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Static burst during call - anyone heard anything like this?

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@Richard_at_F1 wrote:

Heard this static burst on a customer’s system yesterday, they’ve been having a number of these plus some garbled speech and dropped calls, we haven’t been able to find any errors in the Asterisk logs/console related to this and we’re running tcpdumps hoping to capture one and get a closer look. Anyone have any idea what this might be? Not sure if these same bursts are happening on internal LAN-to-LAN calls or only on calls with external numbers.

http://s000.tinyupload.com/index.php?file_id=00871346650226676798

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Sip end user account as sip trunk

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@User_42 wrote:

Hello I have a SIP enduser account from my telco and I was wondering if I could use it as a sip trunk in freepbx?

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Https://mirror.freepbx.org is not working

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@Wasca wrote:

Hi Guys

I’m trying to run updates on my FreePBX install and I’m seeing the following error in the GUI

Warning: Error retrieving updates from online repository(s) (https://mirror.freepbx.org). Online modules are not available.

From the command line I’ve tried this.

fwconsole setting MODULE_REPO https://mirror.freepbx.org
Changing "MODULE_REPO" from [https://mirror.freepbx.org] to [https://mirror.freepbx.org]

Then tried updating the module

fwconsole ma -R https://mirror.freepbx.org upgrade framework
Getting Remote Repo list...Done
No such repo: [https://mirror.freepbx.org], skipping
No repos specified, using: [commercial,standard,extended] from last GUI settings

Starting framework download..
The following error(s) occured:
 - Retrieved Module XML Was Empty

Any other suggestions? I can’t seem to get the updates working.

BTW I can ping that domain.

ping mirror.freepbx.org
PING mirror.freepbx.org (199.102.239.8) 56(84) bytes of data.
64 bytes from 199.102.239.8: icmp_seq=1 ttl=55 time=27.2 ms
64 bytes from 199.102.239.8: icmp_seq=2 ttl=55 time=25.0 ms
64 bytes from 199.102.239.8: icmp_seq=3 ttl=55 time=25.1 ms

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Cannot Connect to Asterisk - Time Sensitive

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@comtech wrote:

Hello,

Any help here is critically appreciated. Came in this morning and our FreePBX could not connect to Asterisk.
FreePBX 14.0.13.17 (12.7.6-1910-1.sng7)
Asterisk 14.7.3

Seeing this error when running asterisk -cvvvvv:

FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x29b10a0 (0) at line 335 in update_file_format_info of media_index.c
[2019-12-20 08:10:36] ERROR[4878]: media_index.c:335 update_file_format_info: Excessive refcount 100000 reached on ao2 object 0x29b10a0
[2019-12-20 08:10:36] ERROR[4878]: media_index.c:335 update_file_format_info: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x29b10a0 (0)

[2019-12-20 08:19:12] NOTICE[6720]: acl.c:715 ast_apply_acl: Manager User ACL: Rejecting '127.0.0.1' due to a failure to pass ACL '(BASELINE)'
[2019-12-20 08:19:12] NOTICE[6720]: manager.c:3392 authenticate: 127.0.0.1 failed to pass IP ACL as 'admin'
[2019-12-20 08:19:12] NOTICE[6720]: manager.c:3426 authenticate: 127.0.0.1 failed to authenticate as 'admin'

Not sure what is causing this nothing has been touched, I believe. We need this server operation in 3 hours or we will miss reminding customers of critical appointments.

Tried so far:

- fwconsole ma update all

- yum update

- fwconsole reload

Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1

- fwconsole restart

Hangs at Starting Asterisk

- Checked manager.conf, which we never touch, it has entries for 127.0.0.1/255.255.255.0

    [admin]
    secret = REALLY GOOD ONE
    deny=0.0.0.0/0.0.0.0
    permit=127.0.0.1/255.255.255.0
    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

    [admin]
    secret = REALLY GOOD ONE
    deny=0.0.0.0/0.0.0.0
    permit=SERVER STATIC IP/255.255.255.0
    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

Any ideas on what is happening?

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Certificate named "SomeName" is going to expire in less than a month. Please update this certificate in Certificate Manager

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@Hawkeye wrote:

Not sure as to why it’s constantly displaying message about SSL Certificate expiring in < month when the certificate manger clearly displays 2021-01-21 (398 days). This happens on 13 and 14 FreePBX.

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Exceptionally long voice queue length queuing

Changing Freepbx GUI password from command Line

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@SiP1701 wrote:

I changed my admin password today and for some reason I am unabel to log back into my FreePBX.
I have access via CLI
There are so many articles saying different things as to how to change this password via CLI
Can anyone give me clear instructions on how to do this via CLI?

12.7.6-1904-1.sng7

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Fwconsole enable/disable all trunks at once

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@AndrewZ wrote:

Is there an option to enable/disable all the trunks with fwconsole at once?

Everything works fine with the individual trunks but I’m wondering what should be the bulk syntax for the command (if any).

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How to see that call is redirected call

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@Broomd wrote:

Hello everybody,

I´m quite new to the freepbx area. I´ve got a question:

At certain times, I want to redirect all calls coming in to my mobilphone ( so redirecting to my private number),BUT is there a way to see which calls are redirected?
Actually I see the number of the caller on my phone, but is there no way that I can see that this is a redirected call?

Thanks a lot :slight_smile:

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Can you allow barge in on a line without parking a call

Hook file '/var/spool/asterisk/incron/firewall.stopfirewall' was not picked up by Incron after 5 seconds. Is it not running?

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@joshzone90 wrote:

Getting this error when doing a restart. Hook file ‘/var/spool/asterisk/incron/firewall.stopfirewall’ was not picked
up by Incron after 5 seconds. Is it not running?

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Kari's Law / Ray Baum's Act

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@BlazeStudios wrote:

There have been quite a few topics that have trailed off into talk about Kari’s Law and Ray Baum’s Act so let’s do it here. It also seems that the focus is mainly on the notifications part, which while its grey area-ish, it is the one part that has the more leniency compared to the rest. So let’s cover some things.

Station = Endpoint/Phone/Device that dials.

  1. Anyone must be able to pick up a phone and dial 9-1-1 without any prefixes or special codes to initiate the call. This is supported in FreePBX.

What this means: The installer/maintainer must program the PBX to support this.

  1. 9-1-1 must be routed to a 24/7 staffed destination. This does not require the destination to be the PSAP.

What this means:If you have, let’s say, a Health & Safety station that is staffed 24/7 with first aid responders, 9-1-1 can be set to ring their station(s) as long as a live person answers the calls. It cannot go to voicemail or another alternate automated destination.

  1. Stations must present their detailed location information when 9-1-1 is dialed. This is regardless of an internal (as in item 2) or external (PSAP) destination. There are caveats to this such as Residential houses regardless of levels and small single level buildings can present a general location for 9-1-1.

What this means: As the installer/maintainer you must make sure that each extension in FreePBX has the Emergency CallerID field set to show the proper station details. This could just be the default station CallerID or a unique DID. FreePBX supports either.

  1. The presented CallerID must be a valid dialable number that can receive call backs.

What this means: When 9-1-1 is dialed and the CallerID is presented, internal or external, the destination answering must be able to return a call to the caller if they are disconnected for any reason. So if the call is going to a PSAP then the number presented must be routable back to the caller.

  1. Outbound/Internal only stations, i.e. they don’t receive inbound calls or have no ability to call the PSTN, must still support dialing 9-1-1, present the station CallerID location and be able to receive an inbound call for 9-1-1 purposes.

The above items cover what is known as the “Dispatchable Location” and new installs will require this after Feb 16th 2020. Most current PBXes support these abilities, including FreePBX, which is why these items are the onus of the installer/maintainer. So if for some reason you are installing a PBX system that can’t support this, you need to stop installing that PBX system.

Now in regards to the notifications, as the installer/maintainer there is zero expectation/requirement for you to modify the PBX hardware/software to support notifications. Kari’s Law puts the onus of that on the makers of the PBX that new systems they introduce to the market after Feb 16th 2020 must support notifications.

While all that is happening there is now Ray Baum’s Act that states existing systems must comply with Kari’s Law with 1-2 years of Kari’s Law being signed into law (so Dec 2019 which means at the latest Dec 2021). They figure that is enough time for businesses to come into compliance.

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Caller ID Manipulation

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@kendalldever15 wrote:

I have a private school that allows the parents to press 7 in the IVR which then calls a dedicated cell phone. Issue is the person with the cell phone ends up answered a bunch of telemarketing calls etc. not knowing if it is a parent.

I currently have 7 go to a ring group that dials the cell phone.

Is there a way to have them set at dummy number in their cell phone (say 999-888-7777) that says something like “Parent” so the person knows it is not a telemarketer and also pulse a fake number such as 999-888-7777 so the cell phone recognizes this number and displays the parent contact?

Or a better way?

Any help is greatly appreciated.

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Mitel crackling audio on system recordings

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@jcadman wrote:

I have FreePBX installed on a raspberry pi which I use for my home. I use 3 x Grandstream 2 Line 1 SIP phones as well as 2 x Mitel 5300 phones with SIP Version 06.03.03.06 which was implemented by TFTP. After setting up the Mitel 5330 phones I realised that there was an issue with the sound recordings on the Mitel 5330 where the sound recordings for voices distort/crackle. This is not the case on some of the sounds but majority of the sounds start to crackle. With my Grandstream phones, the sound recordings do not crackle. When calling in and out using the phones including extensions within the house, the audio is clear. The crackle only appears to be with the system recordings voice.

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System Recordings not working

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@jcadman wrote:

When on ssh I believe that I deleted the custom folder in /var/lib/asterisk/sounds. When I click the system recording module on the GUI, the browser loads the page until I get shown an error.

I have tried to access the ssh and adding the custom folder again however this does not seem to work.

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Letting go of the reins.. need help with my small office setup

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@bonjongp wrote:

Hi, learned a lot here. Was able to setup my own freepbx system for my small business. (startup solo practice doc)
Not sure if this is the right forum for this topic, I apologize if it is not. I’m looking for someone who can help me optimize my system. Need to focus my time on other things and let you, the experts optimize my phone system. And ofcourse, compensate for your time.

Biggest issue I have is I can’t figure out how to setup faxing though I’ve read that it can really be a pain in the rear to do it. Bought a Cisco ATA SPA 122 thinking even at least I can use my fax machine for occasional outgoing faxes (I have a cloud based charting system) where I have my incoming faxes routed to. I only fax out if I have hard copies that I do not need to scan in the system first. Rest of the other issues are small and just really consulting for optimization/review of features we may not be utilizing.

Thanks for your time.

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Tracking down nefarious international calls

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@ashcortech wrote:

Over the weekend I received an alert from my provider saying we’d hit the limit on daily international calling cost. I purposely set this very low so no major disaster.

I do need to track down how this happened though. The sip provider shows several calls over about an hour period. I can only really find one call in the FreePBX CDR’s though and expanding that call log reads like stereo instructions to me.

Can anyone offer some guidance on how to trace down from the FreePBX/Asterisk side who did what?

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Letting go of the reins.. need help with my small office setup

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@bonjongp wrote:

Hi, learned a lot here. Was able to setup my own freepbx system for my small business. (startup solo practice doc)
Not sure if this is the right forum for this topic, I apologize if it is not. I’m looking for someone who can help me optimize my system. Need to focus my time on other things and let you, the experts optimize my phone system. And ofcourse, compensate for your time.

Biggest issue I have is I can’t figure out how to setup faxing though I’ve read that it can really be a pain in the rear to do it. Bought a Cisco ATA SPA 122 thinking even at least I can use my fax machine for occasional outgoing faxes (I have a cloud based charting system) where I have my incoming faxes routed to. I only fax out if I have hard copies that I do not need to scan in the system first. Rest of the other issues are small and just really consulting for optimization/review of features we may not be utilizing.

Thanks for your time.

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Module Admin page not extending modules

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@cagriaksu wrote:

Hello everyone,

Yesterday, after updating our server from 13 to 14, things got messy. I’ve solved most of the problems but couldn’t figure out why I can’t click and extend the modules in the module admin page. I can check for updates, and apply them, but I can’t individually select the modules (they extend downwards when you click on them as you all know)

Does anybody know a reason and solution for this?

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FreePBX not getting IPV4 address

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@gileslacy wrote:

All of a sudden this morning my FreePBX install is not retrieving/setting an IPV4 address on the eth0 interface - only an IPV6 address.

To the best of my knowledge nothing has changed since yesterday when it worked perfectly.

My Internet router is acting as DHCP server and I have a reservation for the MAC address for the NIC on the FreePBX server.

Why would it suddenly choose only IPV6 and not stick with IPV4?

Help please,
John

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