setCommand($command) should be compatible with Symfony\Component\Console\Command\HelpCommand::setCommand(Symfony\Component\Console\Command\Command $command) in /var/www/html/admin/libraries/FWHelper.class.php on line 15
No repos specified, using: [standard] from last GUI settings

Downloading module 'framework'
PHP Deprecated: Methods with the same name as their class will not be constructors in a future version of PHP; xml2Array has a deprecated constructor in /var/www/html/admin/libraries/xml2Array.class.php on line 24
The following error(s) occured:
- Retrieved Module XML Was Empty
...

then tried following command:

2# amportal a ma refreshsignatures

Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
PHP Warning: Declaration of FreePBX\Console\Application\FWHelpCommand::setCommand($command) should be compatible with Symfony\Component\Console\Command\HelpCommand::setCommand(Symfony\Component\Console\Command\Command $command) in /var/www/html/admin/libraries/FWHelper.class.php on line 15
Getting Data from Online Server...
PHP Deprecated: Methods with the same name as their class will not be constructors in a future version of PHP; xml2Array has a deprecated constructor in /var/www/html/admin/libraries/xml2Array.class.php on line 24
..

Kindly help here!

Regards/

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@DarkQuark wrote:

Greetings. I have been working with the version14 distro since RC1. And since then through the release version I have the same issues.

Everything will run just fine (again Distro SNG7-FPBX-64bit-1706-1 and asterisk v13) until I activate the system and change the hostname via the sysadmin module. I normally change the timezone at the same time so it could be related to that as well.

What happens is I will reboot the system so those changes will take effect and this is what happens:

--I get this error on the CLI
[Exception]
Error From Module FreePBX\modules\Sysadmin: 'mkdir(): Permission denied'

--Asterisk will take 5-6mins to start
--The system will become unstable and seemingly fall off the network (loss of connectivity) right itself for some period of time and then do it all again. This is a slow cycle, it will be fine for hours, lose connectivity for hours and so on.
--I cannot seem to find anything related in any log file but my eye is not as well experiences as some of you here

Since I have seen this through close to 10 systems I feel like I can reproduce it at all.

Any help/suggestions/thoughts would be greatly appreciated. If anyone needs further detail about some part of this just let me know.

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@Rednick1990 wrote:

Good Afternoon,

Looking for some help with an issue I have run across. I just bought the Sangoma FreePBX Phone system 100. The current machine is a non distro machine. I used backup & Restore to copy the current FreePBX machine then used Backup & Restore on the new phone system 100 to restore the current working asterisk DB. All seem to go well with inbound and outbound working until about an hour or so went by. The Asterisk DB and MySQL began to crash the appliance making in non functional. I am completely stumped as to why this is happening. I am not too strong with FreePBX and any help or reasons why this is happening is much appreciated!

If you need logs or anything please just let me know!

Nick Fielding

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@therobust wrote:

Hi,

So after fresh install on ubuntu 16.10 (php 5.6), i again see the warnings 'module...signed by invalid key' .
to solve, i tried command: amportal a ma refreshsignatures which runs fine but these warnings are still there.

so, when i try to download/install new module i get this:
Error(s) downloading queues:
File Integrity failed for /var/www/html/admin/modules/cache/queues-13.0.33.tgz.gpg - aborting (GPG Verify File check failed)_

i tried following commands to resolve above error:
sudo -u asterisk gpg --refresh-keys --keyserver pool.sks-keyservers.net
sudo -u asterisk gpg --refresh-keys --keyserver hkp://keyserver.ubuntu.com:80
sudo -u asterisk gpg --refresh-keys --keyserver pgp.mit.edu
sudo -u asterisk gpg --refresh-keys --keyserver keyserver.pgp.com

result same everytime:
# sudo -u asterisk gpg --refresh-keys --keyserver hkp://keyserver.ubuntu.com:80
gpg: Fatal: can't create directory '/root/.gnupg': Permission denied

Kindly help!

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@tonyg wrote:

hi, I am currently running FreePBX 14.0.1.1 and have been for about a month. initially when I did the v14 install, I did not have this issue, but at some point about 2 weeks ago, I noticed that when I made changes (any changes, it does not matter what) and then hit the apply button, not only did asterisk reload (as expected) but all my tls connected endpoints dropped. I am using pjsip with sip-tls and srtp. I do have a couple of endpoints (pjsip) that do not support encryption and they do NOT disconnect (UDP). When the endpoints drop, it takes a while for them to re-connect, sometimes as long as 30 minutes, so I now have to batch any changes for nightime (really inconvenient).
To narrow down the issue, rather than clicking on apply, I tried using the 'core reload' command in asterisk. I noticed that core reload does NOT result in my endpoints dropping. I was wondering if the 'apply' button does more than just a 'core reload'? is it possible that it is restarting the firewall? or something else network related that might be messing with the tcp connections?
I would really appreciate any help with this, this is a real problem for me.

i have attached logs of a core reload that does not drop connections, an apply that does drop connections and a modules version list
thanks
reload issue.tgz (1.3 MB)

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@WB3FFV wrote:

I went to download a current copy of 10.x and realized it's now showing as legacy, not stable. Seeing that I have a few questions before I load up a new server.

If 7 is now stable, is there an update path from the 10.x release?

I know I can load FPBX and then turn it into PBXact, so if I load the 7 release, can that be updated to PBXact, or do I need to use Legacy?

Thanks...

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@v4704 wrote:

Hi, I actually never tried to enter this module before so I'm not sure if it was like that out of the box or not.
Anyhow, on the index page, I tried to enter the "Operator Panel" but it sends me to an error page:
"This site can’t be reached
pbx1.vlit.pro refused to connect."

iSymphony is shown as a service on the firewall module (and my IP is set as Trusted Network). Just in case, I also added the port 58080 to the firewall module under "custom services" (and hit save), but it didn't change much.

Is there anything else I need to do to make it work?

P.S. iSymphonyV3 13.0.4.1 is installed and enabled.

Thanks!

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@sredmond wrote:

I've been fighting with this for a week now and I am at a loss of what to do. I'm trying to deploy S500 and S300 phones the S500 are receiving an IP and Configs and just working. The S300 are receiving and IP, then losing it at receiving Configs. They are configured on a Chan_Sip on port 5160 due to 5060 problems with Spectrum internet.

Barracuda Firewall/Gateway
Cisco S300 managed switch
PBXact 100
7 S500 phones
38 S300 phones

During the initial bootup of the phone, it receives its IP address via DHCP and displays it on the phone then moves on to show "bridge mode" then the Mac address. It proceeds to get the updated firmware and its configurations and does a reboot. After the reboot, it will not show an IP address during the bootup but the DHCP server still shows it active. The Network Info page in the phone shows an IP of 0.0.0.0 but the gateway and DNS are correct. If I reboot the phone again, the DHCP server does not show the phone online and everything in the Network Info page in the phone has been reset to zeros.

It only grabs an IP from a factory reset before it does a firmware update.

I decided to bypass the Cisco switch completely and run a dedicated LAN IP from the Barracuda. The S300 phones would receive the IP from the DHCP Server and attempt to register the extension. after attempting to register the extension, it would reboot and move everything in the IP config on the phone back 0.0.0.0 and the phone would show offline on the DHCP server. I have reverted everything back to yesterday's configuration in the Barracuda System since the switch did not resolve the issue.

S300 phones - I have learned I hate them.
- Will not take an IP via DHCP after configurations and attempted registration of the extension
- will not communicate to firewall/gateway with static IP configured on the phone
- will not a reserved IP that is handed out by the firewall/gateway tied to the phones mac address, although the firewall/gateway will occasionally show it online, the phone itself will never show that it has an IP

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@v4704 wrote:

Hi, I'm using FreePBX Distro 13.0.192.9 on Asterisk 13.16.0.
I would like to block incoming calls from callers without caller ID.

What I tried so far is going to Admin > Blacklist > Settings and set:
"Block Unknown/Blocked Caller ID" to: YES
"Destination for BlackListed Calls" to: Hangup

But I still receive calls from unknown callers.

Also, when I changed this setting, the interface never ask me to "apply" changes, so I'm not sure if it's even applied, and there is no entry in the "blacklist" tab.

Any help would be awesome.
Many thanks!

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@Riick370 wrote:

Dragged and dropped files to my USB drive and I get:

dracut-initqueue timeout. could not boot. /dev/root does not exist.

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@Riick370 wrote:

SNG7-FPBX-64bit-1706-1

Booting from USB goes right to step 7:

https://wiki.freepbx.org/display/PPS/Installing+SNG7+Official+Distro

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@natsort wrote:

Earlier it Was possible to see calling history in UCP on a ring group, DID or queue level.
But now history is only related to extensions.

Anyway to give user access to see calling history for an inbound number (DID) and not only Call transfered to an extension?

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@tonyg wrote:

hi, i have a FreePBX 14.0.1.1 install that has been in production for about 2 weeks, we have had some asterisk core dumps. I have a ticket opened with them and they are starting to focus in on libsrtp as the possible problem. i have about 95 extensions and most are pjsip - sip/tls and srtp.

the libsrtp provided with freepbx distro is 1.4.4, this seems a little old. I was wondering, first, is that something that sangoma bundles with the distro or does that come with the OS?

also, is anyone else experiencing core dumps under similar circumstances?

thanks
https://issues.asterisk.org/jira/browse/ASTERISK-27155

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@mmackenzie202 wrote:

We use Free PBX w/ Asterisk for my office. We are cutting over to a new PRI and purchased hours to have some true PBX support on the Turn Up and Test. I do not have a deployment #, I do not have a serial # for the Sangoma card (this was all taken care of by our now out of business IT company) .

Is there any way for me to find the serial number in the PBX Administrator?
Trying to avoid having to open up the phone server to find a number.

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@hshamallakh wrote:

I need to patch the distro, and in the forum they mentioned to use the src rpm file provided, but checking the link i can only find the 13.14 src, 13.15,16,17 are not there at all .. and it has been a while since they are released.

Is there any intentions to release them? As I saw lots of comments about them being requested for sometime now with no updates.

any help will be appreciated.

Kind Regards

Haytham

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@thehammer86 wrote:

When accessing UCP over HTTPS on port 4443, a constant error message pops up. After closing it, it pops up again and again. This problem does not occur when accessing over plain HTTP over port 81. This is on a fresh installation of SNG7 FreePBX 14 Asterisk 13.

See attached.

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@AdHominem wrote:

Just downloaded the very latest Distro (SNG7-FPBX-64bit-1707-1.iso). When I try to install the Distro on my VMWare ESXi machine, I get to a screen that reads "installation summary." Down towards the bottom, under "Installation Destination" I see "no disks selected." I click on it and select the disk, click done, and then see a message that says "failed to save storage destination" and then it returns to "no disks selected." Without a disk to install to, I cannot complete the installation.

Any suggestions?

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@jhelou wrote:

Hi community, FreePBX has been such a great initiative since its early days it's refreshing to see it flourish into the established leader that it is today. I browsed through the blog on FPBX14 which got a facelift and a few improvements over the 13 distro. I am putting the following improvements for debate, I find that they are long overdue to help Admins optimize the use of FPBX and improve user experience:

• Allow admins to link up directories to UCP user contacts. maintaining an internal directory in print is wasteful and inefficient, the 411 option is also inefficient when in a hurry or on call. Why not allow a default directory(s) that can be assigned to user group(s) and that will remain up to date through a direct link to the asterisk DB?

• Create a configurable set of defaults for creating new extensions, often times the defaults that are set to new extensions do not respond to the organization's needs, and often times an organization's needs do not change for different extensions. Why not turn defaults into a configurable set of parameters?

• create a switch to allow everyone to use a certain outbound routes, as it stands right now users have to be explicitly given access to any route otherwose they are blocked by default. This creates a problem for delegating tasks (creating new extensions) to non-technical staff.

I can always make a longer list, and i'm sure depending on everyone's experience the priorities change. I am a strong advocate of open source and i value a lot all the effort that is being put into the development of FPBX. Great job!

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