@mleger80 wrote:

Morning

I think i’m running into some permissions issues, but i’m not sure what i need to change or if i need to setup a new user.

This is a new install, services fully restarted.

Web Page returns - ‘fwconsole reload’ failed, config not applied - error on the admin dashboard

When i try and run it manually on console, i get the following.

$ fwconsole reload
PHP Warning:  include_once(/etc/freepbx.conf): failed to open stream: Permission denied in /var/lib/asterisk/bin/fwconsole on line 12
PHP Warning:  include_once(): Failed opening '/etc/freepbx.conf' for inclusion (include_path='.:/usr/share/php') in /var/lib/asterisk/bin/fwconsole on line 12
PHP Fatal error:  Uncaught Error: Class 'Symfony\Component\Console\Application' not found in /var/www/html/admin/libraries/FWApplication.class.php:11
Stack trace:
#0 /var/lib/asterisk/bin/fwconsole(66): include()
#1 {main}
  thrown in /var/www/html/admin/libraries/FWApplication.class.php on line 11

So i figured maybe i just needed to run it as root, so sudo fwconsole reload gets me the following

Reload Started

In digium_phones.php line 31:

Methods with the same name as their class will not be constructors in a future version of PHP; digium_phones has a deprecated constructor

reload [--json] [--dry-run] [--skip-registry-checks] [--dont-reload-asterisk]

I found a few things online but all it seems to be a little different than what i’m running into.

Any suggestions on where i can go next?

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@astrakid wrote:

hi,
we are using freepbx for over two years now, everything was fine. today, out of a sudden, CPU spiked top 100% and asterisk was not answering calls anymore. i rebooted the server, because i remembered we had that issue months ago and a server reboot solved the problem.
after the reboot the issue remains.
i can see that calls are incoming via sngrep. in asterisk i can only see that with the call the message
Setting global variable ‘SIPDOMAIN’ to ‘10.xy.ab.cd’
appears. In this moment the CPU goes up to 100% and more. After 2 minutes (!!!) I see normal messaging within asterisk.

There was no change in freepbx. We are connected to “Deutsche Telekom” with pjsip, which worked fine before and still works for my provate freepbx installation.

can anyone help?

edit: because nothing works, i upgraded everything to the latest stable version (last update was done 6 weeks before, don’t remember the version). now running:
asterisk 16.11.1
freepbx 14.0.13.34

here the sip messaging. have a look at the timestamps…:

kind regards,
andre

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@ghurty wrote:

I have two paging systems. One I need to dial 501, then #15
The other one 500 wait 2 seconds, 00 and then 0.

Is there a way to write a custom dial plan that will let me dial one number to make the announcement over both? I know how to send a DTMF tone in the dial plan, but I would need it to call both sets of extensions, send different DTMF tones to each and then conference them all in as one.

Thank you

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@Xnegidx wrote:

Long time lurker and learner from this site, but first time posting.

I’m running into a problem with voicemail greetings from only 1 extension. It is saying the file is not in the correct format, but when I run ‘file busy.way’ all the info looks correct. I’ve rerun fconsole chown to make sure permissions are set correctly, but it still says invalid file. I’ve attached a scrubbed log file of the incoming call. Any help is greatly appreciated.

[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [vmx@macro-vm:10] GotoIf(“SIP/fpbx-1-b6ddd880-0000000f”, “0?s-NOANSWER,1”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [vmx@macro-vm:11] GotoIf(“SIP/fpbx-1-b6ddd880-0000000f”, “1?notdirect”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx_builtins.c: Goto (macro-vm,vmx,13)
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [vmx@macro-vm:13] NoOp(“SIP/fpbx-1-b6ddd880-0000000f”, "Checking if ext 7000 is enabled: ") in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [vmx@macro-vm:14] GotoIf(“SIP/fpbx-1-b6ddd880-0000000f”, “1?s-NOANSWER,1”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx_builtins.c: Goto (macro-vm,s-NOANSWER,1)
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [s-NOANSWER@macro-vm:1] Macro(“SIP/fpbx-1-b6ddd880-0000000f”, “get-vmcontext,7000”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:1] Set(“SIP/fpbx-1-b6ddd880-0000000f”, “VMCONTEXT=default”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/fpbx-1-b6ddd880-0000000f”, “0?200:300”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [s@macro-get-vmcontext:300] NoOp(“SIP/fpbx-1-b6ddd880-0000000f”, “”) in new stack
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] pbx.c: Executing [s-NOANSWER@macro-vm:2] VoiceMail(“SIP/fpbx-1-b6ddd880-0000000f”, “7000@default,u”) in new stack
[2020-09-11 08:05:56] WARNING[30172][C-0000000f] format_wav.c: Not a supported wav file format (49). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are supported with a lowercase ‘.wav’ extension.
[2020-09-11 08:05:56] WARNING[30172][C-0000000f] file.c: Unable to open format wav
[2020-09-11 08:05:56] WARNING[30172][C-0000000f] file.c: Unable to open /var/spool/asterisk/voicemail/default/7000/temp (format (ulaw)): No such file or directory
[2020-09-11 08:05:56] VERBOSE[30172][C-0000000f] file.c: <SIP/fpbx-1-b6ddd880-0000000f> Playing ‘vm-intro.ulaw’ (language ‘en’)
[2020-09-11 08:06:00] VERBOSE[30172][C-0000000f] app_macro.c: Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘SIP/fpbx-1-b6ddd880-0000000f’ in macro ‘vm’
[2020-09-11 08:06:00] VERBOSE[30172][C-0000000f] pbx.c: Spawn extension (ext-local, vmu7000, 1) exited non-zero on ‘SIP/fpbx-1-b6ddd880-0000000f’
[2020-09-11 08:06:00] VERBOSE[30172][C-0000000f] pbx.c: Executing [h@ext-local:1] Macro(“SIP/fpbx-1-b6ddd880-0000000f”, “hangupcall,”) in new stack

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@slip_cougan wrote:

I have an issue where it appears Asterisk is trying to dump a load of stuff to disk causing the host PC to crash.

This seems to stem from one extension calling into a ring group:
The extension in question is a Grandstream GXV-3500 which is configured as a door intercom providing audio and video.

The ring group consists of 7 Crestron TSW-1060 touchpanels and a Polycom VVX-600.

Sometimes this will work fine for a day or so, sometimes a few minutes after a call has been made.
The symptoms can be any of the following:

• The call doesn’t always hang up
• The door release code ‘0#’ fails to work
• Call goes dead

When the above happens there is about a minute or two delay then the PC will reboot.
The HDD LED will come on and glow red as if it’s trying to write a ton of stuff to disk.

I’ve tried directing the Grandstream simply to the Polycom and it ‘seems’ to work fine. I say seems as it did crash once.

I’d like to find out what is happening before the reboot. The logs after a restart don’t contain anything of note that I have been able to see using the FreePBX Asterisk Logfiles.

Is there any way to create a persistent log? Or if there is one already created, where do I find it?

Current Asterisk Version: 16.9.0
FreePBX 14.0.13.28

Thanks
-s

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@f1systems wrote:

I have call ids that I cannot grep from full* logs. There are numerous of these calls and none of the call IDs come up when grepping them from full*

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@hariskar wrote:

In Dashboard there is a notification 32 messages are qeued on ths machine and have not been delivered. I would like to empty Mail qeue, how could I do this?
Than you!

PBX Version:
15.0.16.73
PBX Distro:
12.7.6-1904-1.sng7
Asterisk Version:
13.22.0

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@mvogel4949 wrote:

I have a system that keeps bogging down and shutting down http in the process. When I check htop I find multiple of the maria db pid messages as seen below. Any ideas how I can fix this?

image.png1774×83 4.55 KB

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@ruijorgesilva1 wrote:

I’ can’t load Chan_Mobile.so on asterisk! I’ve created even an chan_mobile.conf file in /etc/asterisk

when I do:
asterisk -rx “module show” | grep chan_mobile
chan_mobile.so Bluetooth Mobile Device Channel Driver 0 Not Running extended

Bluetooth does work and find devices without a problem!
hcitool scan
Scanning …
48:50:XX:XX:XX:XX Windows phone
70:BB:XX:XX:XX:xx XiCS

what could be wrong?

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@Vorms wrote:

Hello,
I have to change the host for the trunk.
I change it on the Connectivity / trunks / sip settings and write the new host (sip provider)
I save it, submit, reload, re start the virtual machine, but the old host is used.
How can I modifiy that ?

Best regards
Thierry

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@tcmtelecom_stu wrote:

Hi
My instance has been running low on diskspace for a few weeks now and having frightened myself googling how to increase it i have been using filezilla to go into /var/log/asterisk and deleting the older logs. This usually got me down to just below 70% and it might be a week before i got the warning emails at 75% - then filezilla again… However, today, Sunday, i took the bull by the horns and using a variety of sources but mainly using " How to extend a partition with unallocated space CentOS 7" on Webcore cloud, i managed it without even shutting down the server - so i thought i would share this with you, most is copied and pasted but some of the partitions are slightly different (if you know what you are doing, unlike me, then you maybe don’t need my full step by step by step) Here goes:

1. Take an image or a backup - you have been warned - if this fails you can go back.
2. Upgrade vultr instance to a bigger disk using their control panel.
3. Extend partition

Copy

fdisk /dev/vda

Enter p to print your initial partition table.

Enter d (delete) followed by 2 to delete the existing partition definition (partition 1 is usually /boot and partition 2 is usually the root partition).

Enter n (new) followed by p (primary) followed by 2 to re-create partition number 2 and enter to accept the start block and enter again to accept the end block which is defaulted to the end of the disk.

Enter t (type) then 2 then 8e to change the new partition type to " Linux LVM ".

Enter p to print your new partition table and make sure the start block matches what was in the initial partition table printed above.

Enter w to write the partition table to disk. You will see an error about device or resource busy which you can ignore.

Update kernel in-memory partition table

After changing your partition table, run the following command to update the kernel in-memory partition table:

Copy

partx -u /dev/vda

Resize physical volume

Use this command to resize the PV to recognize the extra space

Copy

pvresize /dev/vda2

Resize LV and filesystem
lvextend -l +100%FREE /dev/mapper/SangomaVG-root

Then you need to use xfs_growfs command to to resize your partition, so use:
xfs_growfs /dev/mapper/SangomaVG-root

That should be it - hope i helped somebody.

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@brian4 wrote:

I am sure this has been asked many times before however I cant find an answer. While watching activity in the Root directory I notice this.
(10 headers 0 lines) —

Really destroying SIP dialog ‘610d374f46860c6973327f2f6d77c550@[::1]’ Method: REGISTER
Really destroying SIP dialog ‘0_3086077623@10.1.1.241’ Method: REGISTER
Really destroying SIP dialog ‘0_2027421819@10.1.1.58’ Method: REGISTER

Is it something I need to worry about? - Is it something I need to fix? -Is it something I can fix?

Any help would be appreciate as I dont really want to destroy anything

Cheer

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@VoIPTek wrote:

I wanted to confirm if we agree this is a bug before adding it to bugs.
Upon updating an extension I get the following error, the information does get saved, but of course we have the error.

Nothing has changed in some time, with exception of module updates.

image.png1710×914 70 KB

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@staticsyphon wrote:

Hello,

We have a new customer that operates a school. From my research so far, it sounds like we need to have emergency response locations at the Room level.

Is there any way to forward/specify a room number given a specific extension?
E.g. ext. 1201 calls 911 (with same DID as other extensions), it shows room number B201.
ext. 2302 calls 911 (same DID as above), it shows room C302.

Some of what I’ve found online implies 1 (more more??) DIDs are needed per room phone which would normally just be an extension. Can anyone confirm this or if it’s at all possible to dynamically set the room number for E911 when the call goes out to 911?

Thanks in advance for any thoughts/comments on this, really hoping to avoid purchasing 450 DIDs…

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@wassy83 wrote:

Hi to all,
I have this annoying issue coming after 4 years of no problems and without changes to my environment. We hear on incoming external calls something like a CLOCK randomly (so not everytime) and periodically(so when it happens we hear the CLOCK every some seconds ). The issue is only on our side, so on the side of the caller the sound is perfect. After investigeting that there are no issue on the local phones sides, I made two tcp dumps, the first one on the freepbx console, and you can clearly view the interfacence in the picture below that is coming from the italian provider EHIWEB to the local freepbx

screen_rtp.JPG1241×650 98.1 KB

and I made another dump in the router console too(this is a pfsense box with a lot of resources) and I can see again the noise coming directly from their IP to our like this

noise.JPG1616×834 137 KB

at this point I was sure that the issue is coming from the providers side, and after sending them everything I have they told me that a second level IT replied in this way:

"From the analysis of the audio streams it is clear that it is a broacast msg on the customer LAN network, since it is present even before the conversation is established, on the carrier side this anomaly is not present in any other customer "

so I checked again and again the cap that I’m attaching( the router’ s side one router_cap.tgz (1,7 MB)
) but I cannot find any broadcast message (I searched on the wireshark wiki for the appropriate filter eth.dst == FF:FF:FF:FF:FF and other tools) without results, so I cannot understand where they found this message if we are looking at the same cap.
Any help or starting point will be very usefull for me… many thanks

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@dmchatten wrote:

I got an error the Asterisk cannot connect when I was in the web GUI. I’ve tried fwconsole restart, fwconsole stop then fwconsole start and those don’t seem to work.
What else can I do? Was working last week. This is an old install that’s been working fin until today.
I have a VULTR backup from 9/1, should I just restore that?

EDIT: It seems to be working now

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@hariskar wrote:

I activated call recording for all calls an have some quesions:
As I read the wav files are stored internally, this means that in some time I will run out of space. How can I deal with that? I did not find an option in freePNX menu to delete all call recordings older than eg 20 days.
Can I just go from time to time in folder /var/spool/asterisk/monitor and delete the older files ?

Thank you!

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@mcisar wrote:

Have an older FreePBX system at a site that I “inherited” the management of… we have issues intermittently at that location where the trunks simply get stuck in the registration process. The guys there usually use their cels to make outbound calls, so unless they actually happen to try to make a call from a desk phone and get the “all circuits are busy now” recording it could be a couple of days before anybody realizes that the trunks are down (unless somebody tries the VOIP line and then gets ahold of them on their cel to let them know). The problem is usually caused by the internet connection used by the trunks being down (I think).

Host dnsmgr Username Refresh State Reg.Time
provider.one:5060 N xxxx_xxx 120 Request Sent
provider.two:5060 N xxxx_xxx 120 Request Sent

How can I generate a notification when the trunks are stuck in this state? I have another way that I can get the notification out, if I can get FreePBX to just raise the SOS.

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@glsarto wrote:

Hello All,
my log files are flooded with “deprecated” or “depreciated” (sic!) warnings…
I wonder why, since it’s the latest release and is kept up to date automatically.
This makes really tiresome wading through the logs.
How can I fix this?
thanks for your advice,
-Gian

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@amardeeprana wrote:

HI

I am getting this message during active calls ( I have enabled SIP debug ). Please suggest.

##watch -n 1 ‘asterisk -rx “sip show channelstats”’
1626627705- (inv state: None) – -- No RTP active

Thanks

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