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Click a button to dial a number (different number), play a recorded message, then ring off

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@dan_ce wrote:

Basically I’d like to be able to have staff have the option to click a button on our online portal to let customers know their lawnmowers are repaired.
We have a lot of elderly customers who don’t have mobiles so we can’t do SMS.

I spotted this thread which looks VERY similar to what I want to do, but not quite.

I have implemented and tested a call file which can phone a number and play a recorded message, this is done, but what I haven’t sorted is the varying number (the call file number is hard coded) OR the click from the staff portal.

Also, the @ringding code at the thread above isn’t complete, is it? At the very least it lacks a closing php tag…?

Anyway I might be barking up the wrong tree with that approach, it just looked 75% close to what I want to do! :slight_smile:

Any pointers most welcome. I had fun playing with the call file and getting that working but it seems a bit…static at the moment!

Thanks

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System Intermittently Cannot Make or Receive Calls

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@iansatterwaite wrote:

Free PBX 14.0.13.4, Asterisk 13.26.0

Phone system is seemingly randomly going down and is not able to make or receive calls. Timing seems completely random, some weeks we go all week with no issue, other times it goes down 3+ times in a day. Dashboard shows everything is fine, but no calls go through, outgoing just sits with no noise, incoming immediately goes to fail over number. No idea where to look to even begin troubleshooting.

Call Log of Unsuccessful Call


Normally the log is 13 items, the failed calls do not have any BRIDGE items

Any advice on how to solve is appreciated.

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Asterisk is not connected Warnmail on Reboot

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@comfine wrote:

Hi All,

some of our Systems got shutdown every night cause we backup the VM and need to shutdown therefor.
While this is happening we got 4 warnmails with the following content per reboot:

Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 242 Stack trace: 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:242 2. AGI_AsteriskManager->send_request() /var/www/html/admin/libraries/php-asmanager.php:591 3. AGI_AsteriskManager->Command() /var/www/html/admin/libraries/php-asmanager.php:1653 4. AGI_AsteriskManager->parseAsteriskDatabase() /var/www/html/admin/libraries/php-asmanager.php:1622 5. AGI_AsteriskManager->database_show() /var/www/html/admin/libraries/php-asmanager.php:210 6. AGI_AsteriskManager->LoadAstDB() /var/www/html/admin/libraries/php-asmanager.php:1708 7. AGI_AsteriskManager->database_get() /var/www/html/admin/modules/timeconditions/bin/schedtc.php:28

This is annoying and of course this warning should just be send out if asterisk is not connected WITHOUT rebooting. I red about a bug in the past, but that was solved and the system is up to date, so there should no warnmail. Can anybody help?

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Lets Encrypt Automatic Update Killed All Sangoma VPN connected connections?

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@steve_pbuk wrote:

This morning when I logged on FreePBX I had the usual warning message regarding Lets Encrypt renewing certificate. This has worked flawlessly in the past without me having to do anything such as restarting services, etc.

This morning all of my Sangoma S500 connected via VPN had dropped and failed to reconnect. I check /var/log/messages and could see a stream of errors similar to the following:

OpenSSL: error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed

To fix this I tried the following:

  1. Rebooted the server - Didn’t fix the issue.
  2. Issued the “fwconsole -r” command - Didn’t fix the issue.
  3. Did some Google and came across the following page - https://forums.openvpn.net/viewtopic.php?t=23166 I changed “default_days” and “default_crl_days” but the command to regen the files didn’t work. - Didn’t fix the issue

The fix for me was:
Via the web frontend I went into the VPN Server under Admin and I disabled the VPN server. I then re-enabled the VPN server and almost instantly the phones started to connect back in via VPN.

Questions:

  1. Has anyone else experienced this?
  2. Am I correct in thinking this was due to the Lets Encrypt Auto renewal or was it just bad timing?
  3. Is it possible to stop and start the VPN server via CLI incase this problem happens again?

Thanks

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Ring Group Call Waiting

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@CPorter wrote:

Hello FreePBX Users!

I have a ring group set up to dial 2 extensions in an office. They are setup with “Skip Busy Agent” set to no and both have Call Waiting enabled on the extensions. If a call comes in and is picked up by Extension A and then another call comes in, Extension A does not have the ability to pickup the call even if the initial call is hung up while the second is still ringing. Am I missing a setting here?

I can provide more details if needed.

Thanks for the help!

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Error starting Mariadb (freepbx)

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@gettrashed wrote:

I have such problem when i rebooted my server
soo i get this error on GUI :

SQLSTATE[HY000] [2002] No such file or directory::SQLSTATE[HY000] [2002] No such file or directory

then i chechked if mariadb was running it wasn’t running when i try to start
i have this error :

Job for mariadb.service failed because the control process exited with error code. See “systemctl status mariadb.service” and “journalctl -xe” for details.

then i tried to remove ib_logfiles but still nothing
can anyone help mee ?
i have big info so i don’t want to reinstall mariadb but if i can backup database file (like i can see in vim /var/lib/mysql/asterisk/ some files ) if i can backup this and then take it in place then i can reinstall mariadb any sugestions? plz help

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FreePBX call URL API

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@julwo20 wrote:

Hi, Is there a way that I can program a BLF entry to dial and URL? ex. when pressing the second line key on my Digium d40 phoe go to http://192.168.14.21/state.xml?relay65State=2 this is basically telling my door system to release the magnet and open the door.

We have this option on our Digium SwitchVox and I’m trying to implement the same Feature on FreePBX.

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Help to access FreePBX from outside

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@Nucleus wrote:

Hello guy,

N00b here.

I’m using FreePBX 13.0.197

For several years we’ve been using FreePBX with only “internal” phones: all the phones where on the LAN and it worked like a charm.

We mainly use Digium phones (D40, D50 and D70).

Using the “Digium Phones” modules (DPMA), everything has been very easy: the phones see the FreePBX server and provision themselves.

Now for the first time, I’d like to have a few of those Digium phones on a remote location which could connect over the Internet.

Therefore, I’ve been looking for instructions how to do that but haven’t found what I was looking for…

Here are my questions:

  1. What ports do I need to forward to my FreePBX box?

  2. How can the Digium phones provision themselves but from the outside?

  3. How can I make sure the security is tight, if I open my server to the dangerous world wide web?

  4. Anything else I need to know?

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Status Screen not updating on its own?

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@fizgig10 wrote:

I like having the webpage open to show the graph of how many trunks are up, sip lines in use, etc… Only issue is that it doesn’t update on its own - seems like you need to hit the refresh circle-arrow icon to get it to do so.

Is there anyway to keep it updating by itself? Slightly off topic: If all I want to see is that graph part, can I somehow close the other modular windows?

Thanks!

(FreePBX 13.0.197)

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Feature Code --> Plays Recorded Page

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@mvogel4949 wrote:

Does anyone have insight as to how I could activate a recorded page announcement using a feature code? Say I dial *222 and it plays a lockdown message over the paging system or over the phones if paging is enabled on the phone. I looked through page pro and didn’t come across this as an option.

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High Memory Usage - V14

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@mvogel4949 wrote:

I’m running FreePBX V14 with Asterisk 13.22
Current System Version: 12.7.6-1904-1.sng7

I am running queues which may contribute but within a week my memory is pretty well pegged at max and I start to see degraded call quality. Below is my htop page after 6d uptime. Is there a way besides a weekly reboot(which I’ve been doing) to keep my memory under control? Thank you

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Callflow is gone

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@ELindemann wrote:

Hi,

it is very strange, i lost overnight on a FPBX-14.0.5.5(13.22.0) all my callflow (callFlowControl, Announcements/TimeGrp/TimeConditions/Queues etc. all.).
It worked fine, with all devices; it is/was right to say, works as designed.

But somehow(?) :frowning: today, i only can connect to extensions without any callflow, only (old/new configured) direct connects for DID to extension, without any interference of a callflow, like timegroup/timeconditions/queues or any higher logic level.

I also have a backup, two weeks old, the same. :frowning: Should not be.

How do can reconnect, reinit to old, fully functional callflow?
What happened?
BTW: The virt. machine was not crashed. Only shutting down, or reboots. No crashes.

Do freepbx has an internal time limitation, like one year or like that?

I could define a new extension (250), which i could reach/connect, ending on an announcement, like designed.

The defined clients/extensions are online:

  Contact:  211/sip:211@172.16.14.11:5060              xxxxxxxxxx Avail         3.796
  Contact:  212/sip:212@172.16.14.12:5060              xxxxxxxxxx Avail         5.509
  Contact:  213/sip:213@172.16.14.5:5060               xxxxxxxxxx Avail         5.841
  Contact:  214/sip:214@172.16.14.5:5060               xxxxxxxxxx Avail         4.995
  Contact:  215/sip:215@172.16.14.5:5060               xxxxxxxxxx Avail         5.623
  Contact:  216/sip:216@172.16.14.5:5060               xxxxxxxxxx Avail         5.812
  Contact:  217/sip:217@172.16.14.17:5060              xxxxxxxxxx Avail         3.688
  Contact:  218/sip:218@172.16.14.17:5060              xxxxxxxxxx Avail         3.758
  Contact:  219/sip:219@172.16.14.51:5060;uniq=123456 xxxxxxxxxx Avail        16.458
  Contact:  220/sip:220@172.16.14.6:5060               xxxxxxxxxx Avail         9.391
  Contact:  221/sip:221@172.16.14.6:5062               xxxxxxxxxx Avail         8.125
  Contact:  99/sip:99@192.168.114.113:5060             xxxxxxxxxx Avail         1.932
  Contact:  Provider_XYZ/sip:004955123456@sip.provider xxxxxxxxxx Avail        14.969

but they will not ring, because the callflow is not working anymore.
I only hear, that this extension ist not online. :frowning:

If freepbx is down, i ca hear the provider message, no client on the other end. That means, the call from pstn reaches the machine, but it goes on a black hole, not to the defined callflow. Why?

What and how can i report more details, why my callflow does not work anymore?

Kindly asking for help, any help appreciated. Pls.
Thanks.

Greetings
ELindemann

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IVR direct dial extensions on other PBX (through trunk)

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@GeekBoy wrote:

Following this thread, it does not make a lot of sense. I am in the same boat now, but with FreePBX 14 (if that makes any difference)

@lgaetz is recommending to use a custom directory.

The confusion part is that is really not directly dialing an extension, but only you have to enter the IVR code, then it takes you to the directory, then you have to take extra time to dial at least three letters of a name.

So anyway, I just tried one extension to test, and it functions, but that seems like a lot of extra dialing of digits. Also, what if they name is forgotten, but on ly extension is recalled?

Surely, there is a better way to do this?

Update:

Thinking of something else, I was able to come up with using a custom extension and the Extension Routing module.

Using a dial string, and Extension Routing, the custom extensions are only allowed to use the outbound route for that trunk connecting the two PBXs together.

Seems to be working out

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How do I setup keymapping for buttons such as hold, forward, redial, etc?

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@ATHiker wrote:

I have some Mitel 5300 series IP phones setup to use with FreePBX. Any idea how I can go about keymapping all of the feature keys such as hold, forward, and redial and all of the line appearance keys? Thank you!

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Override call record location to SMB share with user and password?

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@shivansps wrote:

Hi,
my Frepbx runs on a VM in Windows Server so i would like to record calls directly to a SAMBA share.
For what ive been searching the forums i found that is enoght to just enter the address and share on “override call recording location”. example: “\192.168.80.2\sharename” BUT i need to auth with a specific user and password.

Any ideas on how to do this?

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PBX takes 2 minutes to shut down on reboot

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@wpns wrote:

When I ssh in and tell the system to reboot, or I use Admin->System Admin->Power Options->Reboot, the system takes just over 120 seconds to stop responding to ping and shut down before restarting.

It’s probably some kind of housekeeping timeout, but is there a way to shorten the time?

Brand new install and fully upgraded to 12.7.6-1904-1.sng7 if that helps.

Thanks!

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How to export extensions, routes, etc from CLI

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@bksales wrote:

I realize that this might be more of a MySQL question but I’ve got a couple servers that are having major issues with the GUI crashing and the backups aren’t any good. Asterisk continues to run but there’s a lot of things I cant do without GUI access. Still not sure what is causing this but I want to rebuild them on new virtual machine ASAP and will have to do it manually.

How can I run the bulk handler from the CLI and get it to spit out a csv?

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Paging Pro Scheduler Delayed Pages?

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@vbman213 wrote:

I’m working on setting up our school bells to broadcast via Mutlicast to overhead speakers.

I’ve set up the scheduler with times, but the bells are triggering on a delay…sometimes up to 30 seconds it seems…with only 3 minutes in between classes, this poses a significant challenge.

Is there any way to increase processing resolution?

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Override call record location to SMB share with user and password?

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@shivansps wrote:

Hi,
my Frepbx runs on a VM in Windows Server so i would like to record calls directly to a SAMBA share.
For what ive been searching the forums i found that is enoght to just enter the address and share on “override call recording location”. example: “\192.168.80.2\sharename” BUT i need to auth with a specific user and password.

Any ideas on how to do this?

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There are 52 System updates - your system is up to date

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@cdsJerryw wrote:

When I log into my FreePBX 14.0.13.4 system it says “There are 52 system updates available. Run yum-updates to update them. Your PBX is up to date.” If I try to yum update it says there are no packages for update.

How can there be 52 updates, but I’m up to date and no updates? Doesn’t one of those things need to be wrong?

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