@wa4zlw wrote:

I've enabled CID Superfecta on all my INBOUND routes with the DEFAULT scheme (thats all I have). Debug mode work and I get an email sent to me (adding some sources that handle spam will break this, right now here is what I have set to YES

When I dial into a route that goes directly to VM no email is generated. any ideas whats going on?
Leon

  	  	Data Source Name 	Description 	Enabled
		Asterisk Phonebook 	Searches the built in Asterisk Phonebook for caller ID information. This is a very fast source of information.

If you are caching to the phonebook, this should be the first lookup source. Yes No
OpenCNAM https://www.opencnam.com This data source returns CNAM data for any NANPA phone number (any number that starts with +1). Yes No
TrueCNAM truecnam.com lookup module can pull CNAM, stored CNAM and spam score. Yes No
Telco Data http://www.telcodata.com - These listings are generally only return the geographic location of the caller, not a name. Yes No
FccComplaints https://opendata.fcc.gov - This module checks for complaints to the fcc against a given phone number Yes No
Who Called https://whocalled.us - Caller ID Name and SPAM report listings as provided by the whocalled service. Authentication is optional, and is configured by clicking on this data source. Yes No
Send to email This module will send a notification email for all inbound calls to the user supplied email addresses. Yes No

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@sam_khan wrote:

Hi

Hope you all will be fine. I am seeing Recording Missing in call recording menu although recording is present in monitor folder.

i did found a thread which posted a solution but no luck still same issue.

.......................................................................................................................................................

in the directory /var/www/html/modules/monitoring/libs y edit paloSantoMonitoring.class.php

search for "private function _rutaAbsolutaGrabacion($file)"
and change:

$basedir = DEFAULT_ASTERISK_RECORDING_BASEDIR.’/’;

to

$basedir = ‘/var/spool/asterisk/monitor/’;

and now looks like:

private function _rutaAbsolutaGrabacion($file)
{
$basedir = ‘/var/spool/asterisk/monitor/’;

Restart http server and the recordings working good in the web admin

......................................................................................................................................................

Please Help !

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@parny wrote:

I've successfully used the migration script to move our PBX from Elastix to FreePBX. Yay!

I ran into a couple of minor problems.

My old system was running Elastix 2.5, which was running on top of FreePBX 2.11.10 r17.

I installed the new system using FreePBX-SNG7-FPBX-64bit-1707-1 on a bootable USB key. I had purchased the System Builder Starter Bundle and the Endpoint manager before installing and activating the new system.

At first the FreePBX install wouldn't accept incoming calls. Turns out Elastix doesn't require an inbound route; once I created a simple inbound route (connectivity/inbound routes) incoming calls worked.

Then I couldn't make outgoing calls. I discovered each extension needs to be given permission to use each outbound route: connectivity, outbound routes, choose an outbound route, additional settings tab, move extensions you want to use the outbound route from blocked extensions on the right to allowed extensions on the left.

And paging/intercom didn't work. We have paging groups for each extension that let you dial that extension and start talking without the other end having to pick up. Worked in Elastix, didn't work in FreePBX. Took a while to figure this out. Applications, paging and intercom, choose a paging group. Look at "busy page group" -- all our paging groups were set to "Valet" -- when I changed that setting to "do nothing" intercom paging worked again.

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@claloano wrote:

I updated a FreePBX 13 following the usual procedure but something went wrong

Now asterisk does not work I tried restarting but error:

[Root @ auto-q ~] # amportal restart

Please wait ...

Amportal is depreciated. Please use fwconsole !!!!
Forwarding all commands to 'fwconsole'
Asterisk not currently running
Running FreePBX shutdown ...

RestApps Server is not running
UCP Node is not running

[Exception]
Unable to locate the FreePBX BMO Class' Userman'A required module could be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install userman 2) fwconsole ma enable userman

Restart [-i | --immediate] [args1] ... [argsN]

Stopping fail2ban: [OK]
Ensuring logfiles are presentStarting in fail2: [OK]

Error while updating FreePBX 13

I updated a FreePBX 13 following the usual procedure but something went wrong

Now asterisk does not work I tried restarting but error:

[Root @ auto-q ~] # amportal restart

Please wait ...

Amportal is depreciated. Please use fwconsole !!!!
Forwarding all commands to 'fwconsole'
Asterisk not currently running
Running FreePBX shutdown ...

RestApps Server is not running
UCP Node is not running

[Exception]
Unable to locate the FreePBX BMO Class' Userman'A required module could be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install userman 2) fwconsole ma enable userman

Restart [-i | --immediate] [args1] ... [argsN]

Stopping fail2ban: [OK]
Ensuring logfiles are presentStarting in fail2: [OK]

If I go to the gui I have other errors, I attach screen capture

Whoops \ Exception \ ErrorException (E_ERROR)
HELP
Call to a member function getUserByUsername () on a non-object
/var/www/html/admin/modules/userman/Userman.class.php
* @param string $ username The User Manager Username
* @return bool
* /
Public function getUserByUsername ($ username, $ directory = null) {
If (! Empty ($ directory)) {
$ User = $ this-> directories [$ directory] -> getUserByUsername ($ username);
} Else {
$ User = $ this-> globalDirectory-> getUserByUsername ($ username);
}

Return $ user;

pbx.png4480x1440 295 KB

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@wa4zlw wrote:

This has been a snit with me for a long time and wondering what is causing this:

Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
90701/ekhsquj8 (Unspecified) D No No A 0 UNKNOWN
90703 (Unspecified) D No No A 0 UNKNOWN
99701 (Unspecified) D No No A 0 UNKNOWN
99703 (Unspecified) D No No A 0 UNKNOWN

The first one just showed up the other day. 99701 and 99703 showed up originally. 701 and 703 are real extensions using PJSIP. I have no idea with the 9070x came from either especially with that userid that looks like it iwas randomly gneerated.

while we're on it under PJSIP there's this:

Endpoint: dpma_endpoint Unavailable 0 of inf

I'm assuming thats for provisioning of some sort?
Thanks leon

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@wa4zlw wrote:

I've noticed that the MWI indicators either don't show up or is very delayed based on my experience with older versions. I have my Grandstream devices set to subscribe for MWI. ANy idea why it takes so long now?

Thanks leon

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@paulv wrote:

Thanks in advance for your assistance. I am a hobbyist/end user, not a tech guy.

I have the current FreePBX newly installed and up and running in my small office. I am trying to replace Grandstream GXP2000 phones with GXP2130 phones,

If I enable BLF on the programable keys of the GXP2130, the BLF function seems to work in that the soft keys accurately show the status of the other lines and dialing out works fine. However, incoming calls (from internal or external phones) fail and the error message says "the user at extension 101 is on the phone.". As soon as I disable BLF, the problem disappears and incoming calls ring the phone just fine.

The "enable call waiting" option in "Advanced Settings" of the extension doesn't seem to do anything.

Any tips will be greatly appreciated. Thanks again.

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@wa4zlw wrote:

on older versions of FPBX there was a refresh button on the Asterisk Info page. Why was it removed?

Thanks leon

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@fttx wrote:

Hi,

i want to create two different contexts to insulate two different group of extensions, so that the extensions from group A can't call the one of group B and vice-versa.
But at the same time, i want to keep all the features inside from-internal, like voicemail, parked calls, etc.
In other words is it possible to create two different instances of from-internal?

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@gatozgz wrote:

Hello good morning,

In another times I have used the Cisco VG224 gateways with FreePBX without problems but yesterday I found a problem.

The problem is that you cant define a username with less than 4 digits.

Example(VG224 config.):
...
voice call send-alert
voice rtp send-recv
...
voice service voip
allow-connections sip to sip
redirect ip2ip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
sip
...
voice-port 2/0
cptone ES
station-id name 100
station-id number 100
caller-id enable
!

voice-port 2/1
cptone ES
station-id name 4000
station-id number 4000
caller-id enable
!
...
dial-peer voice 200 pots
no service stcapp
destination-pattern 100
port 2/0
authentication username 100 password mipassword100
!
dial-peer voice 201 pots
no service stcapp
destination-pattern 4000
port 2/1
authentication username 4000 password mipassword4000

dial-peer voice 100 voip
destination-pattern 0.T
no modem passthrough
session protocol sipv2
session target ipv4:192.168.66.24:5060
session transport udp
dtmf-relay rtp-nte
codec g711alaw
fax-relay ecm disable
fax rate disable
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
...
sip-ua
retry invite 1
retry response 3
retry bye 3
retry cancel 3
retry register 3
timers trying 1000
registrar ipv4:192.168.66.24:5060 expires 120
sip-server ipv4:192.168.66.24
!
...

In this example I can define a dial-peer for 4000 extension without problems but I cant define a dial-peer for 100 extension because the username is less than 4 digits.

Anyone knows if there is any solution to solve this so that I can use the actual extensions of 3 digits with my VG224 and FreePBX?

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@mitterhuemer wrote:

Hello,
does anyone know if the new Grandstream phones will be added to epm?

We need to include the DECT System DP750/DP720
and also the Analog Adapters
HT801/HT802

But in EPM there are only old Phone models.

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@danemgroup wrote:

Cant Access GUI. PBX working fine.

wget -qO- http://localhost | grep FreePBX doesnt give any output.

service httpd status gives this error

Redirecting to /bin/systemctl status httpd.service
● httpd.service - The Apache HTTP Server
Loaded: loaded (/usr/lib/systemd/system/httpd.service; enabled; vendor preset: disabled)
Active: failed (Result: exit-code) since Fri 2017-09-01 12:00:34 +03; 16min ago
Docs: man:httpd(8)
man:apachectl(8)
Process: 31370 ExecStop=/bin/kill -WINCH ${MAINPID} (code=exited, status=1/FAILURE)
Process: 31368 ExecStart=/usr/sbin/httpd $OPTIONS -DFOREGROUND (code=exited, status=1/FAILURE)
Main PID: 31368 (code=exited, status=1/FAILURE)

Sep 01 12:00:34 danemqtrmain httpd[31368]: (98)Address already in use: AH00072: make_sock: could not bind to address [::]:80
Sep 01 12:00:34 danemqtrmain httpd[31368]: (98)Address already in use: AH00072: make_sock: could not bind to address...0.0:81
Sep 01 12:00:34 danemqtrmain httpd[31368]: no listening sockets available, shutting down
Sep 01 12:00:34 danemqtrmain httpd[31368]: AH00015: Unable to open logs
Sep 01 12:00:34 danemqtrmain systemd[1]: httpd.service: main process exited, code=exited, status=1/FAILURE
Sep 01 12:00:34 danemqtrmain kill[31370]: kill: cannot find process ""
Sep 01 12:00:34 danemqtrmain systemd[1]: httpd.service: control process exited, code=exited status=1
Sep 01 12:00:34 danemqtrmain systemd[1]: Failed to start The Apache HTTP Server.
Sep 01 12:00:34 danemqtrmain systemd[1]: Unit httpd.service entered failed state.
Sep 01 12:00:34 danemqtrmain systemd[1]: httpd.service failed.
Hint: Some lines were ellipsized, use -l to show in full.

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@carriba wrote:

Tried everything what I've found in this forum and elsewhere, however do not seem to find the right answer yet in order for me to resolve the problem:

Trying to install the latest downloaded version of FreeBPX 14 for my Raspbian Jessie box.

Asterisk 13.15 is installed and running fine, as can be demonstrated:

# asterisk -rx "core show version"
Asterisk 13.15.0 built by root @ raspbx on a armv6l running Linux on 2017-04-09 17:31:05 UTC
# asterisk -rx "core show uptime"
System uptime: 2 hours, 34 minutes, 36 seconds
Last reload: 2 hours, 34 minutes, 36 seconds
# ps aux | grep asterisk
root      2501  0.0  0.0   1908   112 pts/0    S    13:40   0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk  2504  0.1  2.1  37852 21104 pts/0    Sl   13:40   0:05 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
root      2720  0.0  0.1   4276  1952 pts/0    S+   14:57   0:00 grep asterisk

I did launch at beforehand, the command "/usr/src/freepbx/start_asterisk start" to startup the Asterisk application.

However, when firing the command "/usr/src/freepbx/install -n", I'm getting the following output:

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...Error!
Error communicating with Asterisk.  Ensure that Asterisk is properly installed and running as the asterisk user
Asterisk appears to be running as asterisk
Try starting Asterisk with the './start_asterisk start' command in this directory

Above output does not really point to me the potential root cause

The sitaution is reproducable even when I re-install the OS and Asterisk from scratch again.

By means, anybody have some pointer here what to look further (instead of reverse engineer the PHP install code)?

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@doronazl wrote:

im sorry if i have a newb questions but couldnt find an answer for this one on google.
i got centos with freepbx 13.12.1 on it, when i go into asterisk -rv to see the live logs
this what happens
whenever i make calls, all the calls are going through just fine, incoming and outgoing, but the logs are flooded with the same errors
pbx_functions.c:608 ast_func_read: function PJSIP_HEADER not registered.
now im happy that everything is working correctly but should i ignore these ?
i'd rather keep the logs clean.

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@doronazl wrote:

cause 20
i started getting this error today
i got a goip gsm gateway connected to a voip server i got running on my pc
ports are forwarded
everything worked fine then it started giving me this error on outgoing calls
calls between extensions go through just fine
incoming calls from outside works great as well
just when calling out

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@madav wrote:

Hi,

I just ran a whole series of upgrades through the System Admin Pro, I started at 5.211.65-1 and am now at 6.12.65-32.

System Admin Updates says I am up to date but I would like to get to 13 or 14.

I don't see the option on my dashboard to upgrade to 13.

All my modules are totally up to date and when I try to upgrade via the CLI I get the below error:

[root@phone upgradescripts]# amportal a ma upgrade framework

Fetching FreePBX settings with gen_amp_conf.php..

Found module locally, verifying...Redownloading
Redownloading
Redownloading
Downloading 9676048 of 9676048 (100%)
Downloading 9676048 of 9676048 (100%)
The following error(s) occured:
 - Cant Verify downloaded module /var/www/html/admin/modules/_cache/framework-13.0.192.16.tgz.gpg. Unable to trust GPG Key - aborting (Cause: Unable to start GPG, the command was: [/usr/bin/gpg --homedir /home/asterisk/.gnupg --no-permission-warning --keyserver-options auto-key-retrieve=true,timeout=5 --status-fd 3 --list-keys 2016349F5BC6F49340FCCAF99F9169F4B33B4659])
[root@phone upgradescripts]# sudo -u asterisk gpg --refresh-keys --keyserver pool.sks-keyservers.net

I tried

[root@phone upgradescripts]# sudo -u asterisk gpg --keyserver pgp.mit.edu --recv-key 3DDB2122FE6D84F7
gpg: requesting key FE6D84F7 from hkp server pgp.mit.edu
gpg: key FE6D84F7: "FreePBX Mirror 1 (Module Signing - 2016/2017) <security@freepbx.org>" not changed
gpg: Total number processed: 1
gpg:              unchanged: 1

[root@phone upgradescripts]# sudo -u asterisk gpg --refresh-keys --keyserver pool.sks-keyservers.net
gpg: refreshing 3 keys from hkp://pool.sks-keyservers.net
gpg: requesting key 69D2EAD9 from hkp server pool.sks-keyservers.net
gpg: requesting key B33B4659 from hkp server pool.sks-keyservers.net
gpg: requesting key FE6D84F7 from hkp server pool.sks-keyservers.net
gpg: key 69D2EAD9: "FreePBX Mirror 1 (Module Signing - 2014/2015) <security@freepbx.org>" not changed
gpg: key B33B4659: "FreePBX Module Signing (This is the master key to sign FreePBX Modules) <modules@freepbx.org>" not changed
gpg: key FE6D84F7: "FreePBX Mirror 1 (Module Signing - 2016/2017) <security@freepbx.org>" not changed
gpg: Total number processed: 3
gpg:              unchanged: 3


[root@phone ~]# amportal a ma download framework

Fetching FreePBX settings with gen_amp_conf.php..

/var/lib/asterisk/bin/freepbx_engine: line 493: 31008 Killed                  $AMPBIN/module_admin $3 $4

Not sure what to try next. Thanks in advance for any tips.

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@jeelani wrote:

Hi there,
I have FreePBX version is 2.10.1.2 installed on my server. I want to have two different custom template one for SNOM300 and one for SNOM710. But when I check in the "EPM Advance Settings" and "Product Option/Configuration Editor" I can see the base templates for snom300.... and snom710...., here for snom300 I have some special settings and I want to have different settings for snom710. I can see here as one can customize the base template and save it to the database but I have no idea how to call/use these database saved template for the snom710 phones because my base templates settings for snom300 are being pushed in the files namely "general.xml" and "general_custom.xml" in the tftp folder and the snom$model.xml file has the settings like <?xml version="1.0" encoding="utf-8"?>




and if I try to create a different file name for snom710 then its not pushing the settings in the files which I created for as "snom710_general.xml" and "snom710_general_custom.xml".

Really I want to have two different files.
can someone help me with this thing.

Thanks.

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@Tuman007 wrote:

I install module Folow me and it gives an error
Freepbx 13 debian 8.8

exit: 255
Unable to continue. SQLSTATE[HY000]: General error: 1267 Illegal mix of collations (utf8_general_ci,IMPLICIT) and (utf8_unicode_ci,IMPLICIT) for operation '=' in /var/www/html/admin/modules/findmefollow/Findmefollow.class.php on line 1145

0 /var/www/html/admin/modules/findmefollow/Findmefollow.class.php(1145): PDOStatement->execute()

1 /var/www/html/admin/modules/findmefollow/functions.inc.php(79): FreePBX\modules\Findmefollow->getAllFollowmes()

2 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): findmefollow_get_config('asterisk')

3 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks('asterisk', Array)

4 {main}

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@doronazl wrote:

some weird issue i got here
i have a freepbx server on my pc
it has one extension , phone that is connected to the same router as the computer with the freepbx
and in addition theres a goip gsm gateway sitting there too

im currently in another country, with my phone which has zoiper on it set to work with the 2nd extension i have.
now if i call between extensions i have no problems, i can hear the tone when ringing, can speak on the phone normally, all is well
but when dialing out or getting calls to the gsm sim card which is in the gsm gateway, the calls are getting through but i dont get sound whatsoever, no ringing tone or voices.
now when i was at home next to it, on the same network, it worked just fine, gsm calls and everything sounded perfect
now when im out it giving me hell
whats weird for me is that if this was a port forwarding problem the extension wouldnt work neither right ?
in any case i tried DMZ to my server just to see if it makes a difference, same same
where i go from here ?

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@WWEPBX wrote:

Hi,

We have been using the cloud hosted solution for a number of weeks without many issues (the odd teething issues to start). As of last week we have started experiencing issues with people not being able to hear callers, but the callers can hear the user. We are using a Technicolor router in the office.

The firmware version is showing as:
PBX Firmware:
10.13.66-12

As we are using the cloud solution, i am unsure what NAT policies we can put in place internally to address this (if any) Has ayone else experienced the same issues with the cloud solution?

We have also experienced issues whereby an inbound call will come in fine, this will then be transfrerred to another extension. The next inbound call then appears to disconnect the previously transferred call if still active.

Any recommendations would be greatly appreciated.

Thanks

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