I have a FreePBX server that needs to have a second real-world interface added. As in, both eth0 and eth1 have routable ip addresses. Setting up the second address was relatively simple as per a quick google search: https://www.thomas-krenn.com/en/wiki/Two_Default_Gateways_on_One_System 1. edit /etc/iproute2/rt_tables and added an entry "200 second" 2. added routes for the table called "second": ip route add ip.subnet.on.eth1/27 dev eth1 src ip.address.on.eth1 table second ip route add default via router.ip.for.eth1 dev eth1 table second (where "ip.subnet.on.eth1" "ip.address.on.eth1" and "router.ip.for.eth1" are the addresses needed) 3. added rules to send traffic through the right interface via source-routing: ip rule add from ip.address.on.eth1/32 table second ip rule add to ip.address.on.eth1/32 table second Which gave me some partial success: I could now route to the server's second ip address, such that it would immediately respond to normal requests (ie. icmp, sshd, apache, etc). But it is utterly failing to work with the udp services needed for Asterisk. Any client can connect fine to the "normal" ip address on eth0. But trying to connect to the eth1's ip, the client just times out. I even tried setting the server to accept tcp for sip, which then allowed me to log in; but it was ultimately unsuccessful, as the rtp (voice) traffic is also udp. Now, similar setups DO work, such as when the second eth1 device doesn't need to be globally routable, ie. for an internal LAN. So if the pbx has one real-world ip on eth0, and a private ip on eth1, then it's only using a single unified routing table and everything just works. I can definitively confirm that there is NO firewall blocking traffic to/from the sip server. In fact I've even got an explicit set of entries allowing ALL tcp and udp traffic to my sip client's subnet, bypassing all the fail2ban and other shenanigans. PS. if I haven't made myself pedantically clear, the end-goal is to have the Freepbx/asterisk functioning on BOTH interfaces eth0 & eth1, which both have fully-routeable real-world ip addresses. As it currently stands, only eth0 is functional

I configured wifi on FreePBX. When I run FreePBX I see wlan0 and ipv6 address. I can ping that address from other machine from the same LAN network. iwconfig: wlan0 IEEE 802.11 ESSID:"my name ssid" Mode:Managed Frequency:2.412 GHz Access Point: xx:xx:xx:xx:xx Bit Rate=86.7 Mb/s Tx-Power=22dBm Retry short limit:7 RTS thr:off Power Management:on Link Quality=40/70 Signal level=-70dBm Rx invalid nwid:0 Rx invalid crypt:0 Rx invalid frag:0 Tx excessive retries:0 Invalid misc: 19 Missed beacon:0 I called commands: dhclient -r dhclient wlan0 And when I type ifconfig I have IPv4 address assigned to wlan0. But I can't ping that address from other machine in the same LAN network. I can't connect using webbrowser too.

I have computer that only have Wifi network. There is no RJ45 socket. I have to configure Wifi using command line. My Wifi: Authentication : WPA2-Personal Cipher : CCMP Connection mode : Profile Channel : 1 On FreePBX I have file: /etc/wpa_supplicant/wpa_supplicant.conf: ctrl_interface=/var/run/wpa_supplicant ctrl_interface_group=wheel network={ ssid="mynetworkname" psk="mypassword" } When I type command: wpa_supplicant -B -i wlan0 -c wpa_supplicant.conf -D wext and I get error: Successfully initialized wpa_supplicant Failed to open config file '/etc/wpa_supplicant.conf', error: No such file or directory Delete '/var/run/wpa_supplicant/wlan0' manually if it is not used anymore Failed to initialize control interface '/var/run/wpa_supplicant'. You may have another wpa_supplicant process already running or the file was left by an unclean termination of wpa_supplicant in which case you will need to manually remove this file before starting wpa_supplicant again.

Using FreePBX 15. This is a test system for evaluating the possible use of FreePBX in our university environment. From a phone, I used the Asterisk star code *75 to enter a speed dial. Now I want to delete that. I cannot find any way to do it. Asterisk doesn't provide a cancellation code. I checked the Asterisk Phonebook, and my entry is not there, yet it is still active when I dial the speed code from a phone. Any help with this would be much appreciated. Thanks,

I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m back to asterisk. I installed asterisk on my debian VPS, then installed free PBX. At first i was having issues with getting the freePBX installer to find my asterisk server. After creating an aserisk user on debian and changing the run user-group in /etc/default/asterisk the installer worked. After the FreePBX installation, localhost/ was redirecting to localhost/admin/config.php but was only showing a blank screen. After running fwconsole ma installall the page started working. Despite that, connection to asterisk cannot be established. Running fwconsole start works just fine, but when running fwconsole restart I get get UCP Node Server is not running. When running fwconsole restart again I get Core FastAGI Server is not running & UCP Node Server is not running. **After checking logs for freepbx I found this:**

[2020-Nov-23 14:42:32] [freepbx.INFO]: Connection attmempt to AMI failed [] []

**The UCP error log in asterisk yielded this:**

2020-11-23 13:40 +00:00: { [Error: Can’t connect to MySQL server on ‘::1’ (111 “Connection refused”)] code: 2003 }

2020-11-23 13:40 +00:00: There was an error with MySQL Connection

**ucp_out.log in asterisk displays this:**

2020-11-23 13:40 +00:00: Starting FreePBX…
2020-11-23 13:40 +00:00: { AMPDBUSER: ‘freepbxuser’,
2020-11-23 13:40 +00:00: AMPDBPASS: ‘****’,
2020-11-23 13:40 +00:00: AMPDBHOST: ‘localhost’,
2020-11-23 13:40 +00:00: AMPDBNAME: ‘asterisk’,
2020-11-23 13:40 +00:00: AMPDBENGINE: ‘mysql’,
2020-11-23 13:40 +00:00: datasource: ‘’ }

I am using mariadb, is there a way maybe this user or database does not exist? Thank you for your help! --- Forgot to mention that when accessing Advanced Setting in FreePBX I get a blank page.

I needed a freepbx lxd image for some test, and since I didn't find it in public image servers (well, default images: one and google search but nothing, I think for licence's issues but don't really know). I tried to create one by installing FreePBX 15 on virtualbox, mounting the vdi (via qemu-nbd), tarballing it, creating a simple metadata.yaml and importing it on the server, but * when I launch it seems to be in a broken state * when I launch

exec freepbxtest bash

it starts but I can see mysql is down, and *

start mariadb

complains about polkit not running What can be the problem? I don't really want to run it as a vm. The lxd host is arch linux.

I'm currently using a Raspberry PI 4 to run FreePBX. I'm using raspbx to run it. So far everything was going good for a few days. Today I started getting a lot of problems receiving calls. I logged into the freepbx GUI and saw that there was no disk space available. So I ssh'd into my pi and ran df -h. I see that /dev/root was at 100%. I emptied out the database cdr and removed log and temporary files. Still only down to 80%. I now see that there's only 3.5 GB on this /dev/root partition. My SD card is 32 GB. So I ran fdisk -l to see where all of my space went. How do I extend my partition to use that space on my SD card? A screenshot of fdisk -l https://gyazo.com/5fb9e080a03f4b8d6d18fb6f0e075867 A screenshot of df -h

I've Sangoma Linux 7.5.1805 I installed a second ethernet card into motherboardls and since then the integrated interface stopped working As I plug cable into the second NIC the network starts to work again. ifconfig shows me 2 interface eth0 and eth1 I even have 2 ifcfg files **/etc/sysconfig/network-scripts/ifcfg-eth0**

DEVICE=eth0
BOOTPROTO=static
ONBOOT='yes'
IPADDR=19.2168.16.98
NETMASK=255.255.255.0
GATEWAY=192.168.16.2
DESCRIPTION="unset"
TYPE=Ethernet
ZONE=trusted
PEERDNS=no
IPV6_PEERDNS=no

**/etc/sysconfig/network-scripts/ifcfg-eth1**

DEVICE=eth1
BOOTPROTO=static
ONBOOT='yes'
IPADDR=19.2168.16.216
NETMASK=255.255.255.0
GATEWAY=192.168.16.2
ZONE=trusted
TYPE=Ethernet
PEERDNS=no
IPV6_PEERDNS=no

**ipconfig** I also can ping the 192.168.16.98 from another computer **lspci -vv | grep net** How can I make both cards work?

I created a ring group on freepbx, for the option no answer I set the destination to this ring group that I created, just to be like a loop 'til someone hangup. The problem... When someone calls to this ring group, asterisk register everything in the database, in other words, the 1st extension that rangs asterisk saves to database, the 2nd that rangs asterisk still saves to database and so on, I know that this is right, but **there is a way to config that asterisk only register or save the last attempt?**

I am trying to find out what mail service is being used to send voicemail to email on Centos 7 / FreePBX14 Distro.

We run a FreePBX server on our LAN and softphones can register using the local SIP server IP. I need these softphones to be able to register over the internet too so we have configured the firewall and created a dns entry for sip.ourdomain.com. When the softphones are configured to use sip.ourdomain.com then can register over the internet fine however when they are in the office and are connected to the wifi they are unable to register. I suspect this is because when in the office they are trying to register to sip.ourdomain.com which resolves to the public IP that redirects to the sip server on the local LAN. How can this be resolved? **Edit1** LAN is 192.168.1.X/24 & SIP Server is 192.168.1.8

I want to turn off buffering of SIP calls in freeswitch pbx software. Freeswitch holds RTP data from clients in buffer and sends it every 20ms. I want to freeswitch pass throught packets without holding. How to configure it? ----- EDIT (additional info) ----- I have two SIP clients and FreeSwitch PBX. Voice 8 kHz sample rate, A-Law coding (8 bytes per sample, no compression) When I call directly from one client to another the tcpdump output on one client is: 00:00:00.000475 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031599 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.032012 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000315 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031775 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000384 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031499 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031986 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000475 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031578 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031936 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.000419 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 00:00:00.031573 IP 10.64.177.34.5440 > 10.64.0.42.5082: UDP, length 172 But when I connect from one client to another using pbx as middle point I got: 00:00:00.020013 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019969 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020017 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019984 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020078 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020016 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019850 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020045 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020012 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019974 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.020054 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019996 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 00:00:00.019972 IP 10.64.72.24.30230 > 10.64.0.42.5078: UDP, length 172 The average time in both cases is aproximate 20ms (slighter less in direct case), but non regular data portions seems better for client because there is no gaps in heard voice (in speaker or headphones). I suppose that regular period portions of data cause problems due to clock drift problem. **So I want to turn off this feature in FreeSwitch, so the data will come in original timestamps.**

I have used the same configuration to successfully add a second menu entry and boot to it via "grub2-reboot # && reboot" in CentOS 7 but nothing I do seems to change the menu entry selection for next boot in the FreePBX Distro 7. Any ideas on what to try to troubleshoot/test/fix? Configs: **cat /etc/default/grub** [root@freepbx ~]# cat /etc/default/grub GRUB_TIMEOUT=5 GRUB_DISTRIBUTOR="$(sed 's, release .*$,,g' /etc/system-release)" GRUB_DEFAULT=saved GRUB_DISABLE_SUBMENU=true GRUB_TERMINAL_OUTPUT="gfxterm" GRUB_CMDLINE_LINUX="crashkernel=auto biosdevname=0 net.ifnames=0 rhgb quiet" GRUB_DISABLE_RECOVERY="true" GRUB_BACKGROUND=/boot/splash.png **cat /boot/grub2/grub.cfg** [root@freepbx ~]# cat /boot/grub2/grub.cfg # # DO NOT EDIT THIS FILE # # It is automatically generated by grub2-mkconfig using templates # from /etc/grub.d and settings from /etc/default/grub # ### BEGIN /etc/grub.d/00_header ### set pager=1 if [ -s $prefix/grubenv ]; then load_env fi if [ "${next_entry}" ] ; then set default="${next_entry}" set next_entry= save_env next_entry set boot_once=true else set default="${saved_entry}" fi if [ x"${feature_menuentry_id}" = xy ]; then menuentry_id_option="--id" else menuentry_id_option="" fi export menuentry_id_option if [ "${prev_saved_entry}" ]; then set saved_entry="${prev_saved_entry}" save_env saved_entry set prev_saved_entry= save_env prev_saved_entry set boot_once=true fi function savedefault { if [ -z "${boot_once}" ]; then saved_entry="${chosen}" save_env saved_entry fi } function load_video { if [ x$feature_all_video_module = xy ]; then insmod all_video else insmod efi_gop insmod efi_uga insmod ieee1275_fb insmod vbe insmod vga insmod video_bochs insmod video_cirrus fi } if [ x$feature_default_font_path = xy ] ; then font=unicode else insmod part_msdos insmod ext2 set root='hd0,msdos2' if [ x$feature_platform_search_hint = xy ]; then search --no-floppy --fs-uuid --set=root --hint-bios=hd0,msdos2 --hint-efi=hd0,msdos2 --hint-baremetal=ahci0,msdos2 --hint='hd0,msdos2' 5820774d-3fcf-44fb-86b9-972fab5de985 else search --no-floppy --fs-uuid --set=root 5820774d-3fcf-44fb-86b9-972fab5de985 fi font="/usr/share/grub/unicode.pf2" fi if loadfont $font ; then set gfxmode=auto load_video insmod gfxterm set locale_dir=$prefix/locale set lang=en_US insmod gettext fi terminal_output gfxterm insmod part_msdos insmod ext2 set root='hd0,msdos1' if [ x$feature_platform_search_hint = xy ]; then search --no-floppy --fs-uuid --set=root --hint-bios=hd0,msdos1 --hint-efi=hd0,msdos1 --hint-baremetal=ahci0,msdos1 --hint='hd0,msdos1' 0123444f-dbb6-421e-ad8a-6f6bfb0a0901 else search --no-floppy --fs-uuid --set=root 0123444f-dbb6-421e-ad8a-6f6bfb0a0901 fi insmod png background_image -m stretch /splash.png if [ x$feature_timeout_style = xy ] ; then set timeout_style=menu set timeout=5 # Fallback normal timeout code in case the timeout_style feature is # unavailable. else set timeout=5 fi ### END /etc/grub.d/00_header ### ### BEGIN /etc/grub.d/00_tuned ### set tuned_params="" ### END /etc/grub.d/00_tuned ### ### BEGIN /etc/grub.d/01_users ### if [ -f ${prefix}/user.cfg ]; then source ${prefix}/user.cfg if [ -n "${GRUB2_PASSWORD}" ]; then set superusers="root" export superusers password_pbkdf2 root ${GRUB2_PASSWORD} fi fi ### END /etc/grub.d/01_users ### ### BEGIN /etc/grub.d/10_linux ### menuentry 'Sangoma Linux (3.10.0-514.2.2.el7.x86_64) 7 (Core)' --class sangoma --class gnu-linux --class gnu --class os --unrestricted $menuentry_id_option 'gnulinux-3.10.0-514.2.2.el7.x86_64-advanced-5820774d-3fcf-44fb-86b9-972fab5de985' { load_video set gfxpayload=keep insmod gzio insmod part_msdos insmod ext2 set root='hd0,msdos1' if [ x$feature_platform_search_hint = xy ]; then search --no-floppy --fs-uuid --set=root --hint-bios=hd0,msdos1 --hint-efi=hd0,msdos1 --hint-baremetal=ahci0,msdos1 --hint='hd0,msdos1' 0123444f-dbb6-421e-ad8a-6f6bfb0a0901 else search --no-floppy --fs-uuid --set=root 0123444f-dbb6-421e-ad8a-6f6bfb0a0901 fi linux16 /vmlinuz-3.10.0-514.2.2.el7.x86_64 root=UUID=5820774d-3fcf-44fb-86b9-972fab5de985 ro crashkernel=auto biosdevname=0 net.ifnames=0 rhgb quiet initrd16 /initramfs-3.10.0-514.2.2.el7.x86_64.img } if [ "x$default" = 'Sangoma Linux (3.10.0-514.2.2.el7.x86_64) 7 (Core)' ]; then default='Advanced options for Sangoma Linux>Sangoma Linux (3.10.0-514.2.2.el7.x86_64) 7 (Core)'; fi; ### END /etc/grub.d/10_linux ### ### BEGIN /etc/grub.d/20_linux_xen ### ### END /etc/grub.d/20_linux_xen ### ### BEGIN /etc/grub.d/20_ppc_terminfo ### ### END /etc/grub.d/20_ppc_terminfo ### ### BEGIN /etc/grub.d/30_os-prober ### ### END /etc/grub.d/30_os-prober ### ### BEGIN /etc/grub.d/40_custom ### # This file provides an easy way to add custom menu entries. Simply type the # menu entries you want to add after this comment. Be careful not to change # the 'exec tail' line above. menuentry "Parted Magic 64-bit LiveISO" { set root=(hd0,msdos2) set isofile="/home/isos/pmagic.iso" loopback loop $isofile linux (loop)/pmagic/bzImage64 iso_filename=$isofile load_ramdisk=1 initrd (loop)/pmagic/initrd.img /pmagic/fu.img /pmagic/m64.img } ### END /etc/grub.d/40_custom ### ### BEGIN /etc/grub.d/41_custom ### if [ -f ${config_directory}/custom.cfg ]; then source ${config_directory}/custom.cfg elif [ -z "${config_directory}" -a -f $prefix/custom.cfg ]; then source $prefix/custom.cfg; fi ### END /etc/grub.d/41_custom ### I've also tried the following and get the same results: sudo rm /boot/grub2/grub.cfg sudo rm /etc/grub.d/* -R sudo rm /etc/sysconfig/grub yum reinstall -y grub2-tools grub2-mkconfig -o /boot/grub2/grub.cfg grub2-install /dev/sda So FreePBX Uses there own Repo, so I followed the above code, but manually downloaded grub2-tools, now I get the follow error: [root@freepbx grub]# grub2-install --target=x86_64 grub2-install: error: /usr/lib/grub/x86_64/modinfo.sh doesn't exist. Please specify --target or --directory. [root@freepbx grub]# grub2-install /dev/sda grub2-install: error: /usr/lib/grub/i386-pc/modinfo.sh doesn't exist. Please specify --target or --directory.

Per telecomworld : > FreePBX Operator Panel (FOP) > > The secret code for performing transfers and hangups in the Flash > Operator Panel is passw0rd (zero not the letter O) I'm on Elastix 2.5 and am just curious about this -- it seems a questionable choice to set any password to a static value. The MySQL password is set to this value for either FreePBX or PIAF (PBX In A Flash). Why? In Elastix 2.5 running elastix-admin-passwords --change invokes: #!/bin/bash clear echo "This script resets your MySQL password. IT MUST BE passw0rd!" echo "Script also changes your admin password for Elastix MT access." read -p "To continue, press Enter. Otherwise, press Ctrl-C to abort." echo " " elastix-admin-passwords --change echo " " echo "Done. Access Incredible PBX at http:///sbin/ifconfig eth0 | grep "inet " | cut -f 2 -d ":" | cut -f 1 -d " "" echo " "​ I'm as much interested in how that came about as to whether or not it's a "good" idea.

I am trying to solve a problem with my IVR, i am trying to figure out how to continue the call when the clerk hangup, to a evaluation. Well... I discovered that the queue command can do that with some specific option, the thing is that the queue command breaks asterisk using 100% of CPU... (so the queue command is not working at all) If i replace this working code: exten => 1235,n,Goto(ext-queues,2026,1) By this: exten => 1235,n,queue(2026,c) The queue looks fine, the music start playing, but there is no phone ringing..... and the CPU goes to 100% braking asterisk. In FreePBX forum i get some feedback saying that the thing i am trying to do is not possible in FreePBX context in the new FreePBX versions, so how to escape this context or do some kind of downgrade, or edit some embedded script to solve this issue? There is some alternative for me to use queue command, or to made the program to continue before the clerk answer hangup, and the asterisk program continues executing?