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Appliances (Phone System 40) - Upgrade to v14?

December 27, 2017, 11:26 am
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@nmarques wrote:

Hello,

I am setting up a new Sangoma Phone System 40. Before I even get started, should I just go ahead and upgrade the distro to FreePBX 14? It came with 13. I don’t see any problems doing so, but I can’t find much info if Sangoma does different branches for their own appliances. This isn’t the PBXact product.

I know how to install from USB on 3rd party hardware, but am not clear on what the Sangoma hardware entails. Do I just make a USB and boot to it? I assume the distro has everything the appliance needs?

EDIT:
Looks like I could just follow this??

Thanks.

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Troubleshooting SIP Registration (Alerts, Recovery, Etc)

December 27, 2017, 12:53 pm
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@nmarques wrote:

Hi all,

I am using FreePBX 14 Stable with Asterisk 13.17.2. We use VOIP.ms for DIDs.

We just realized that we were not receiving calls all morning. Outgoing still worked as it doesn’t require registration. I did not see any incoming calls on the FreePBX CDR, but did see them on the VOIP.ms CDR.

When I checked registration status, it says NOT REGISTERED.

I went to the trunk entry in FreePBX, and edited the entry, changed nothing, and clicked Submit / Apply Config, and it came back online.

How do I go through logs to find what failed and why? When I look at the Asterisk history graph on the dashboard, I see no time where there is no trunk online.

How often does FreePBX or Asterisk register with the trunk?

How do I setup an e-mail alert when the registration is not online? Does it automatically retry? I guess what I am wondering is why did “refreshing” the config bring it back up?

It’s previously been up for a couple months.

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Using "User name" and "Authentication name" for SIP REGISTRATION

December 27, 2017, 3:49 pm
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@jmorrison876 wrote:

When registering a sip enpoint to Freepbx, the extension # is used for both the username and authentication name in the device config. Being new to Freepbx, I must ask is their a way to configure the PBX to accept a separate entry for the authentication name? The need is simply to add added layer of security to the system and is a generic feature in many well know PBXs out there.

Thanks

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NAT Configuration

December 28, 2017, 5:48 am
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@casajaguar wrote:

Hi,
I had installed FreePBX and configured succesfully for LAN.
I’m using a DD-wrt router with dynamic ip and a VM with FreePBX installed behind.
I would like to access to PBX with SIP from the outside but using a non-standard UDP port to avoid attacks with a Port forwarding moving the outside to 5060 in the inside.
I used to have an asterisk that worked this way.

Can you help me with the PBX configuration?
Regards

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Need an advice about time conditioning

December 28, 2017, 5:58 am
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@akadimka wrote:

Hi to all!
Here’s the question.
Have several extension with overlapping working time. E.g.:
1st is working from 8 a.m till 6 p.m. -> inc. calls are going to 1st ext. till 9 a.m.
2nd - from 9 a.m. till 7 p.m. -> inc. calls are going to 1st and 2nd exts from 9 till 10 a.m.
3rd - from 10 a.m. till 7 p.m. -> inc. calls are going to 1st, 2nd and 3rd exts from 10 a.m. till 6 p.m. & to 2nd and 3rd exts from 6 to 7 p.m.
from 7 p.m. to 8 a.m. goes to another destination

Could anyone advice an elegant soution to manage this through Time Groups, Time Conditions etc.?
“Elegant” means less groups, conditions with understandable logic

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Fresh Install Problems

December 28, 2017, 8:19 am
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@Telleraine wrote:

Greetings Community,

Hope everyone had a great holiday season. I am reaching out today in the hopes someone here can do what Google couldn’t.

When I installed the latest x64 distro yesterday (SNG7-FPBX-64bit-1712-2) everything went smooth as butter. But I am now stuck with 2 issues.

  1. When I go to reboot the machine, it just sits there, with a blank screen, doing nothing. It has been like that since yesterday afternoon. I have to physically press the RESET button on the server.

  2. I get an error when trying to do any yum commands about “cannot find a valid baseurl for repo: sng-base/7/x86_64”. I tried moving the Snagoma-* repo files out of the repo.d folder, and the error went away, but then it cannot find ANYTHING I am trying to install (webmin and the necessary prerequisites).

  3. This may be more for support, but none of my paid commercial modules from the old deployment came over when I built this one and activated it.

Thanks alot for any feedback you can provide!

Brian

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Black and White hardware requirements

December 28, 2017, 9:48 am
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@WQGJ587 wrote:

I cant seem to find, a old fashion black and white hardware requirement list for the newest FREEPBX Distro.

I find articles, that to be honest, dont say a whole lot of anything except words,

Can someone point me?

I know that there is certain requirements to be meet because I am running a distro version 13 now. but can not update to 14. Has to be hardware issues. So, If I want to continue to stay current, I will have to step up in hardware.

Thanks
Tom

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Moving from plain asterisk to freepbx

December 28, 2017, 12:49 pm
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@haljbr wrote:

Hello Guys.

I am a little confused with all those sip_xxx.conf and extensions_xxx.conf that freepbx has.
On my plain asterisk I have this configuration below and I am not sure were to configure this on freepbx.
I tried on different screens under freepbx GUI interface but my trunk is not coming up.

This is on my plain asterisk:
sip.conf
[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes

[8558]
type=peer
mailbox=8558@test
username=8558
host=10.2.22.231
transport=tcp
vmexten=8558

[8559]
type=friend
mailbox=8559@test
username=8559
security=8559
host=dynamic
vmexten=8559

[8528]
type=peer
mailbox=8528@test
username=8528
host=10.2.22.231
transport=tcp
vmexten=8528

extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no

[public]
include => from-internal

[test]
type=peer
context=from-internal
host=10.2.22.231
port=5060
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
transport=tcp

[from-internal]
exten => 6060,1,NoOp(${CALLERID(num)})
exten => 6060,2,NoOp(${CALLERID(rdnis)})
some more stuffs in here

Any suggestion is appreciated.

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Failure of Backups

December 28, 2017, 2:22 pm
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How to test Asterisk new installation?

December 28, 2017, 7:41 pm
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@vlad1977 wrote:

Hi everyone,
I am a totally new user, looking for a guidance. Installed FreePBX 13 with the main goal to create telephony station for my small dental office.
I’m at the stage of testing the environment, able to login to the admin console, get through the menus, read warnings, update modules, unban the IP through the console (happened during testing the virtual extension).
I have to say it is not a user-friendly setup, or at least it does not seem to be one to me at the moment.
I have no idea what to do as the next step after the installation and all the help and tutorials got me nowhere.
so, if anyone fancy to become my mentor here, I’d really appreciate it.
So far I followed the official tutorial that I found by googling |FreePBX+Distro+First+Steps+After+Installation" (as a new forum use i’m not allowed to insert links) and stopped at “creating extensions”.
What I have done so far:

  1. Installed the FreePBX appliance, logged into GUI.

  2. Created a user named 1111 and linked the extension 1111 with a custom password.

  3. No ports opened yet, want to test virtual extensions inside the local network before complicating things more.

  4. Downloaded linphone and installed it.

  5. Set up the option “use SIP account” as follows:

  • “username” = “1111”

  • “SIP domain” = “192.168.XX.YY” (the ip of the FreePBX appliance)

  • “password” = XXXXXXXXX

  • “transport” = “UDP”

Getting the message “unable to authenticate” and eventually the IP of the desktop running linphone app gets banned, so I had to use the console to unban and subsequently whitelist the ip.
What do I do wrong?

I don’t mind if you guys point me to some meaningful rookie tutorial that covers this step.

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Inbound call is allowed to make transfer

December 29, 2017, 4:23 am
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@esarant wrote:

Hello,

I am really puzzled by a problem I face. I got a SIP trunk with a SIP provider. If I let the context to context=from-trunk then the caller can’t make a transfer (*2 or ##). If I change the contect to context=add-9 then the caller can make transfers. I am refering solely to inbound calls.

The context is really simple

[add-9]
exten => _X!,1,Set(CALLERID(num)=9${CALLERID(num)})
include => from-trunk

It basicaly adds a 9 at the incoming number so phones can call back with redial.

Why is this happening? Can anyone enlighten me?

Thank you,
esarant

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How to make Chan_SIP extension work with IP phone

December 29, 2017, 7:04 am
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@kashifiqb wrote:

I just added an Extension of Chan_SIP type to my Freepbx. But now when I try to connect to it using my IP Phone or Soft phone, the phones/soft phones are not getting connected. What could be the reason and how to solve this?

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Cosmetic issue after upgrade to 14 (and workaround)

December 30, 2017, 5:06 am
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@StevenA wrote:

I have just completed the upgrade to 14 from 13 using the upgrade module. The UCP module load failed because of missing modules pm2 and certman, but the system came back fine with just an error message about UCP.

However, in the module admin tab, I was no longer able to expand each module to see info and select options (upgrade/noupgrade)

I manually installed pm2 and certman, and then upgraded UCP, hoping that this would resolve the problem, however the problem still exists.

There is a workaround to select download all, and then work through the list.

I think that this is a JS issue related to bootstrap-select - the console showed that bootstrap-select.js.map was missing.

It is not a showstopper, but any suggestions on how to address this would be appreciated - it could be that there are other related issues still to be found!

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Master/Slave APC UPS for multiple Server

December 31, 2017, 5:00 pm
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@okynnor1 wrote:

Hi,

In FreeNAS it is possible to allow the setup of a master/slave config in order to allow multiple server to be shutdown. Is this feature available from the Web interface in FreePBX?

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Option 66 data altered on S500

January 1, 2018, 11:31 am
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@MacsOffice wrote:

I’m using option 66 from my switch and the data is altered.

Switch config:
ip dhcp excluded-address 192.168.200.1 192.168.200.200
ip dhcp excluded-address 192.168.200.250 192.168.200.254
ip dhcp pool Phones
network 192.168.200.0 255.255.255.0
default-router 192.168.200.254
option 66 ip 192.168.200.200

result is:
Option 66 screenshot

Can anyone tell me why this would be?

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Config not send Config to Sangoma S500 phones from FreePBX GUI buttons

January 1, 2018, 11:39 am
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@MacsOffice wrote:

After configuring the extensions and mapping them, FreePBX will not issue the configuration to the phones using the save, rebuild and send option or the button on the right column of endpoint manager

DHCP comes from a CIsco switch and that works (Except option 66 which is on a different thread) the phone all get IPs

I can ping the phones from the FreePBX box on the correct IPs, I can also reach the phones GUI from my workstation… so what can I have wrong where FreePBX wont send the config to the phone?

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Can't open UCP

January 2, 2018, 12:23 am
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@AB81 wrote:

Hello,

I get this error “array_merge(): Argument #2 is not an array” when I open UCP.

7.Whoops\Exception\ErrorException
/­var/­www/­html/­admin/­modules/­ucp/­htdocs/­includes/­less/­Less.php:454

I need help, please.

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Unable to use Asterisk General Call Pickup *8

January 2, 2018, 1:26 am
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@lblokland wrote:

Hi,

Best wishs to all!

I’m trying to setup call pickup (*8) but cannot get it up and running.

When I dial *8 on a phone it fails, the only two lines of output in the Asterisk log are:
== Setting global variable ‘SIPDOMAIN’ to 'ip-address-pbx’
and one second later:
WARNING[54402]: res_pjsip_pubsub.c:681 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event

Not sure if those are related…

I’ve tried the add the extensions to multiple group, with call pikcup enable and also added the groups to the extensions on the call pickup.
Groups or queues, it doesn’t work.

Please assist.
Cheers,
Leon

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FreePBX Installation

January 2, 2018, 1:42 am
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Strange emails

January 2, 2018, 3:18 am
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@jonnybobs wrote:

Hi All,

Keep getting an email from the system and can’t work out where from within or what its about.

I get emails from it in other areas with no issues…

hwew below is what we get.

The response was:
DNS Error: 40877664 DNS type ‘mx’ lookup of voip.freepbx.local responded with code NXDOMAIN Domain name not found: voip.freepbx.local

Final-Recipient: rfc822; asterisk@voip.freepbx.local
Action: failed
Status: 4.0.0
Diagnostic-Code: smtp; DNS Error: 40877664 DNS type ‘mx’ lookup of voip.freepbx.local responded with code NXDOMAIN
Domain name not found: voip.freepbx.local
Last-Attempt-Date: Mon, 01 Jan 2018 16:00:05 -0800 (PST)

---------- Forwarded message ----------
From: “(Cron Daemon)” j***********@gmail.com
To: asterisk@voip.freepbx.local
Cc:
Bcc:
Date: Tue, 2 Jan 2018 00:00:03 +0000 (GMT)
Subject: Cron asterisk@VOIP /usr/sbin/fwconsole pms mk_dirty
Make the rooms dirty.

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