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Freepbx 14 with PHP7 and mysql 5.7 in Centos 7

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@vijo wrote:

Dear Team,

Whether Freepbx 14 is compatible with PHP 7 and mysql 5.7 in Cnetos 7x ?

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Queue Stats (Queue Wallboard)

RTP packets stop randomly

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@Bert121164 wrote:

Been having issues with random calls where during call the person on the other end can’t hear us but we can always hear them.If the person doesn’t hang up the audio comes back after 30 to 40 seconds.This can happen to any call at random the only thing consistent is we can always hear them. Our sip provider monitored for a while and said that the RTP packets from our end just stops. Has anyone had this issue. Will take any suggestion on what to monitor or check to clear this up.
FreePBX 13.0.194.5

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Asterisk and Salesforce integration

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@aysam wrote:

Hi,
Actually we made a Salesforce click-to-call app that uses ARI in Asterisk 13.
we want to integrate Salesforce with Asterisk using REST API ARI…
I would like to know the difference between chan_sip and chan_pjsip and what is the best between them…

thank you in advance.

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Unable to install freepbx 14 in centos 7

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@vijo wrote:

Dear Team,

When i am trying to install freepbx 14 in cent os 7 with 5.6 and mysql 5.7 i am getting the below error.

**[root@ip-172-31-31-231 freepbx]# ./install

Database engine [mysql]:
Database name [asterisk]:
CDR Database name [asteriskcdrdb]:
Database username [root]:
Database password:
File owner user [asterisk]:
File owner group [asterisk]:
Filesystem location from which FreePBX files will be served [/var/www/html]:
Filesystem location from which Asterisk configuration files will be served [/etc/asterisk]:
Filesystem location for Asterisk modules [/usr/lib64/asterisk/modules]:
Filesystem location for Asterisk lib files [/var/lib/asterisk]:
Filesystem location for Asterisk agi files [/var/lib/asterisk/agi-bin]:
Location of the Asterisk spool directory [/var/spool/asterisk]:
Location of the Asterisk run directory [/var/run/asterisk]:
Location of the Asterisk log files [/var/log/asterisk]:
Location of the FreePBX command line scripts [/var/lib/asterisk/bin]:
Location of the FreePBX (root) command line scripts [/usr/sbin]:
Location of the Apache cgi-bin executables [/var/www/cgi-bin]:
Directory for FreePBX html5 playback files [/var/lib/asterisk/playback]:
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…Yes. Determined Asterisk version to be: 14.7.6
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install…Yes (No /etc/freepbx.conf file detected)
Database Root installation checking credentials and permissions…Error!
Invalid Database Permissions. The error was: SQLSTATE[HY000] [2002] No such file or directory

can anyone help on this?

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Outbound routes trough extention

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@svchost1101 wrote:

Hi.
For some reason my gateway GXW4108 does not connect the trunk. So line with outgoing call ability connected with extention. I guess that I need to add extensions_custom.conf rules for outgoing calls or there are pretty solution with freepbx web gui?

Second question is:
Can anybody help to write dialplan for outbound call?

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PJSIP trunks do not re-register

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@MiMar wrote:

Hi everyone. We are running FreePBX 13.0.194.2 and using pjsip for our trunks. Even though we have max retires set to 0 on the trunks, if we have an issue that prevents registration (an Internet outage for example) the trunks stop attempting to register after a certain number of attempts. When the Internet comes back, the trunks do not try to re-register. Am I missing something here? Is there another setting that needs to be set?

2018-04-16 02:30:15] WARNING[15782] res_pjsip_outbound_registration.c: Maximum retries reached when attempting outbound registration to ‘sip:username@sipserver:5060’ with client ‘sip:username@server:5060’, stopping registration attempt

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Extend time for user to enter extension for transfer

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@snaplink wrote:

Current PBX Version 14.0.2.14
PBX Firmware: 12.7.4-1804-1.sng7
PBX Service Pack: 1.0.0.0

I’m looking for a way to extend the time a user has to enter an extension to transfer a call.

I have a user who is a backup for the main receptionist and received overflow calls. She has had to transfer a couple of call this morning to voicemail using *123 for example and the first time she had to lookup the desired extension but by the time she found it and had already pressed * it errored out. If she dials it all in one sweep it works fine. This was an employee she didn’t have a BLF for.

Is there a way to give her or other users a little longer to enter extensions to transfer before it will error?

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Missed Call Redial

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@casper2018 wrote:

Hellow everyone,

I am new to this forum. I am looking for a solution. I have asterisk 13 and freepbx 13 running on Centos 7. Fanvil phones as sip clients. When we have missed calles on phones, and operator redials the call I whant to replace first 4 digits with one digit. Can anyone suggest any solution?

Thank you

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Voip.ms Registration timed out, trying again

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@knowledgemonster wrote:

Hi,

I have a system running FreePBX 14.0.1.36.
I have 4 sip trunks setup, 3 from Voip.ms and 1 from voipmuch.
The voipmuch line always registers, never have problem with it.
One of the voip.ms line always works but is very lightly used (5 calls per day)
the other 2 voip.ms (about 300 calls per day) lines had lots of problems with registration failing when i first setup (1 month ago) then I set the udp timeout on my nat rules to 120, and tadaa, issue resolved. (mikrotik router) UNTIL… rebooted the freepbx system and then the disconnects started happening again.

I have contacted voip.ms and they say that everything looks fine on their end. nothing wrong with my connection string peer details etc… they say it must be something in my router or network.

I will also add that i have another freepbx system on the same network with 2 lines from voip.ms and never have a problem with those but they are also light use… (maybe 30 calls per day.) Obviously this is urgent so any help from anybody is greatly appreciated.

I realize i have not provided any details so what ever you want to see let me know.
I will also mention that i had another setup where i had similar issue and i replaced his rogers cable modem/router with a tp-link router and hes never had a problem since. (I dont think there is anything wrong with my router because i have other lines which are not used as much work fine all the time, although i do think some settings need to be changed possible due to the 2 lines being used more than the rest)

My problem is pretty much identical to this one: Dropping ip registrations

If i restart asterisk then it stays registerd for 1 - 15 minutes

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Intercom calling, channels get stuck

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@princetech wrote:

FreePBX 14 (Sangoma 7 iso) & Asterisk 13 or 15, fully updated, & yealink phones

Any intercom call, the channel gets stuck post hangup and i have to log into asterisk to “kill” it.

Happening on two separate deployments, any help would be greatly appreciated.

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HELP! Just did the latest update, FreePBX reports "Tampered File"

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@ptravel wrote:

I’m running:
Current PBX Version:14.0.2.16
Current System Version:12.7.4-1803-1.sng7
Total Module Count:118
Enabled:115
Disabled:1

I just ran module update. It finished and I updated the configuration as usual. Now, Dashboard displays this:

Worse, I’ve tried doing restores from recent backups and it reports “error uploading file.”

What do I do?

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Freepbx page is blank

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@vijo wrote:

Dear Team,

I have installed freepbx 14 in cent os 7x.
After installation when am trying to access the web page am getting This page isn’t working

As checked HTTPD service is running.And also i have created an index page in HTML and am able to view that index file.but am not able to access myip/admin

Can anyone help on this?

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Security FreePBX

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@TestAcc wrote:

Hey,

I have installed FreePBX on a server hosted at an external hosting provider, and I would like to know if I did enough to secure the system. If you guys have any tips, tell me :slight_smile:

  • All extensions have a strong password
  • Only the required ports are open, and only to a few IP-adresses (they are static)
  • Backups are made every night
  • Updates are installed just after the backup every night
  • Admin Panel is only reachable from my (static) IP.
  • Voicemails are being send by e-mail
  • Maximum five calls out are possible at the same time

Have a nice day :+1:

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Post call recording script not saving

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@nahakr wrote:

Hi All,

We are having an issue with the post recording script MIXMON_POST.

When we set the value in the Admin GUI to:

/var/lib/asterisk/agi-bin/script.sh ^{MIXMONITOR_FILENAME} “^{AMPUSER}” “^{CALLERID(number)}” “^{CDR(dst)}” ^{CDR(start)} “^{CDR(src)}”

The value is truncated on save to:

/var/lib/asterisk/agi-bin/script.sh ^{MIXMONITOR_FILENAME}

Any ideas?

PS. FreePBX version is 10.13.66-1

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Rename Provisioning Template?

Upgrading from FreePBX 13 to 14 on centos 6.9

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@cDGo wrote:

Hi I have a freepbx 13 running on a centos 6.9 server with php 5.3.3.
When I want to upgrade via the admin control panel, I get 2 errors during pre check.

  1. nodejs is not installed
  2. PHP 5.6 or higher

When installing php 5.6, I get this error
PHP Fatal error: Incompatible file format: The encoded file has format major ID 4, whereas the Loader expects 7 in /var/www/html/admin/modules/sysadmin/agi-bin/LoadLicenseIfExists.php on line 0
Whoops\Exception\ErrorException: Incompatible file format: The encoded file has format major ID 4, whereas the Loader expects 7 in file /var/www/html/admin/modules/sysadmin/agi-bin/LoadLicenseIfExists.php on line 0 Stack trace:

  1. () /var/www/html/admin/modules/sysadmin/agi-bin/LoadLicenseIfExists.php:0

So I do not understand how to do this, in order to upgrade freepbx I need to upgrade the PHP version, but doing so makes the system unusable?

I’ve read quite some postings about this matter, but these make me confused even more.

So what would be the best thing to do?

  1. Upgrade Centos to the latest version 7 and migrate freepbx or do a fresh install?
  2. Update this centos 6.9 some how and keep what’s allready there?
  3. Hire an expert (where?) to let them do it for me?

As one could have expected, I’m no expert on freepbx nor centos, so please consider this if you reply.

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Inbound calls randomly directed to non-reception number

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@lgraham wrote:

Modules upgraded over the weekend, suddenly I have some random behavior where inbound calls go to an emergency extension (an internal extension that routes to an external cell phone) instead of following the call flow (to reception).

CDR log shows ANSWERED on that ring group, but in reality the call went to this external cell phone.

I have gone over the call flow and I cannot find any discrepancy. Can anyone help point me in the right direction?

FreePBX 13.0.194.5

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Freepbx distro 10 to to SNG7

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@Vyspa wrote:

We have been running Freepbx distro 10 on Proxmox for a while without issue and recently decided to upgrade to sng7.

The system has 20 extensions 4 queues. Nothing special.

We fired up a new vm with the same spec and installed the latest distro. This took a while but the forums said it would so we were not concerned.

Now that the unit is running looking at the Load stats things dont seem to be where they should.

I’d expect load to be prety much 0 most of the time especially when we are closed. However this does not seem to be the case. We have uninstalled XMPP and ucp as suggested by others but are still seeing a load average of around 1.1 when there is no usage. We havent even set up any extensions on it yet.
My concern is if the iowait is already high when there is no activity then waht will it be like when we start taking calls and recording them.

Are there some tweaks we need to do to make SNG7 not consume so much resource when supposedly idle?

This is on SNG 7

[root@pbx ~]# sar
Linux 3.10.0-693.21.1.el7.x86_64 (pbx) 17/04/18 x86_64 (1 CPU)

00:00:01 CPU %user %nice %system %iowait %steal %idle
00:10:01 all 3.69 0.00 2.27 8.13 0.62 85.29
00:20:01 all 3.52 0.00 2.22 9.59 0.41 84.25
00:30:01 all 3.54 0.00 2.25 7.56 0.49 86.16
00:40:01 all 3.64 0.00 2.27 7.25 0.31 86.54
00:50:01 all 3.59 0.00 2.27 7.94 0.52 85.69
01:00:01 all 3.44 0.00 2.20 6.58 0.47 87.31

Freepbx distro 10
[root@pbx ~]$ sar
Linux 2.6.32-504.8.1.el6.x86_64 (pbx) 17/04/18 x86_64 (1 CPU)

00:00:01 CPU %user %nice %system %iowait %steal %idle
00:10:01 all 3.71 0.00 2.04 1.10 0.63 92.52
00:20:01 all 3.66 0.00 2.04 0.58 0.72 93.00
00:30:01 all 3.31 0.00 1.83 0.61 0.67 93.58
00:40:01 all 5.73 0.00 2.14 1.43 0.82 89.88
00:50:01 all 3.64 0.00 1.65 0.61 0.70 93.40
01:00:01 all 3.30 0.00 1.53 0.60 0.69 93.89

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Phones Not Updating via Endpoint Manager

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@mikeaph wrote:

When making a change to a template for a S500, I submit the change and the phone reads that it is provisioning but the change is not made. Also, new phones are not provisioning. I had this problem before and I rolled back to an older version of Endpoint Manager and that worked. The system has since been updated and I am having the problem again. I tried rolling back to the oldest available (.12), but it is still not working. Any thoughts?

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