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Locked Out of FreePBX 15

September 24, 2019, 10:28 am
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@side1track1 wrote:

I’m currently locked out of FreePBX 15 via SSH and the GUI - I’ve tried multiple computers on the network accessing the IP using a browser and SSH via Putty (connection timed out) and I believe it is because fail2ban and the firewall were both enabled. Phones are still currently registered and incoming/outgoing calls work; however, I’m at a loss for how to now access. Does the fail2ban delete after a certain amount of time? I can still ping the IP via a command prompt. I’m going to try accessing via a VM; however, does anyone else have any ideas on how to access? Thank you!

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Intermittent Static, 1-way audio - Any Diagnostics?

September 25, 2019, 12:33 pm
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@RealRuler2112 wrote:

I have 2 PBXact servers running in HA mode interfaced to the outside world via a SIP trunk. Endpoints are a mix of Sangoma phones (305, 400, 505, 705) and Cisco 7960 phones flashed to SIP. We have 100 meg synchronous fiber and 100x10 cable internet tied through a Meraki MX64 edge device for service.

Recently, I’ve been getting some reports of static in calls and spontaneous 1-way audio in the middle of a call. I know for a fact that part of it is the little crackle of static whenever the animation moves on the Sangoma phones with a display - waiting on new firmware to become available in order to remedy this. I was able to find a recording of one such call and could hear both ends of the conversation on the recording no problem. It’s very intermittent though, happening only a handful of times in a given day. The SIP provider said to check with the ISP, whom I am currently waiting to hear back from.

In the meantime, I was wondering if there are any diagnostics, network statistics, etc available in PBXact that might possibly shed any light on what’s going on? I’ve poked around and have found the CDRs and call recordings, but that’s about all - nothing that would show if network congestion were detected at the time, etc.

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Dead air for only specific dial patterns on outbound calls

September 25, 2019, 7:02 pm
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@tlarrea wrote:

Had our FreePBX server running for a couple of years now with no real issues. For some reason in the last few days, when we dial out to numbers with 1300 XXX XXX, or even just 13 XX XX, we get dead air. The call appears to answer, but I can’t hear anything.

I get the following in the asterisk logs.

[2019-09-26 11:56:43] NOTICE[2098] chan_sip.c: Disconnecting call ‘SIP/0282432310-00000260’ for lack of RTP activity in 31 seconds

The strange thing is calls to any other number format work just fine. Not sure what to look at. I’ve tried deleted and recreating the outbound route for the problem numbers but that didn’t help. Also checked the firewall to confirm that RTP 10000-20000 is allowed.

Any thoughts on where I should look next?

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FreePBX doesn enter GUI without internet

September 26, 2019, 2:14 am
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@bendingbender1 wrote:

My FrePBX server working in closed LAN. I activated them and install system admin module. All working except Web GUI, its stuck after entering login/password in any browser. Chrome shows that config.php wont loading, in error.log i got message

client denied by server configuration: /var/www/html/admin/index.html

I figured out that after i enter command systemctl restart network in console, web page is loaded, but then it stuck again and i cannot access to other menu.

If server have access to Internet GUI is unstuck automaticaly. So i have a question: what hostname’s should FreePBX have access to throug firewall, is there a list? Or is exis another solution to work without internet?

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Outbound calls dropping while leaving VM

September 26, 2019, 6:20 am
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@mvogel4949 wrote:

I’m having a very odd situation where I’m getting a lot of dropped calls that all seem to drop while I’m leaving a VM for the person I’m calling. Any ideas why this might be happening?

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Dutch Voicemail problems - Solved!

September 11, 2019, 3:43 am
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@berjanschroer wrote:

Hi Everyone,

Currently i’m using FreePBX 14. when we have 1 voicemail in our inbox we are finding some troubles.
when we call *98XXX (or *97) we hear the greeting: You have 1 { call hangs up}.

In my research on the internet i found the following issue: https://issues.freepbx.org/browse/FREEPBX-17153
This guy has the same problem that we have, when you read the article the problem has been found!

my million dollar question is, how can i solve this problem? where can i copy the sound file and renaming it to fit on the missing one?

Thank in advance,

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Conference for normal calls

September 27, 2019, 1:02 am
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@NorColorNorName wrote:

Hello,

I have a perfectly working webRTC with asterisk 13.17 and FPX 13.

My task is to convert the classic call way into a conference.

My task : when a local extension call someone, they have to inquire the number of the called AND a conference number.

So, the classic way to process a call is the context “from-internal” but i need to insert the application “Confbrigde [conferencenumber],[bridge],[user],[menu]”
somewhere in the dialplan.

Do i have to rewrite an entire dialplan ? Or there is a maccro i can use to insert my custom application into “from-internal”

Thanks !
Have a nice day !

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Kari's Law - how are we going to comply? It becomes effective 2/20/2020

September 27, 2019, 4:40 am
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@sippy wrote:

When a 911 call is placed from a multi-line telephone system, a notification must be sent to on-site personnel, alerting them to the emergency. The notifications to an appropriate contact can take the form of phone calls, visual alerts on a monitor, audible alarms, text messages, and/or emails.

I understand that the PagePro commercial module handles this as far as notifying other extensions via multicast page, or by calling a group of extensions and joining them to the call, initially muted. If they want to speak, they press ‘1’ to open their microphone so they can talk.

What if the site wants to notify the security desk with silent notification, such as a screen pop, or test message?

What if this is a school and they want the local jurisdiction to be notified?

And now that Ray Baum’s Act section 506 had been adopted in a ruling by the F.C.C., what is the best way to implement the requirement that “the street address of the calling party, and additional information such as room number, or similar information necessary to adequately identify the location of the calling party”?

What are the minimum legal requirements for existing customers, and for new installations?

What are the potential liabilities for organizations who do not address these new requirements?

What about remote locations registered to the PBX?

Comments?

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NOTICE[15420]: manager.c:3504 authenticate: 127.0.0.1 failed to authenticate as 'admin'

September 27, 2019, 9:27 am
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@miken1343 wrote:

I’m seeing this message every time I login to Freepbx.

Asterisk Manager Password

Default Value: amp111

AMPMGRPASS=**********

Asterisk Manager Port

Default Value: 5038

ASTMANAGERPORT=5038

Asterisk Manager Proxy Port

Default Value:

ASTMANAGERPROXYPORT=

Asterisk Manager User

Default Value: admin

AMPMGRUSER=admin

[general]
webenabled = yes
enabled = yes
port = 5038
bindaddr = 127.0.0.1
displayconnects=no ;only effects 1.6+

[admin]
secret = **********
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=***.***.***.***/***.***.***.***
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000

#include manager_additional.conf
#include manager_custom.conf

This is some of my config. I did a /etc/init.d/asterisk reload and a fwconsole after i edit the files but im still get a “can not connect to asterisk” at the top right of freebpx.

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Installing chan_sccp manager

September 27, 2019, 8:53 pm
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@benjamyn_lynch wrote:

Hi all,

I’m completely new to linux and have installed the FreePBX Distro

I am trying to install chan_sccp manager however the doc says to run
mysql -u root asterisk < mysql-v5_enum.sql

This command does nothing. Can someone point me somewhere as the documentation isn’t great.

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Extensions generation

September 12, 2019, 6:04 am
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@NorColorNorName wrote:

Hello !

I’m using Asterisk 13.17.0 and FBX 13.0.

I want to configure my FBX to receive php request to create extensions for new user of sipML5.
I said php but i’m open for any other solutions.

My task is to create extensions for new users without the need to reload asterisk.

I heard about virtual extensions, is that a good way to start?

I’m a bit noob with FBX and asterisk but any advices will help,
Thanks by advance,
NCNN.

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Sound Files

September 29, 2019, 10:46 am
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@nortelvoip wrote:

Hello,

Does anyone know if Joan Kenley has a sound file package for Asterisk/FreePBX? She was the voice of all the Nortel voicemails like Meridian Mail.

Thanks!

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Cepstral app_swift Can't find asterisk include files

September 29, 2019, 6:50 pm
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@iracomm wrote:

Greetings programmers,
I’m trying to install the module, app_swift, in order to use the Cepstral Text-2-Speech engine.
I have the voices and installed.
When installing the app_swift module, one first runs “./configure”. If successful, it instructs the user to run “make”. If that is successful, “make install” is ran. And than “make reload”.

I get as far as “make”. Which gives the error: “include Asterisk files not found”.

I have searched the drive for any …/include/asterisk/ directories and found none.

If anyone has installed the app_swift module on any Asterisk box, please let me know how you did it.
Or at least, where or how to get the include files for asterisk.

Our asterisk version is 13.

TIA, Ira

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How to tell what system recordings are in use

September 12, 2019, 10:43 am
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@travisdietrich wrote:

Is there a way to tell what system recordings are in use and where?

I am trying to clean up files from our IVR and I find that if a system recording id in use and deleted, the place in use just reverts to “None” with no warning.

Want to delete some recordings, but don’t want to inadvertently break an announcement somewhere.

(Even a config file I can look thru would be great)

Thanks,

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Zulu Android Mobile App certificate warning

September 25, 2019, 7:57 am
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@PeterJ wrote:

Hi,

I’m just setting up a new install for a client. It’s possible that in the future they may want to use Zulu so I
have installed the 2 user free license and got Zulu working for the PC desktop app. When I try to login to the Zulu Android app on my phone I get the error:

"Invalid Server Certificate"
"Zulu Mobile requires a trusted SSL Certificate to maintain a secure Connection. Your server’s certificate is invalid. Please contact your administrator and ask them to correct this issue"

For various reasons we use Sectigo (formerly Comodo) certificates and the installed certificate on this server is trusted by Chrome, IE, etc. On a previous install where we used Sangoma phones I seem to remember that they didn’t like Comodo certificates - could this be the same issue with Zulu Mobile ?

Thanks for reading,
Regards,
Peter.

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My IP Address Is Blocked

September 30, 2019, 7:12 am
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@dagius wrote:

My IP address keeps getting blocked and I cannot access the admin page. Any idea how to correct this?

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All circuits are busy - unsure of how to diagnose cause

September 30, 2019, 9:14 am
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@AdamAtAvalon wrote:

Hi all,

I’m very new to FreePBX (only deployed one instance into production), and it seems to be working fine most of the time, except that it has started to occasionally give the message “all circuits are busy”. I have tried googling around to see if I can find the issue, but it would appear that this is an issue with multiple causes, and I am unsure of how to go about diagnosing my specific issue. Thus, I have turned to the community for help. I would like to post my log files, but attempting to do so gives me the message that “new users can’t post links”. If someone lets me know how to circumvent this, I will be happy to do so. Thanks in advance!

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SQLSTATE 42S22: column not found : 1054 Unknown Column 'fcode_lang'

September 30, 2019, 11:12 am
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@sentinelace wrote:

I restored from backup onto a new fresh PBX. Everything works fine except when I try to upload a recording, I get the following error:

capture

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How to configure instant messaging in freepbx 14 & Asterisk 13

September 30, 2019, 12:26 pm
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@ranahashem wrote:

:disappointed_relieved::disappointed_relieved:
Hi everyone, i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can anyone help me !

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Httpd Service keeps stopping in SNG7 (Freepbx 14 Distro)

September 13, 2019, 11:27 am
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@fdoula01 wrote:

Hi everyone!

I was running my Freepbx 13 on a Dell PowerEdge R610 hardware with no problems. I tried to upgrade from 13 to 14 and ran into too many problems (mainly NIC card issues with new naming as they said it will likely to happen) that i decided to reformat and install the fresh copy of the Freepbx 14 Distro. After installations and all the configurations and everything done, system was running fine. Till recently, when I could not access the GUI, even though everything is working. and i have to go and do “restart httpd service” command every time i need to access the GUI now. Doing that it will fix the issue but puttying in and doing that every time is a bit annoying. I wanna save myself the hassle. I tried to look up for a solution, but most were complaining about the Apache server not starting or cannot access because of the ban. Mine is just the httpd service keeps going offline and needs a kick in a butt!

Any help or advise is greatly appreciated. I am a novice in Linux but i can find my way around the problem if i am pointed to the right direction.

Any ideas? what am I missing here? i have not done any changes but to occasionally install the updates on the modules that i get warned about.

Thank you so much in advance
Sean

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