Extensions steadily become unreachable after PBX runs for a few days

johnoneszeros · January 4, 2022, 1:24pm

I wonder if anyone can help me figure out what is going on.
I have a customer who is setup on a Vultr hosted freepbx 15 and has about 25 extensionsAfter a few days the odd extension will become unreachable for 60 seconds, then come back as reachable. As time goes by other extensions will start to do the same. If I leave it another day it will become more common. These phones are in five or six different locations and the fault can happen in any location to any extension. It’s almost as thought the PBX is running out of resources, yet CPU use is ticking over around 2% and i have given it 2GB ram and it shows .5GB spare.
HTOP shows no runaway processes.

Any Ideas?

PBX Version:
15.0.17.64
PBX Distro:
12.7.8-2107-3.sng7
Asterisk Version:
15.7.4

vespino · January 4, 2022, 2:37pm

I maintain a similar setup and I’m experiencing something similar. I have built a fail save that forwards calls to another location (one not connected to the PBX) in case all extensions get disconnected. I thought it could be an unstable internet connection, but seeing you have the same issue, it might be something else.

johnoneszeros · January 4, 2022, 3:23pm

Thanks for replying. I don’t think its internet connection related as I am seeing it from multiple locations and the main site has a 1000Mbs leased line. It looks more like it is related to a problem building over time on the server.

davidg · January 4, 2022, 4:17pm

What happens if you configure the phone to use a static SIP port?

I’m assuming SIP ALG isn’t in play?

vespino · January 4, 2022, 4:29pm

In my case I have set static SIP ports to each of the extensions because they were causing problems not registering behind the local firewall. This solved that problem. My client is behind a simple fiber connection and from what I know the router they have doesn’t have a SIP ALG setting.

johnoneszeros · January 4, 2022, 5:11pm

I am not sure how setting a phone to use a static port will help. The problem is not in evidence for 48 hours or more, then steadily builds. Rebooting the the PBX cures the problem for a further 48 hours, but the phone doesn’t know the PBX is rebooted. Can it be fixed through the PBX GUI or the endpoint manager? (the port)

yois · January 6, 2022, 1:54pm

Are your routers SonicWALL? This symptom is very typical for those devices.
If yes, switch to TCP for SIP transport, or adjust the UDP timeout on the router.
Or, even better, rip out the SonicWALL… They are horrible in general and especially with VoIP.

johnoneszeros · January 15, 2022, 1:50pm

Sorry to have been quiet on this. Shortly after my last response the info about the Phone apps hack reached me and I realised this machine was infected. After a lot of work the hack has been cleared up and the initial problem has gone.

johnoneszeros · January 24, 2022, 8:47am

Actually this does not appear to be over. Checking the system on Sunday night showed phones regularly dropping and coming back.
I didn’t reboot. I just made 1 minor config change and hit the apply config button. The phones stopped dropping!

vespino · January 24, 2022, 10:10am

What was your solution? Did you make a minor config change to the phone or to the PBX?

johnoneszeros · January 24, 2022, 10:36am

I wouldn’t say it is a solution. I made a config change to the PBX

vespino · January 24, 2022, 7:37pm

“A” or a specific?

ashcortech · January 24, 2022, 11:48pm

and that config change would be???

johnoneszeros · January 25, 2022, 8:10am

I added the server room phone to the pickup group

johnoneszeros · January 25, 2022, 9:23pm

Same thing happening this evening. An extension drops for 60 seconds, then comes back. Happens every 5 minutes or so.
I created a conference, hit apply, the dropping stops.

I checked all the stats before I did that, CPU ticking over, plenty of spare RAM, nothing much on the network. HTOP output looks normal.

bksales · January 25, 2022, 9:38pm

I see this happen occasionally. The PBX does not seem to be respecting the “qualify” timer in the extension settings. Usually changing it from the default 60 to something lower like 20 or 25 fixes it.

vespino · January 26, 2022, 7:50pm

What exactly does this setting?

david55 · January 26, 2022, 7:58pm

Whilst the following is for the deprecated driver, because the description is more complete, the current, chan_pjsip, driver has a similar option:

github.com

asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L293

      
        
            ;cos_text=3                     ; Sets 802.1p priority for RTP text packets.
            
            
;maxexpiry=3600                 ; Maximum allowed time of incoming registrations (seconds)
            ;minexpiry=60                   ; Minimum length of registrations (default 60)
            ;defaultexpiry=120              ; Default length of incoming/outgoing registration
            ;submaxexpiry=3600              ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
            ;subminexpiry=60                ; Minimum length of subscriptions, default: minexpiry
            ;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
            ;maxforwards=70			; Setting for the SIP Max-Forwards: header (loop prevention)
            				; Default value is 70
            ;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
            				; and reported in milliseconds with sip show settings.
                                            ; Set to low value if you use low timeout for NAT of UDP sessions
            				; Default: 60
            ;qualifygap=100			; Number of milliseconds between each group of peers being qualified
            				; Default: 100
            ;qualifypeers=1			; Number of peers in a group to be qualified at the same time
            				; Default: 1
            ;keepalive=60                   ; Interval at which keepalive packets should be sent to a peer
            				; Valid options are yes (60 seconds), no, or the number of seconds.
                                            ; Default: 0

vespino · January 26, 2022, 9:53pm

Is it the server that checks the extension or the extension that checks the server? And how does this effect the phone the extension is on?

david55 · January 26, 2022, 10:35pm

The host refers to the host parameter in sip.conf (or pjsip.conf), which is the remote device, the phone in your case.

The device receives more traffic than it would, with the default parameters.

If the host actually represents an ITSP (trunk in FreePBX terms), the host may be doing a similar thing to make sure that Asterisk is still there.