A: We restored from a recent FreePBX full backup and restarted the system. This cleared the problem.
Q:
We have a FreePBX system that has been running for two years. Last week we shut it down to clone the HDD as we suspected that the drive was about to fail from the smartctl reports. This turned out to be an error in interpretation due to the way Seagate report raw error rates.
However, after restarting the system we began to experience intermittent cut outs to audio reception on the extensions. We have made no direct changes to any of the configuration files, everything has been handled through the web interface. In any case there were no changes made there either since the last restart of Asterisk before the reboot.
After the reboot we also had problems with the fax lines not picking up. This was traced back to a corrupted astdb.sqlite3 database that was dealt with through the web interface by simply resubmitting the device and applying the configuration.
This morning in an attempt to clear up any residual sqlite3 problems we shut down asterisk and renamed astdb.sqlite3, and restarted asterisk to recreate the database from scratch. This asterisk did do. However, it also brought on the symptom where none of the internal extensions can accept calls, either internal or external. Outgoing calls are not affected.
This has killed our pbx. We have run fwconsol stop and fwconsole start but nothing has changed insofar as we can see.
This is the output fromfwconsole start:
fwconsole start
Running FreePBX startup…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
27381 [============================]
Finished setting permissions
Running Asterisk pre from Dahdiconfig module
Wanrouter: No valid Sangoma Hardware found, if you have no Sangoma cards this is OK
Starting DAHDi for Digium Cards
DAHDi Started
Running Asterisk pre from Firewall module
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Restarting fail2ban
fail2ban Restarted
Updating License Information for 15423705
Checking Vpn server
Starting Asterisk…
[============================] 34 secs
Asterisk Started
Running Asterisk post from Dahdiconfig module
Running Asterisk post from Endpoint module
Running Asterisk post from Pagingpro module
Running Asterisk post from Restapps module
Starting RestApps Server…
[>---------------------------] 4 secs
Started RestApps Server. PID is 9940
Running Asterisk post from Ucpnode module
Starting UCP Node Server…
[>---------------------------] 9 secs
Started UCP Node Server. PID is 10017
Running Asterisk post from Vqplus module
RestApps is not licensed.
Running Asterisk post from Xmpp module
[>---------------------------] < 1 secStarting Chat Server…
[>---------------------------] 7 secs
Started Chat Server. PID is 10272
Running Asterisk post from Zulu module
This product is not licensed
Important extension data is stored in astdb. You should be able to get most of it back by going to each of your extensions and re-saving, then applying the config. Users’ DND and call forward settings will be lost.
Checking out /etc/from prior to the HDD clone does not change the behaviour. The person at extension . . . plays regardless. I have located where in the call sequence the problem manifests but there no other information.
Processing of what? Did you go through and re-save all the extensions? If you do database show at the Asterisk CLI do you see a listing of AMPUSER and DEVICE keys, among others?
# asterisk -r
Asterisk 13.19.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.19.1 currently running on voinet09 (pid = 16866)
voinet09*CLI> database show
/AMPUSER/41710/answermode : disabled
/AMPUSER/41710/outroute : restrictedroute-2-3-4-5-6-7-8-10-14-15
/AMPUSER/41710/outroute_hash : restrictedroute-b21c639e50e125f6cb7570b0f93ddaea
/AMPUSER/41711/answermode : disabled
/AMPUSER/41711/followme/annmsg :
/AMPUSER/41711/followme/dring :
/AMPUSER/41711/followme/grppre :
/AMPUSER/41711/followme/postdest : ext-local,vmu41711,1
/AMPUSER/41711/followme/remotealertmsg :
/AMPUSER/41711/followme/ringing : Ring
/AMPUSER/41711/followme/rvolume : 0
/AMPUSER/41711/followme/strategy : ringallv2-prim
/AMPUSER/41711/followme/toolatemsg :
/AMPUSER/41711/outroute : restrictedroute-2-3-4-5-6-7-8-10-14-15-16
/AMPUSER/41711/outroute_hash : restrictedroute-fb975d0ada0ab5ee43469a98eb69ec73
/AMPUSER/41712/answermode : disabled
/AMPUSER/41712/followme/annmsg :
/AMPUSER/41712/followme/dring :
/AMPUSER/41712/followme/grppre :
/AMPUSER/41712/followme/postdest : ext-local,41712,dest
/AMPUSER/41712/followme/remotealertmsg :
/AMPUSER/41712/followme/ringing : Ring
/AMPUSER/41712/followme/rvolume :
/AMPUSER/41712/followme/strategy : ringallv2-prim
/AMPUSER/41712/followme/toolatemsg :
/AMPUSER/41712/outroute : restrictedroute-2-3-4-5-6-7-8-10-14-15
/AMPUSER/41712/outroute_hash : restrictedroute-b21c639e50e125f6cb7570b0f93ddaea
. . .
/DEVICE/41710/default_user : 41710
/DEVICE/41710/dial : SIP/41710
/DEVICE/41710/type : adhoc
/DEVICE/41711/default_user : 41711
/DEVICE/41711/dial : SIP/41711
/DEVICE/41711/type : adhoc
/DEVICE/41712/default_user : 41712
/DEVICE/41712/dial : SIP/41712
/DEVICE/41712/type : adhoc
/IAX/Registry/4570 : 127.0.0.1:4570:120
/IAX/Registry/4571 : 127.0.0.1:4571:120
/SIP/Registry/41711 : 192.168.6.111:4425:3600:41711:sip:[email protected]:4425;transport=tls;line=mx64ede8
/SIP/Registry/41712 : 192.168.6.112:4285:3600:41712:sip:[email protected]:4285;transport=tls;line=lribn4wt
/SIP/Registry/41713 : 192.168.6.113:4456:3600:41713:sip:[email protected]:4456;transport=tls;line=53lyd75r
/SIP/Registry/41714 : 192.168.6.114:3766:3600:41714:sip:[email protected]:3766;transport=tls;line=vrq1lv2d
/SIP/Registry/41715 : 192.168.6.115:4377:3600:41715:sip:[email protected]:4377;transport=tls;line=dl5vgag2
/SIP/Registry/41716 : 192.168.6.116:3127:3600:41716:sip:[email protected]:3127;transport=tls;line=ljlo09wd
/SIP/Registry/41717 : 192.168.6.117:4330:3600:41717:sip:[email protected]:4330;transport=tls;line=qyb7ieoe
/SIP/Registry/41718 : 192.168.6.118:3207:3600:41718:sip:[email protected]:3207;transport=tls;line=oqx96an3
/SIP/Registry/41719 : 192.168.6.119:4510:3600:41719:sip:[email protected]:4510;transport=tls;line=ysqtj947
/SIP/Registry/41721 : 192.168.6.121:4283:3600:41721:sip:[email protected]:4283;transport=tls;line=4klwn6vn
/SIP/Registry/90017 : 99.226.129.44:50139:600:90017:sip:[email protected]:50139;transport=tls;registering_acc=harte-lyne_ca
I am only configuring 41710,11 and 12 at the moment since that is what I am using for testing. I do not understand what /AMPUSER/41712/answermode : disabled means but I suspect that it is not good.
That extension is simply not connected to the LAN at the moment. There are a number of other extensions that are also disconnected. This has never been a problem before.
The hang that occurs when the Apply button is pressed occurs here: [2020-06-12 13:48:27] VERBOSE[17055] config.c: Parsing '/etc/asterisk/sip_notify_additional.conf': Found
It may be that this is not a hang at all and I am misinterpreting what tail /var/log/asterisk/full is telling me. I could be that the file is parsed successfully and that there is noting else to do. But generally, after this point asterisk goes out and polls the extensions and this is not being logged.
The contents of /etc/asterisk/sip_notify_additional.conf are:
# cat /etc/asterisk/sip_notify_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[aastra-check-cfg]
Event=>check-sync
[aastra-xml]
Event=>aastra-xml
[cisco-check-cfg]
Event=>check-sync
[cortelco-check-cfg]
Event=>check-sync
[digium-check-cfg]
Event=>check-sync
[grandstream-check-cfg]
Event=>check-sync
[htek-check-cfg]
Event=>check-sync
[linksys-cold-restart]
Event=>reboot_now
[linksys-warm-restart]
Event=>restart_now
[obihai-check-cfg]
Event=>sync
[polycom-check-cfg]
Event=>check-sync
[polycom-reboot]
Event=>check-sync
[sangoma-check-cfg]
Event=>check-sync
[sync-noreboot-sangoma]
Event=>check-sync\;reboot=false
[sync-reboot-sangoma]
Event=>check-sync\;reboot=true
[sync-hints-sangoma]
Event=>server_restart
[reboot-sangoma]
Event=>reboot
[sipura-check-cfg]
Event=>resync
[reboot-snom]
Event=>reboot
[spa-reboot]
Event=>reboot
[reboot-yealink]
You’re missing AstDB entries still. A healthy extension will have at least 2 dozen ampuser entries, and critically will have an AMPUSER/xxxx/device entry, i.e.: