@alep wrote:

Hi, is there some methods to autmate user import from Active Directory without using gui, but only with shell script or something like that (e.g. fwconsole) ?

Thanks

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@phagez wrote:

After a fresh install of ubuntu 14.04 and FreePBX 13 I get the "Can not connect to Asterisk" red box. I was following this guide: http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+Ubuntu+Server+14.04.2+LTS

fwconsole restart from a root console fails connecting to asterisk however if i just start asterisk with asterisk -cvv -U asterisk -G asterisk then everything works.

I assume this is a simple issue but im very new and there isnt much documentation around for fwconsole yes and as such im not sure what to be looking at.

Is there a startup script that freepbx/fwconsole uses to start asterisk that i could look at?

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@claloano wrote:

In Italian it is called " corner unit "

The phone corner is defined in Italy 's Extension Ringing rings when the target phone to the one corner

Come you say in English this What?

You can set up FreePBX ?

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@dnadih wrote:

Hello, recently I discovered that I cannot upgrade or install any new module in FreePBX 12.0.76.2 due to GPG verification check. I followed recommendations in similar posts, refreshing keys.. etc.
Current situation is this;
running gpg --verify (path to..)/admin/modules/cache/queues-12.0.21.tgz.gpg_
I get this:
gpg: Signature made Tue 01 Mar 2016 10:07:20 PM CET using RSA key ID 69D2EAD9
gpg: Good signature from "FreePBX Mirror 1 (Module Signing - 2014/2015) "
but running amportal a ma download queues gives me:
Downloading 318769 of 318769 (100%)
The following error(s) occured:
_ - File Integrity failed for ../admin/modules/_cache/queues-12.0.21.tgz.gpg - aborting (GPG Verify File check failed)_
The same thing happens with upgrading framework or any other module...
I would very much like to avoid reinstalling FreePBX if possible...
Thanks!

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@sumitk wrote:

Hello,

I have a 8 port fxo device which is configured with my Freepbx as a trunk. all 8 fxo ports have analog lines which is connected on another legacy PBX system.
I am able to dial in two stage like first i dial 6102 that picks a FXO line and then i have to dial ext number. Is there any way to make it dial directly.. like 6102extno.

below is trunk configuration -

Trunk Name: 6102
PEER Details:
username=6102
type=peer
secret=nav6102
qualify=60
host=dynamic
context=from-pstn

FXO is running in one stage mode.

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@jdmage_mx5 wrote:

I have a FreePBX install running. I have had a couple of test endpoints running for a few weeks now just to test calling in/out etc. Everything seemed good so I deployed it across our school district the other night. Everything is going pretty good but I am having one HUGE problem. Randomly through out the day all new calls STOP. External can call into an IVR and get the IVR. I found out I can run service asterisk restart and within a minute everything comes back up.

I am fairly new to FreePBX and Asterisk and have no clue on the best place to start troubleshooting. Can anyone lend me some advice?

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@jimgb17 wrote:

Hi all
I am trying to install freepbx latest freepbx from usb i downloaded the latest usb img from the site and flashed the img to my systems drive but the drive wont been from the usb drive
i have tested with laptop and my intel nuke.....

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@normanghenderson wrote:

Moved as requested from Issue 12207:

Initially it appeared that libmyobdc has been removed from Ubuntu 16.04, however this isn't really the case, but there are a number of version issues (see #12207 for details). The original point was to get FreePBX 13 to work with the current stable LTS release of Ubuntu.

When trying to do anything in the FreePBX WebGUI, there were warnings that "module SimpleXML is already loaded".

Therefore, I commented out "extension=simplexml.so" from /etc/php/5.6/mods-available/simplexml.ini and restarted apache2. The Web GUI now works, until I try to "Reload Config" and then I get an error:
PHP Fatal error: Call to undefined function FreePBX simplexml_load_file() in /var/www/html/admin/libraries/BMO/Hooks.class.php on line 84
Whoops\Exception ErrorException: Call to undefined function FreePBX
simplexml_load_file() in file /var/www/html/admin/libraries/BMO/Hooks.class.php on line 84
Stack trace: 1. () /var/www/html/admin/libraries/BMO/Hooks.class.php:84

I suspect that extension=simplexml.so is in fact required, so the question is how to get rid of the warning (which obscures the whole GUI) that "module SimpleXML is already loaded".

Help please...

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@brianccampbell wrote:

Using FreePBX distro 10.13.66-14 and trying to migrate the configuration of an existing FreePBX install from one deployment ID to another Deployment ID. In this particular case the original deployment was virtual and the new one is running on Sangoma hardware (that was 50% off last week what a great deal!). I don't simply want to switch everything from the original deployment ID over to the new one and have found that apart from system admin (which was licensed with the hardware already) and endpoint manager (which I don't mind buying again for this new deployment ID) there really aren't a lot of commercial modules I want/need.
My question is this: with the backup and restore module, is there a way that I can backup my existing configuration (extensions, users, routes, trunks, etc,) and restore on the new hardware without bringing over the original deployment ID and without bringing over anything that will expect to see a commercial module that I don't plan on licensing on the new Deployment ID? The original deployment was an OSS one so almost all commercial modules are currently licensed. In theory I would like to have both deployments running (on different ip addresses and trunks only registered on the original) so that I may have time to test and perform a planned cut at a specific time. My fear of the unknown is that I will end up restoring a section of the backup from the original deployment ID instance on the new deployment ID and have the original deployment ID transferred over and require immediate activation. I already re-registered each of these deployment ID's previously so I don't have any Zend resets available through the system admin>activation section.

Thanks,

Brian

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@Msh wrote:

Hi folks,

I'd be more than happy of someone describes, what is the process of adding a new item to Extension General Settings. I've planning to add a call restriction field to my SIP extensions, so admins can restrict call duration for specific extensions.

I searched but didnt find anything in the GUI, thought it might be an idea to add it to the GUI but need some info on contribution rules and methods.

Sorry if my question does not comply with forum rules.

TIA
MSH

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@mppkll wrote:

So I attempted to sync User Manager to AD and discovered several things that make it not ready for prime time, at least for me. Does anyone know how to remove the cached users that where pulled in by AD sync?

So far I have tried changing the LDAP server info and clicking submit to cause a re-sync and changing to local "FreePBX Internal Directory" then back (clicking submit each time) yet the users and groups still remain even when invalid server info is entered for the AD server.

in case any developers see these are the minimum things that I think need to be added to make this feature useable.

1. Users pulled from AD should sync to iSymphony
2. There should be filter option to the LDAP query so not everything in the BaseDN gets pulled (i.e. service accounts, admin accounts, etc.)
3. Only specified groups should be pulled in, not just any group in the BaseDN with no way to control them.
4. There needs to be a manual way to delete unwanted users/Groups.

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@sbrauss wrote:

Hi!

I today wanted to configure a FTP backup server.
It basically works, but I have problems with setting the path:

If I leave it empty, output is as follows:
August 10, 2016, 4:52 pm - data: Creating directory '/Daily_Full_Backup'
August 10, 2016, 4:52 pm - Directory '/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 4:52 pm - data: Directory '/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 4:52 pm - ftp_mkdir(): Create directory operation failed.
August 10, 2016, 4:52 pm - data: ftp_mkdir(): Create directory operation failed.

Strangely, it adds a '/', so I think it is correct for the ftp server to deny creating /Daily_Full_Backup.
Then, I tried setting it to he full path, in my case /home/voip, but I get:

August 10, 2016, 4:48 pm - Creating directory 'home/voip/Daily_Full_Backup'
August 10, 2016, 4:48 pm - data: Creating directory 'home/voip/Daily_Full_Backup'
August 10, 2016, 4:48 pm - Directory 'home/voip/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 4:48 pm - data: Directory 'home/voip/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 4:48 pm - ftp_mkdir(): Create directory operation failed.
August 10, 2016, 4:48 pm - data: ftp_mkdir(): Create directory operation failed.

Again, strangely, it now removes the '/', and I think the FTP server correctly denies to create.

So I created the folder daily_backup in the FTP home directory of user voip, and set the path to /daily_backup.
Now, the output is:
August 10, 2016, 5:03 pm - Creating directory 'daily_backup/Daily_Full_Backup'
August 10, 2016, 5:03 pm - data: Creating directory 'daily_backup/Daily_Full_Backup'
August 10, 2016, 5:03 pm - Directory 'daily_backup/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 5:03 pm - data: Directory 'daily_backup/Daily_Full_Backup' did not exist and we could not create it
August 10, 2016, 5:03 pm - ftp_mkdir(): Create directory operation failed.
August 10, 2016, 5:03 pm - data: ftp_mkdir(): Create directory operation failed.

Has someone an idea what is wrong?

Thanks,
Stephan

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@larzeb wrote:

FreePBX 13.0.163
Asterisk 13.10.0

Suddenly I cannot make any outbound calls on any extension. A log snippet from the attempted outbound call through Flowroute:

[2016-08-09 14:34:42] WARNING[21362][C-00000006]: app_dial.c:2432 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/300-00000006", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack

I attempted to get Module updates, but received the following message:

Warning: Cannot connect to online repository(s) (http://mirror1.freepbx.org,http://mirror2.freepbx.org). Online modules are not available.

Any ideas on what might correct this?

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@jhayes wrote:

When attempting to upload a MP3 or Wav file via the Music on Hold module we are receiving an error.

RuntimeException
Failed to convert /var/spool/asterisk/tmp/temp.1470850251933.sln48 to /var/lib/asterisk/moh/mr_ups_60.wav! Command 'file convert /var/spool/asterisk/tmp/temp.1470850251933.sln48 /var/lib/asterisk/moh/mr_ups_60.wav' failed.
File:/var/www/html/admin/libraries/media/Media/Driver/Drivers/AsteriskShell.php:151

I can manually place the file in the /var/lib/asterisk/moh directory and run "/usr/bin/sox File.wav -r 8000 -c 1 -s File1.wav -q" and the file works without issue. I am curious if there is some reason, that I am missing, why the upload process isn't working as it is intended.

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@rbamburg wrote:

Have an Installation that uses a phone # ending in xxxxxxxx30 to register the SIP trunk connection.
When I setup call forwarding on a phone, the system uses xxxxxxxx07 to attempt the SIP trunk registration which the provider kicks back as forbidden.

The xxxxxxxx07 # is the caller ID for the client but xxxxxxxx30 is the # that has to be used for registration.

Calling from the Extension works, just not when forwarded. Both attempts use the same trunk definition as we only have one outbound route.

Any Ideas as to what I need to set to resolve this??

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@vvehs wrote:

Hello

I use FreePBX 10.13.66-14, asterisk 13.10.0 and iSymphony.

I use 6 queues with static members round-robin with memory and enabled autofill, 3-9 agents.
Extensions use CISCO hardphones or X-Lite.
Customer come to the queues from IVRs.

I use iSymphony to monitor the queues, so I can see state of extensions and all calls in Queue in process and waiting.

Mostly Queues work well, but sometimes (3-4-5 time a day) we see the following:
There are calls in the Queue,
There are available agents in the same Queue, they are really available in the screenshot you can see, they have white circle, this means they are registered and free.
But extensions don't ring, even iSymphony shows that they don't ring ( iSymphony shows yellow circl when extension rings)

imgpsh_fullsize (1).png2339x718 60 KB

And extensions are really registered and free. I can see them with asterisk -x 'sip show peers'

I didn't find anything bad in logs,

After some time it just starts ringing correctly.

I tried to change:
RingStrategy
FollowMe .... disable/enable - some of our agents use it, but I tried to disable it nothing changed

I tried to set "Queue state detection" in extension module to "Ignore State" and it looks like it helped, but we got about 5 asterisk segfault a day so I had to revert it to "Use state".

I saw posts with similar issue but they are too old and no solution has been found.

--------------------- UPDATE ----------------------
FreePBX is installed in HyperV virtual server
HyperV drivers are installed

And a Interesting thing:
1)I see stacked call in queue, in the queue there are free agents but they don't ring
2) I call to any agent from the Queue - just 1 ring abd drop
3) The Queue starts working properly
- But it helps not in all cases.

If you need any other information or test .. I will provide.

Best regards.

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@Ten66 wrote:

I have FreePBX 13.0.163
I am trying to use a VoipStunt SIP trunk.
It works but it only works if I match the correct IP "host" address, they, voipstunt, seem to use two different ones.
77.72.169.134 and 77.72.169.129
I have had a look around here and the web and either I have missed something or I have not understood, unsure right now.
I have tried putting in two host entries and Freepbx made it look like this; " host=77.72.169.134&77.72.169.129", it only worked for one of these unfortunatly.
I have tried to make two separate trunks, one for each but one of them always stays disabled.
I have spent a fair while on this and have had to use restore a few times when I did something to annoy the system but that has been very good at getting me back to where I was.

I get an incoming call, it either says "rejecting unknown SIP connection from ..........
or it works. it depends if I have the correct ip address in the incoming host entry.
problem, I need both ip addresses unless you know better.
I don't want to use anonymous it appears. I only have a domestic set up so no fancy kit to play with.

I hope this makes sense and if I have missed something really obvious, sorry, any information I need to give to help, please ask me, thanks.

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@donker wrote:

Hi everyone,

I've Googled on "voipbusterpro" in combination with FreePBX or Asterisk, but to no avail. So this would be the first hit for the next one to face this issue. So Voipbusterpro is only an outgoing Voip provider. Setting up the outgoing trunk I'm getting stuck and has left me confused. I spent about a day on this, so I've tried a bit already.

Situation: I'm behind a firewall with the latest FreePBX and Voipbusterpro does not wish to authenticate.

1. I've routed UDP for ports 6050, 6051, 4569, 10000-30000 to my FreePBX

Then for the Trunk I've done the following:

1. Trunk name: voipbusterpro

2. PEER details:
   disallow=all
   allow=ulaw
   context=from-trunk
   dtmfmode=auto
   fromdomain=sip.voipbusterpro.com
   fromuser=username
   host=sip.voipbusterpro.com
   insecure=port,invite
   qualify=yes
   secret=password
   type=peer
   username=username
   authname=username
   canreinvite=no

3. Register String:
   username:password@sip.voipbusterpro.com/**username**

I'm having enormous difficulties keeping the trunk alive after I update due to registration timeouts and I cannot seem to be able to route calls over this trunk, continuously running into service not available (noservice).

Any ideas? Has anyone gotten voipbusterpro to work with FreePBX?

Best,

Peter

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@WLS_IT_Guy wrote:

Yesterday afternoon I upgraded some modules and then applied the above update as well as 10.13.66-14. Since then 4 digit extension calls work but nothing incoming or outgoing?

Not a linux guy but I can follow directions. I had this issue once before but I called support and they fixed it rather quick.

I also noticed on my dashboard this error:

Unable to write to /etc/wanpipe/global.conf

Thanks in Advance,
Jeff

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@amdr wrote:

Hello everyone
I tried to install addons call center,
but I get this message

how can I solve that?

Unable to read primary_db from repo: elastix-extras

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