Fusionpbx Docs

FusionPBX Documentation
Release master
Mark J Crane
Jul 30, 2017

Contents
1 FusionPBX 3
2 Benefits of FusionPBX 5
3 FusionPBX Features 7
4 Our Ecosystem 9
4.1 Getting Started . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
4.2 Home . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
4.3 Accounts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4.4 Dialplan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
4.5 Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
4.6 Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
4.7 Advanced . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
4.8 Additional Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
4.9 Contributing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
4.10 Documentation Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
Bibliography 135
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FusionPBX Documentation, Release master
Contents 1
2 Contents
CHAPTER 1
FusionPBX
An open source project that provides a customizable and flexible web interface to the very powerful and highly scalable
multi-platform voice switch called FreeSWITCH.
FusionPBX will run on a variety of operating systems (Optimized for Debian 8) and hardware of your choice. Fu-
sionPBX provides a GUI for unlimited extensions, voicemail-to-email, music on hold, call parking, analog lines or
high density T1/E1 circuits, and many other features. FusionPBX provides the functionality that business need and
provides corporate level phone system features to small, medium and large businesses. Click here for the FusionPBX
youtube channel .
3
4 Chapter 1. FusionPBX
CHAPTER 2
Benefits of FusionPBX
1. Adding extra functionality to the incredibly robust FreeSWITCH VoIP Platform.

2. Makes FreeSWITCH easy to administer while at the same time still allowing you to work directly within
FreeSWITCH Command Line Interface (fs_cli) when you need to.
3. Gives your users and tenants an attractive GUI interface to interact with.
5
6 Chapter 2. Benefits of FusionPBX

CHAPTER 3
FusionPBX Features
Call Block Call Broadcast Call Flows Call Center

Call Detail Records Conference Contacts Fax Server
Follow Me Hot Desking IVR Menus Ring Groups
Multi-Tenant Music on Hold Queues Recordings
Time Conditions WebRTC ready Voicemail and lots more...
7
8 Chapter 3. FusionPBX Features

CHAPTER 4
Our Ecosystem
We are a global community that has an open and very friendly ecosystem. We encourage community engagement,
contribution and feedback. Please join us by getting involved with giving feedback, new feature ideas, helping out
with code or Documentation.
Most of the core folks who develop and use FusionPBX can be found hanging out in Freenode IRC in the #fusionpbx
channel. Come join us and meet the team.
Getting Started
Welcome! Lets install FusionPBX. Follow the menu to the left and you will have a working PBX in no time. For PDF
and Epub formats of this documentation click the bottom left on v:latest and a menu will pop-up to choose from.
Note: There are many ways to install FusionPBX depending on how you want to build your solution. What is
presented here represents the quickest, easiest, best supported way to a FusionPBX system. For advanced topics like
Bi Directional Replication or High Availability, consider attending the in person or online training at http://fusionpbx.
com.
Quick Install
9
Welcome to the FUSIONPBX installation guide.
FUSIONPBX can be several different operating systems. However this install is focused on a minimal install of
Debian 8 with SSH enabled. This install has been designed to be fast, simple and modular. On many systems it will
install in 5 minutes or less. Installation times depend on CPU, RAM and bandwidth. Install Video https://youtu.be/
YmIht8hEHYU
1. Run the following commands under root. The script installs FusionPBX, FreeSWITCH release package and its
dependencies, IPTables, Fail2ban, NGINX, PHP FPM and PostgreSQL.
Start with a minimal install of Debian 8 with SSH enabled. Paste the following commands in the console window one
line at a time.
apt-get update && apt-get upgrade -y --force-yes

apt-get install -y --force-yes git
cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx-install.sh.git
chmod 755 -R /usr/src/fusionpbx-install.sh
cd /usr/src/fusionpbx-install.sh/debian
./install.sh
If using Debian on Proxmox LXC containers please run the following BEFORE starting the FusionPBX install.
apt-get update && apt-get upgrade

apt-get install systemd
apt-get install systemd-sysv
apt-get install ca-certificates
reboot
2. At the end of the install, the script will instruct you to go to the ip address of the server (or domain name) in your
web browser to login. The script will also provide a username and secure random password for you to use. This can
be changed after you login. The install script builds the fusionpbx database. If you need the database password it is
located in /etc/fusionpbx/config.php .
10 Chapter 4. Our Ecosystem

Installation has completed.
Use a web browser to login.

domain name: https://000.000.000.000
username: admin
password: zxP5yatwMxejKXd
The domain name in the browser is used by default as part of the authentication.
If you need to login to a different domain then use username@domain.
username: admin@000.000.000.000
Official FusionPBX Training

Admin Training 24 - 26 Jan (3 Days)
Advanced Training 31 Jan - Feb 2 (3 Days)
Timezone: https://www.timeanddate.com/worldclock/usa/boise
For more info visit https://www.fusionpbx.com
Additional information.
https://fusionpbx.com/support.php
https://www.fusionpbx.com
http://docs.fusionpbx.com
After the install script has completed go to your web browser and login with the information provided by the install
script.
4.1. Getting Started 11

Security
Similar to medieval fortifications it is recommended to provide your servers with multiple layers of defenses. Be sure
to use Firewalls, Strong passwords, SSH, and make sure your servers are kept up to date for all software being used.
This includes the operating system, FreeSWITCH and FusionPBX.
FusionPBX
The latest Debian install script configures IPTables firewall for you. FusionPBX extensions set strong passwords for
you by default. You can increase the password complexity using settings in Advanced -> Default Settings to increase
the length of the passwords that are generated by default.
Firewall
Although the new install script configured IPTables for you it is recommended that you review the settings. On Debian
and Ubuntu you can check your firewall with the following command.
iptables -L
Firewall page
SSL / TLS
SSL and TLS are very necessary in todays internet applications from VOIP to Websites. FusionPBX by default uses a
self signed certificate. However you can use certificate providers where you can purchase certificates and there are free
options as well. With domain based multi-tenant wildcard certificates can be useful. Also when deciding on which
certificate provider to use you should look at the phones manufacturers documentation to find one that is compatible
HTTPS provisioning.
Lets Encrypt provides free certificates for a single domain but they dont support wildcard certificates.
Setup Lets Encrypt with FusionPBX
Upgrade
Security problems are fixed as they are discovered and are updated for master and the latest release. Upgrades are
considered an important part of keeping the server secure. Upgrades always need to be done on the operating system,
FreeSWITCH and FusionPBX. On Debian and Ubuntu you can check your firewall with the following command.
Latest install script will install FreeSWITCH packages by default to upgrade them and operating system packages run
the following commands.
apt-get update
apt-get upgrade
If you need help upgrading safely please consider paid support.
XML RPC
New install mod_xml_rpc is not enabled by default. It is recommended to run a firewall on all FusionPBX servers.
The latest debian install script configures the firewall by default. However it is recommended to check to make sure it
is installed and running.

Mod_xml_rpc allows running remote commands to FreeSWITCH. Ensure you have a firewall that is protecting the
XML RPC port. Consider changing the XML RPC password. At very least do not allow access to the public. Advanced
-> Settings page in the interface allows you to change the password or the port. Do not allow public access to the XML
RPC port.
Latest Debian install script installs iptables firewall which prevents public access to the mod_xml_rpc port. If you
are not using a firewall on the server you should even if its protected by by an external firewall. Some not informed
co-worker could expose the server to the public internet at some point in the future. Multiple layers of security is
considered best practice.
XML RPC is secure by default for 2 reasons.
The module is disabled by default.
Install script firewalls XML RPC port 8787 and does not allow access to it by default outside of 127.0.0.1.
If you were to start the module and open port 8787 on the firewall you would want to set a really good password for it
under Advanced -> Settings. It would be recommended to use a VPN to like OpenVPN to access XML RPC over port
8787 instead of opening port 8787 on the firewall.
Fail2ban
Fail2ban is also used to protect SSH, FreeSWITCH, the web server as well as other services. You can view the IP
addresses blocked by Fail2ban with the following command.
iptables -L
SSH
Use strong passwords with SSH or even better use SSH keys for better protection of your servers.
Backup
Its always good to have a backup method in place. Here are the steps to a basic backup method with FusionPBX.
Command line settings
Be sure to change the password by replacing the zzzzzzzz in PGPASSWORD=zzzzzzzz with your password.
cd /usr/src/fusionpbx-install.sh
git pull
cd debian/resources/backup/
vim fusionpbx-backup.sh
#!/bin/sh
now=$(date +%Y-%m-%d)
echo "Server Backup"
export PGPASSWORD="zzzzzzzz"
mkdir -p /var/backups/fusionpbx/postgresql
#delete postgres logs older than 7 days
find /var/log/postgresql/postgresql-9.4-main* -mtime +7 -exec rm {} \;

#delete freeswitch logs older 3 days

find /usr/local/freeswitch/log/freeswitch.log.* -mtime +2 -exec rm {} \;
pg_dump --verbose -Fc --host=$database_host --port=$database_port -U fusionpbx
fusionpbx --schema=public -f /var/backups/fusionpbx/postgresql/fusionpbx_pgsql_$now.
sql
echo "Backup Complete";
To save the file press escape then :wq for write and quite
You should have the script ready to execute. (Default the script will use FreeSWITCH package paths. If you have an
older install using source be sure to change this by commenting the package line #22 and uncomment the source line
#25.)
Crontab settings
Setting crontab -e
crontab -e
Choose 1 for nano
Goto the last blank line and paste in the next line.
0 0 * * * bash /etc/cron.daily/fusionpbx-backup.sh
press enter then save and exit.
cd /usr/src/fusionpbx-install.sh/debian/resources/backup/
cp fusionpbx-backup.sh /etc/cron.daily
chmod 755 fusionpbx-backup.sh
Once this is complete you will have the backup ready to execute by ./fusionpbx-backup.sh or from the daily cron job.
Gui settings
From the Gui.

FreeSWITCH Package install paths.
Goto Advanced > Default Settings.

Settings for FreeSWITCH package backup paths.
path array /var/backups/fusionpbx/postgresql True

postgresql
path array /usr/share/freeswitch/scripts True scripts

path array /var/www/fusionpbx True
fusionpbx
path array /var/lib/freeswitch/storage True storage

path array /var/lib/freeswitch/recordings True
recordings

path array /etc/freeswitch/conf True conf
Click "Reload" at the top of the page.
FreeSWITCH Source install paths.
Settings for FreeSWITCH source backup paths.
path array /var/backups/fusionpbx/postgresql True postgresql

path array /usr/local/freeswitch/scripts True scripts
path array /usr/local/freeswitch/recordings True recordings
path array /var/www/fusionpbx True fusionpbx
path array /usr/local/freeswitch/conf True conf
path array /usr/local/freeswitch/storage True storage
Click "Reload" at the top of the page.
Download Backups
From Advanced > Backup you can download the backup also.
FreeSWITCH Source install paths.

FreeSWITCH Package install paths.

Firewall
Basic ports used

SIP TCP/UDP
5060, 5061, 5080
RTP UDP
16384-32768
SSH
22
HTTP
80, 443
Iptables
Iptables are used in the Debian install script.
Basic Rules
iptables -A INPUT -i lo -j ACCEPT

iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
iptables -A INPUT -p tcp --dport 22 -j ACCEPT
iptables -A INPUT -p udp --dport 5060 -j ACCEPT
iptables -A INPUT -p udp --dport 5080 -j ACCEPT
iptables -A INPUT -p udp --dport 16384:32768 -j ACCEPT
iptables -P INPUT DROP
iptables -P FORWARD DROP
iptables -P OUTPUT ACCEPT
Optional Rules
OPENVPN: iptables -A INPUT -p udp --dport 1194 -j ACCEPT

ICMP: iptables -A INPUT -p icmp --icmp-type echo-request -j ACCEPT
Friendly Scanner
Rules to block not so friendly scanner
iptables -I INPUT -j DROP -p tcp --dport 5060 -m string --string

"friendly-scanner" --algo bm


iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
Optional
iptables -I INPUT -j DROP -p tcp --dport 5060 -m string--string

"VaxSIPUserAgent" --algo bm
"VaxIPUserAgent" --algo bm
"VaxSIPUserAgent" --algo bm
"VaxIPUserAgent" --algo bm
Show iptable rules
sudo iptables -L -v
Show line numbers
iptables -L -v --line-numbers
Delete a line
Delete line 2
iptables -D INPUT 2
Block IP address
iptables -I INPUT -s 62.210.245.132 -j DROP
Save Changes
Debian / Ubuntu
apt-get install iptables-persistent

service iptables-persistent save
dpkg-reconfigure iptables-persistent

PF
Packet Filter is used in the FreeBSD setup script.
Basic Rules
set skip on lo0

scrub in all
antispoof for lo0

table <fail2ban> persist
pass out quick all

pass quick on lo0 all
block in all
block in quick from <fail2ban>
pass in quick inet proto icmp all
pass in quick inet6 proto icmp6 all
pass in quick inet proto tcp from any to any port 22 keep state
pass in quick inet proto udp from any to any port 5060 keep state
pass in quick inet proto udp from any to any port 5080 keep state
pass in quick inet proto udp from any to any port 16384:32768 keep state
Firewall Devices
Firewall device settings that help with SIP connections.

SIP ALG- Most of the time this setting is set to off or disabled and varies. Rarely this will be enabled.
Home
4.2. Home 19
In the home menu you have the options for Account Settings, Logout and User Dashboard.
Accounts
In the Accounts menu you have access to devices, extensions, gateways and user manager.
Gateway
Gateways provide access into other voice networks. These can be voice providers or other systems that require SIP
registration. Check out the Youtube video .
In this example we will be using VoiceTel . Each Gateway provider has their own setings to use.
Click to visit
Select Accounts from the drop-down list and click on Gateways.

Click the
button on the right. Enter the gateway information below and Click on Save once complete.
Gateway: VoiceTel
Username: 0123456789
Password: 1b3d5f7h9j
From user: 0123456789
From domain: sbc.voicetel.com
Proxy: sbc.voicetel.com
Register: true
Enabled: true
4.3. Accounts 21
XMPP Manager

XMPP Manager is an optional menu item. In order to have the option for XMPP Manager there are a few step to take
to enble XMPP.
XMPP Profile
FusionPBX menu.
Accounts -> XMPP manager.
Click the
on the right to create a profile.
4.3. Accounts 23
In this example we will setup Google Talk and by creating a profile called gtalk.
Profile Name: gtalk

Username: your_user_account@gmail.com (use your account)
Password: use the correct password
Auto-Login: yes
XMPP Server: talk.google.com
Two approaches can be used for the next part.

Option 1.
Lets say my gmail number was 13051231234. This approach will send the inbound calls to the inbound routes with a
destination number that is the default extension number that is set.
Default extension: 13051231234

Advanced -> Context: public
Option 2.
On a single tenant system. This will send the call to extension 1001 in the default context.

Advanced -> Context: default
Option 3.
On a single tenant system. This will send the call to extension 1001 in the multi-tenant domain name.

Advanced -> Context: your.domain.com
Save the settings and restart the module. Restart the XMPP module from Advanced -> Modules page.
Go back to Accounts -> XMPP if the status says AUTHORIZED then you are ready to go.
Note If you are not getting AUTHORIZED you might need to goto the google account settings and choose Allow
less secure apps: ON under the Sign-in & security section.
4.3. Accounts 25
Outbound Routes
For this example we will use 11 digit dialing.
Gateway: XMPP
Dialplan Expression: 11 digits
Description: Google Talk
Press Save
If your XMPP profile is named something other than gtalk edit the outbound route you just created.
Bridge statement should look like: dingaling/gtalk/+$1@voice.google.com replace gtalk with the profile name you
chose and then save it.
Enable XMPP
XMPP manager is used to configure client side XMPP profiles. It can be used as a client to register to make and
receive call with Google Talk or other XMPP servers.
GIT Manually add XMPP
After version 3.8 XMPP is optional. To add XMPP do the following

Goto command line
cd /tmp
git clone https://github.com/fusionpbx/fusionpbx-apps.git
cd fusionpbx-apps/
mv xmpp/ /var/www/fusionpbx/app/
cd /var/www/fusionpbx/app
chown www-data:www-data -R xmpp/
Goto Fusionpbx GUI
Goto the GUI and click advanced > menu manager > edit icon > click Restore Defaults at top right
Then goto Advanced > Upgrade click Schema, Data Types, and Permission Defaults then click execute
Click status > sip status > Flush Memcache
Log out then back in
You should now have XMPP Manager under Accounts
Dialplan
In the Dialplan menu you have access to Destinations, Dialplan Manager, Inbound Routes and Outbound Routes.
4.4. Dialplan 27
Destinations
Inbound destinations are the DID/DDI, DNIS or Alias for inbound calls. Click here for the youtube video
Configure Inbound Destinations: (This will auto-configure an Inbound Route also)
Select Dialplan from the drop-down list and then click Destinations.
Click on the
button on the right.
Enter the route information below and Click Save once complete.

Type: Inbound
Destination Number: ^(?:\+?1)?(\d{10})$
Action: Select desired destination from the drop-down list. We choose "Extension 100
" in our example.
This is where the call will route to.

Enabled: true
Description: VoiceTel-in
Optional: Replace ^(?:\+?1)?(\d{10})$ in Inbound Routes with either 0123456789 or a

DID Number
depending on the Route Destination setting.
4.4. Dialplan 29
Dialplan Manager
The dialplan is used to setup call destinations based on conditions and context. You can use the dialplan to send calls
to gateways, auto attendants, external numbers, to scripts, or any destination.
Inbound Routes
Route incoming calls to destinations based on one or more conditions. It can send incoming calls to an IVR Menu,
Call Group, Extension, External Number, Script. Order is important when an anti-action is used or when there are
multiple conditions that match.
Inbound routes can be used for advanced reasons. Dialplan > Destinations will create and configure the Inbound
Route for you.
Outbound Routes
Route outbound calls to gateways, tdm, enum and more. When a call matches the conditions the call to outbound
routes. Check out the youtube video .
Configure Outbound Route.

Select Dialplan from the drop-down list and then click Outbound Routes .
Click the
button on the right. Enter the route information below and Click Save once entry is complete.

4.4. Dialplan 31
Gateway: VoiceTel
Dialplan Expression: ^(?:\+?1)?(\d{10})$ (You can also choose more than one from the
drop down list also as needed)
Order: 000
Enabled: true
Description: VoiceTel-out

By using VoiceTel you help support FusionPBX. Thank you for your support!
Applications
In the Applications menu (Apps) section you will find Call Block, Call Broadcast, Call Center, Call Detail Records,
Call Flows, Conference Center, Contacts, Fax Server, Follow Me, IVR Menu, Music on Hold, Operator Panel,
Phrases, Queues, Recordings, Ring Groups, Time Conditions and Voicemail.
4.5. Applications 33
Call block
A list of numbers from which to block calls.
To block a call click on the plus icon on the right

Fill out the fields with pertinent information
Action:
Reject- Will reject the call
Busy- Will send a busy signal
Hold- Will put the call on hold
Voicemail- Will send the call to the specified voicemail box
Call Broadcast
Broadcast calls (a light dialer) to a defined list of phone numbers.
To create a call broadcast click the plus on the right

Fill in the following fields

Name- Name for the Call Broadcast

Accountcode Used by some billing systems
Timeout- Amount of time till hangup
Concurrent Limit- Number of calls at once
Caller ID Name- Name that will be used on outbound caller id
Caller ID Number- Number that will be used on outbound caller id
Destination Number- Where the Phone Number List will connect to
Phone Number List- List of phone numbers to call in the call broadcast
Voicemail Detection- Set True or false to detect an answering machine
Description Help organize and label what the call broadcast is for
Once you have everything filled out click the Call Broadcast name you just created. On the top right click the
Send Broadcast button to start the call broadcast. To stop the call broadcast click STOP BROADCAST on the
top right.
Call Center
List of queues for the call center.
Call Center Queues
To add a Call Center Queue click the plus edit icon on the right
Once a Queue is created click the edit pencil icon on the right. At the top right you can view, stop, start, restart
and save the queue
Call Center Agents
List of call center agents.
From Apps > Call Center click Agents at the top right to access Call Center Agents
Click the plus icon on the top right to add agents
Call Detail Records
Call Detail Records (CDRs) are detailed information on the calls. Use the fields to filter the information for the specific
call records that are desired. Records in the call list can be saved locally using the Export button.

CID Name- Caller ID Name

Source- Where the call came from
Destination- Where the call went to
Recording- A link will appear if the call recorded
Start- Time the call entered the system
TTA- Time To Answer the call
Duration- How long the call was
PDD- Post Dial Delay
MOS- Mean Opinion Score is a measure of voice call quality
Hangup Cause- Details about the entire calls. Usually will be Normal Clearing
Call Flows
Direct calls between two destinations by calling a feature code.
Name: Define the name of the call flow

Extension: Define what extension to use. (This will make an extension not allready created)
Feature Code: Define what * number to use
Context: Domain context (typically leave as is)
Status: Define what currently is in use.
Pin Number: Define a pin number in order to execute either mode.
Destination: Define where the call will go in the intial mode.
Sound: Define the sound that will play once mode is engaged.
Destination: Define what the destination will be.
Alternative Label: Label that will show when alternative mode is in use.
Alternative Sound: Define the sound that will play once alternative mode is engaged.
Alternative Destination: Define where the call will go in the alternative mode.
Description: Label what this call flow does.
Call Flow Example
In the Call Flow example below we have the name as Call Flow. Made the Extension number 30 that didnt exist until
now. Feature code we made with a *code as *30. Kept the context as is with training.fusionpbx.com . Status to show
which mode. Made a pin number to help secure the call flow. Made the detination label as Day Mode. Picked a sound
to familiarize which mode is activated. Choose a destination for the alternative mode. Made the alternative detination
label as Night Mode. Picked an alternative sound to familiarize which mode is activated. Choose a destination for the
alternative mode. Finally describe what this call flow does.

Conference
Conferences is used to setup conference rooms with a name, description, and optional pin number.
Note: For advanced conferencing use Apps -> Conference Center
Enable Conference
By default Conferences are hidden from the menu.

To add Conferences to the menu goto Advanced > Menu Manager and click the pencil edit icon on the right
Then click the pencil edit icon on the right of Conferences
Select from the Groups dropdown list superadmin and click add then save
Conference Center
Conference Centers are a group of conference rooms. They can be organized by cost center, geographically, or other
criteria.
To Acess Conference Center goto Apps > Conference Center

To view rooms click the ROOMS at the top right.
Note: For basic conferencing use Apps -> Conferences
Contacts
Contacts is a list of individuals and organizations.

To create a contact click the plus and to edit a contact click the pencil icon on the right.
Fill out the fields with pertinent information and click save.
Users- Select the users that are allowed to view the contact
Groups- Select the group that are allowed access to the contact.
Go back into the contact to fill out more information that wasnt available when you first created the contact.
To generate a QR code click the QR CODE button at the top right

Fax Server
To receive a FAX setup a fax extension and then direct the incoming to it. Click here for the Youtube video
New: Create a new fax to send.

Inbox: Faxes received.
Sent: Faxes sent.
Log: Sucessful and failed attempts for both incoming and outgoing.
Active: Shows the faxes in queue.
Fax Server Settings
There are more settings for fax under Advanced > Default Settings then fax category.
To create a fax server goto App > Fax Server. Click the + on the right. Leave the Destination Number blank or
faxing wont work. Destination Number is used in the Fax Server Dial Plan and is set based on the fax server internal
extension number. Define the fields, the ones in bold are required. It is a good idea to organize so define the name
thoughtfully. The extension you must use one that is not allready created. Account Code should autofill. Again, leave
the Destination Number blank. A prefix can be defined when sending a fax. Email is for inbound faxes and will be
on the server and sent to the defines email. Define the Caller ID Name and Number. Leave the Forward Number and
Greeting blank for normal settings. Number of channels define with a numerical value. Keep organized by adding a
Description.

New
To send a fax the items in bold are required. To send a proper fax it is best to fill out all fields and attach any documents.
Keep in mind that the upload max MB is limited by Nginx and PHP config files.

Inbox
Click PDF to view the fax or right click on PDF and left click on Save Link As. If you defined and email address in
the email field you will receive the fax also to that email address.
Sent
Click PDF to view the fax or right click on PDF and left click on Save Link As.
Fax ATA
To connect to a fax machine with an ATA you will most likely need to adjust settings in the ATA web interface and in
FusionPBX.
Create an extension for the FAX machine. You can optionally set bypass media to true under advanced in the
extension settings.
FAX Default Settings
Goto Menu -> Advanced -> Default Settings then category Fax
Variables are used as defaults for the dialplan for sending and receiving faxes
fax_enable_t38_request=false (Can be true or false)

ignore_early_media=true (Can be true or false)
Some carriers its better for fax_enable_t38_request=true and for some its better for it to be false.
Its best not to make an assumption and to do testing with different settings to get the best results for your
particular carrier.
The variable fax_enable_t38_request=false will send a T38 reinvite when a fax tone is detected. In some cases
the re-invite always fails for some carriers which is why it is default to false.
Troubleshooting Tips
Faxing will fail at times. Fax Server should automatically try different methods for sending. There are different
combinations like;
With T-38 on/off
ECC on/off
Sending a wav file
Send a fax to HP faxback. This will test sending and receiving 1-888-473-2963
Test sending with Faxtoy.net This will display what is faxed on their website. 1-855-330-1239 or 1-213-294-
2943
Follow Me
Define alternate inbound call handling for the following extensions.
Call Forward- (Disabled or Enabled) Input the destination number

On Busy- (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
No Answer- (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
Not Registered- (Disabled or Enabled) If endpoint is not reachable, forward to this destination before going to
voicemail
Follow Me- (Disabled or Enabled)
Destinations- Can set Delay, Timeout and Prompt to accept the call.
Ignore Busy- (Disabled or Enabled)
Do Not Disturbe- (Disabled or Enabled)
This example has both the extension 1301 itself and and external number to call. If you dont put the extension itself
the extension wont ring when in Follow Me. This is due to the flexible nature of FusionPBX where if you didnt want
that extension to ring like if you were out of the office on a business trip.

IVR Menu
Welcome to the adding IVR section. Here you will find how to add IVRs. Click here for the youtube video
Click on Apps then IVR Menu
Then click the
on the right.
Options in bold are mandatory.

Name: Enter a name for the IVR menu
Extension: Enter the extension number (This must a new extension that isnt allready created)
Greet Long: The long greeting when entering the menu.
Greet Short: The short greeting is played when returning to the menu.
Options: Define caller options for the IVR menu.
Timeout: The number of milliseconds to wait after playing the greeting or the confirm macro.
Exit Action: Select the exit action to be performed if the ivr exists.

Direct Dial: Define whether the callers can dial directly to registered extensions.
Ring Back: Defines what the caller will hear while the destination is being called.
Caller ID Name Prefix: Set a prefix on the caller ID name.
Enabled: set the status of the IVR Menu.
You can get very creative with IVRs and are almost limitless in possibilities. In the basic example below we;
Name the IVR IVR Main
Extension 200
Greet Long a phrase that was made from the phrase section under apps
Number entry in options ,choose an extension for Destination and descriptions ie sales, billing, tech support,
and after hours. timeout 3000 milliseconds
Exit Action to the extension 109 (after hours)

Direct Dial to False and Ring back to Default.
You now have a list of IVRs to go back to and edit or delete as needed.

Phrases
Create phrases of audio files to be played in sequence.
Click the plus on the right to create a phrase and the pencil icon to edit a phrase
Music on Hold
Music on hold can be in WAV or MP3 format. To play an MP3 file you must have mod_shout enabled on the Modules
tab. You can adjust the volume of the MP3 audio from the Settings tab. For best performance upload 16 bit,
8/16/32/48 kHz mono WAV files.
Click the edit pencil on the right to customize music on hold options. This can be done on each kHz group.
Provision
Manual
Cisco
Polycom
Grandstream
Yealink
Zoiper
Automatic
Polycom
Grandstream
Yealink
Zoiper
Phone Screen Capture
Screen Capture
Queues
Queues are used to setup waiting lines for callers. Also known as FIFO Queues.

Recordings
Dial *732 to create a recording, or (for best results) upload a 16bit 8khz/16khz mono WAV file. Click here for the
youtube video.
To view and set the pin number goto Dialplan > Dialplan Manager > Click on Recordings > pin_number=8675309 at
the bottom.
Create a Recording
1. Dial *732 and wait for the voice prompt

2. Enter the password (pin_number) followed by the pound sign# Enter at least a 3 digit number. This will label
the recording file. (recording100.wav)
3. start talking to make the recording after the voice prompt and press the pound key #
4. Press 1 to accept the recording then hang up or press 2 to start over.

Applying Recordings
Once you have a recording made you can use the recordings in different areas of FusionPBX. Custom IVRs and
phrases would be the typical uses.
Ring Group
A ring group is a set of destinations that can be called with a ring strategy.
To add a ring group click the plus. Click for the youtube video .
Name Simply the meaningful name of the Ring group (shows after the Extension in menu selections).
Extension The Dial-able extension for this group standard config states as a 2-7 number extension.
Strategy The selectable way in which the destinations are being used.
Simultaneous Rings all defined Destinations.
Sequence Where order that is lower goes first.

Enterprise Works with follow me.
Rollover calls destinations in sequence and skips busy destinations.
Random A random destination will ring.
Destinations The extensions that this ring group applies to.
Prompt Where you determine if the call must have a dial to confirm before a pickup event.
CID Name Prefix The string that is added to the caller ID when it displays on the ringing extension.
CID Number Prefix The Number that is added to the caller ID when it displays on the ringing extension.
Ring Back What the caller hears when they are waiting for the Destinations to answer.
Context The grouping that this ring group will search as specified in the configuration of your Extensions (if
this excludes an extension it will not ring)

Ring Group Example
In our example we will have 4 extensions all ring at the same time until one of them pick up first. Click the + to create
a ring group. Fill in the fields that are in bold. In the Extension box type a number that is NOT allready created.
This new extention wont be in the extension list. The strategy will be Simultaneous. Enter in the destination the 4
extensions 1001, 1002, 1003, 1004.
Time Conditions
Dynamically route calls to an IVR menu, external numbers, scripts, or other destinations based on time conditions.
Fields in bold are mandatory.
Name Name of the Time Condition.
Extension Define an extension number that is NOT allready created.
Presets US Holiday presets.
Alternate Destination If the condition doesnt match the call will goto the defined alternate destination.
Order Changes the order of which condition is evaluated first.
Enabled If the ring group is enabled.

Time Conditions Example
In our example we have an employee that will receive calls during a set time range and set days. Below is what the
settings look like for Monday through Friday at 5:00pm to 11:00pm. If the employee doesnt answer the call will be
directed to the Timeout Destination. Label the Name as Oncall and invent the Extension as 10011. In the Settings
choose from the dropdown lists for Day of Week for the condition, Monday for the Value and Friday for the Range.
Next set of dropdown list choose Time of Day for the condition, 5:00 PM for the value and 11:00 PM for the Range.
If other options are needed just click the + to the right of Range.
The next dropdown choose the extension where the call is intended for. If the call is outside the date and time specified
the call will goto the Alternate Destination dropdown. Be sure Enabled is set True and click save.
Voicemail
To edit voicemail settings click the pencil edit icon on the right of the extension number.
Here you can edit voicemail settings.

Greeting- When you dial *97, record a greeting and set a number you can choose which greeting to use

Options- Can create a voicemail IVR

Mail to- have voicemails emailed to this address
Voicemail File- Auto File Attachment, Listen Link, Download Link
Keep Local- To keep the voicemail on the server after being emailed or not
Forward Destinations- Forward voicemail messages to additional destinations
Enabled- Enable or disable the voicemail box
Voicemail Options
Access an extensions voicemail away from the extension dial the extension and interupt the greeting with the *star
key.
*97 To access that extensions voicemail from the extension or the voicemail button
*98 To access any extensions voicemail
*99[ext] To access a specific extension voicemail
Main Menu
press 5 For advanced options
Advanced Options
press 1 Record a greeting
press 2 Choose a greeting
press 3 Record name
press 6 Change password
press 0 For main menu
Voicemail Transcription
Uses API services to transcribe voicemails into text to be used in the app-sms and the voicemail to email options.
Currently (11/3/16) only supports microsoft bing
Sign up and language information is located on Microsoft Site
Warning: We cannot use mod_shout to record Voicemails because the transcription service needs an uncom-
pressed version of the audio. Therefore we will record in WAV and then use LAME to re-encode in MP3. This
could cause added resource utilization to your system.
Goto Advanced > Default Settings. Add the following entries

Category Subcategory Type Value Enabled
voicemail transcribe_provider text microsoft True
voicemail microsoft_key1 text {your microsoft key #1} True
voicemail microsoft_key2 text {your microsoft key #2} True
voicemail transcribe_language text en-US True
voicemail transcribe_enabled boolean true True
Click Reload at the top of the page.
Goto Status > Sip Status.
Click Flush Memcache, Reload XML and Rescan.
If you entered your keys correctly, you should now start getting transcriptions delivered in your voicemail to email
and you will also see them on the Messages page.
Status
In the Status menu you have the options for Active Call Center, Active Calls, Active Conferences, Active Queues,
Agent Status, CDR Statistics, Emails, Extension Summary, Log Viewer, Registrations, Services, SIP Status, System
Status and Traffic Graph.

Status
Active Call Center
Select a Call Center Queue from the list below to view its activity.
Active Calls
Use this to monitor and interact with the active calls.
Active Conferences
List all the conferences that are currently active with one or more members.
Active Queues
List all the queues that are currently active with one or more callers.
Agent Status
List all the call center agents with the option to change the status of one or more agents.
4.6. Status 65
CDR Statistics
Call Detail Records Statics summarize the call information.
Emails
Manage failed email messages.
Extension Summary
Extenson number, Number Alias, Missed, No Answer, Busy, ALOC, Inbound Calls, Inbound Duration, Outbound
Calls, Outbound Duration and Description.
Log Viewer
View recent PBX activity and option to download the logs.
Registrations
View the devices that are registered. This will show User, Agent, IP, Port Number, Hostname and Status. You can also
UNREGISTER, PROVISION and REBOOT supported devices from here.
Services
Shows a list of processes, the status of the process and provides control to start and stop the process.
SIP Status
This will show sofia status of internal, internal-ipv6, external, and external-ipv6 profiles. With profiles you can
see REGISTRATIONS, and START/RESTART/RESCAN/FLUSH REGISTRATIONS. You can also FLUSH MEM-
CACHE, RELOAD ACL, RELOAD XML and REFRESH. View UP time, sessions since startup, max sessions, and
current stack size/max.
System Status
System Information, FusionPBX Version, Git Version, Switch Version, Memory Information, CPU Information, Hard
Drive Information and Memcache Information.
Traffic Graph
A browser (or plugin) that supports Scalable Vector Graphics (SVG) is required to view the traffic graph.

Advanced
In the Advanced menu you will find Access Controls, Adminer, App Manager, Backup, Command, Databases,
Default Settings, Domain, Grammer Editor, Group Manager, Menu Manager, Modules, Notifications, PHP Editor,
Provision Editor, Script Editor, Settings, SIP Profiles, SQL Query, Upgrade, Variables and XML Editor.
Welcome to the adding a domain section. Here you will find how to add a domain so that you can reach the specific
tenant from the multi-tenant domain side menu to configure and allow secure administration from the world wide
web.
Adminer
Adminer provides a way to access FusionPBX database.
4.7. Advanced 67
To enable auto login goto Advanced > Default settings and change False to True
To access Adminer goto advanced > adminer.
Access Controls
Access control list can allow or deny ranges of IP addresses. There are several purposes for using the ACL.
The main purpose is for your carriers ip addresses.
Be careful with what and how you use ACL.
Most common mistakes result in calls not working between extensions and other undesirable results.
Be sure to keep Domains access control to default deny.
Do not put your public ip or phone IP addresses in the domains access control list.
Dont supply both the domain and the cidr on the same node.
If adding a single IP address to the CIDR field make sure to add /32 on the end of the IP address.
Access Control Example
Goto Advanced > Access Controls. Click the edit icon for domains. At the bottom under nodes click the plus icon.
Type choose allow

CIDR enter the 123.456.789.000/32
Domain (Leave Blank, used for advanced scenarios)
Description (Carrier Name)
Click save
Goto > Status > Sip Status and click reloadacl.
Under Status > log viewer you should notice the ip added. This can be seen also from command line fs_cli by using
reloadacl
[NOTICE] switch_utils.c:545 Adding 123.456.789.000/32 (allow) [] to list domains
Command
Provides a conventient way to execute system, PHP, switch and SQL commands.

Click the drop down box on the right to choose from Switch, PHP, Shell and SQL to execute commands.
Domains
Welcome to the adding a domain section. Here you will find how to add a domain so that you can reach the specific
tenant from the multi-tenant domain side menu to configure and allow secure administration from the world wide web.
Click here for the youtube video
Adding a domain
Control the list of domains to manage.

There are several reasons to create a domain (tenant). One reason would be to organize customers and so customers
have a unique login ie superadmin@domain.tld or superadmin@subdomain.domain.tld as the username.
In this example we will create a domain.
Goto advanced then click Domains.
4.7. Advanced 69
Then click the
on the right.
This will bring you to enter domain info. (Be sure to create an A record from your domain hosting account)
Click save once entry is complete.

Domain Selection
Changing to a different domain click the stack of three dashes on the top right
A menu will pop open on the right of the screen. Click on the domain that you want to manage. You will always see
the domain you are in by looking at the top right beside the three stacked dashes.
Group Manager
Permit access levels to different group of users. The group permissions allow customizing permissions for existing
groups or custom groups.
superadmin- the global administrator
4.7. Advanced 71
admin- the domain administrator

users- the group for regular users
User Manager
Create, edit, remove users.

Goto Advanced > Group Manager and click USERS at the top right to create, edit or remove a user.
Click the plus at the right to add a user or pencil to edit an existing user.
Fill in the boxes with pertinent information.

Group- assign the user to a group. Be wise as to who has access to what.
Sip Profiles
Advanced -> SIP Profiles

Internal
Internal sip profiles (port 5060/5061) require registration or access controls cidr range to allow the IP address in
without SIP authentication. Once the access controls are setup correctly, the carrier will be allowed to send calls to the
internal profile.
External
External sip profiles (port 5080-5081) allow anonymous connection to FusionPBX and is optional. External profile is
optional when freewitch has a public ip address. Can be useful when setting behind nat. Being anonymous doesnt
mean totally open due to the inbound routes call conditions.(call filtering)
Internal ipv6
Internal ipv6 sip profiles (port 5060/5061) require registration or access controls cidr range to allow the IP address in
without SIP authentication. Once the access controls are setup correctly, the carrier will be allowed to send calls to the
internal ipv6 profile.
If you dont have ipv6 then the ipv6 profiles should be disabled.
Be sure to stop the profile before disabling it. To disable goto Advanced > SIP Profiles and click the pencil edit
icon to the right of the profile you want to disable. From the dropdown box select enabled to false.
External ipv6
External ipv6 sip profiles (port 5080-5081) allow anonymous connection to FusionPBX and is optional.
If you dont have ipv6 then the ipv6 profiles should be disabled.
Be sure to stop the profile before disabling it. To disable goto Advanced > SIP Profiles and click the pencil edit
icon to the right of the profile you want to disable. From the dropdown box select enabled to false.
Upgrade
The FusionPBX code is constantly evolving. Bug fixes being submitted, additions to improve security, making Fu-
sionPBX look nicer, to be more flexible, more scalable, and new features. A complete summary of the changes can be
found on the github code page https://github.com/fusionpbx/fusionpbx/commits/master.
Go to the menu then click on Advanced and then Upgrade. This tool allows you to update the source code, update the
database structure, restore the default menu and permissions. Click here for the Youtube video.
4.7. Advanced 73
Update the source from command line
* cd /var/www/fusionpbx
git pull
chown -R www-data:www-data *
Back to the GUI
*Upgrade Database with advanced -> upgrade schema

*Update permissions
*Update the menu
*Logout and back in
How to Upgrade

To upgrade you will need to get the latest source code. Depending on how extreme the changes have been or the
version you currently are on since your last update, you may need to follow version specific upgrade instructions to
bring your install up to date.
Step 1: Update FusionPBX Source
1. GUI -> Advanced -> Upgrade (doesnt update all files)
Used to update FusionPBX to the latest release.

Upgrade the code via Github/GIT
Login into the web interface with a user account assigned to the superadmin group.
Login to the console with either the ssh, the locally.
Backup Its a good idea to make a backup. If using sqlite, your backup will easily include the SQL database.
mkdir /etc/fusionpbx
mv /var/www/fusionpbx/resources/config.php /etc/fusionpbx
mv /usr/local/freeswitch/scripts/resources/config.lua /etc/fusionpbx
cd /var/www
cp -R fusionpbx fusionpbx_backup
Change the directory''' to the FusionPBX directory
cd /var/www/fusionpbx
Update the source code (example assumes fusionpbx is in /var/www/fusionpbx)
git pull
Permissions
Reset the permissions on the fusionpbx directory tree. When you do git pull it sets the permissions on any updated
files to match the account that you are running git pull with. If that account is different to the web server account it
will result in some files no longer being accessible and a red bar error at the top of the upgrade screen on the GUI. To
fix this you should reapply the permissions in fusionpbx and recursively in all directories inside it.
The example assumes the web server runs as user www-data and fusionpbx is installed to /var/www/fusionpbx.
(chown -Rv Ownername:GroupName /var/www/fusionpbx)
chown -R www-data:www-data *
Step 2: Update Freeswitch Scripts
NOTE: As of FusionPBX 3.8.3 (Stable Branch), the scripts should be automatically updated when updating the
Source Code, using the Advanced > Upgrade page. Any customized scripts, having the same name as the default
scripts, will be overwritten. (An option to disable this default behavior is available using Default Setting: switch >
4.7. Advanced 75
scripts_update > false) Missing scripts will be restored, and any additional files within the scripts folder will remain
untouched.
FusionPBX is a fast moving project where features are constantly being added and bugs are being fixed on a daily
basis so I would also suggest upgrading the Freeswitch scripts directory as part of any normal upgrade process.
Update Freeswitch
Use github to get the updated files. You have to do this from an empty directory.
cp -R /usr/local/freeswitch/scripts /usr/local/freeswitch/scripts-bak
rm -Rf /usr/local/freeswitch/scripts/
cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx.git
cp -R /var/www/fusionpbx/resources/install/scripts /usr/local/freeswitch
chown -R www-data:www-data /usr/local/freeswitch/scripts
cp -R /usr/local/freeswitch/scripts-bak/resources/config.lua /usr/local/freeswitch/
scripts/resources/config.lua
(The last step above is not required if your config.lua file is being stored in a different location, such as the
/etc/fusionpbx folder.)
Clean out this scripts directory

An alternative is to remove the Lua scripts. Only do this if you havent customized any LUA scripts
cp -R /usr/local/freeswitch/scripts /usr/local/freeswitch/scripts-bak
rm -rf /usr/local/freeswitch/scripts/*
Pull the most recent scripts down
Here you need to go directly to step 3 and make sure you run upgrade schema from the GUI immediately otherwise
your calls will not complete.
Restore the config.lua file (IMPORTANT!!)
If your config.lua file was located in scripts/resources/, then youll need to restore it (from the backup previously
performed) to scripts/resources/config.lua.
Step 3: Upgrade Schema
Many updates have changes to the database and to the Freeswitch scripts. The upgrade_schema script

Upgrade from the GUI
From the GUI, run Advanced -> Upgrade Schema which will add any needed newer tables or columns.
Then run App Defaults. If you removed the scripts on Step 2 then run this twice.
Upgrade from the Command Line

An alternative to running upgrade_schema.php from the GUI is to run the upgrade.php from the command line. It
was designed to make the upgrade easier. If you did not login when updating the FusionPBX source code then you
will need to run the upgrade.php file from the command line. Make sure to use the full path to the PHP file.
As root run the following
/usr/bin/php /var/www/fusionpbx/core/upgrade/upgrade.php
4.7. Advanced 77
If your screen was nicely formatted with a fusionpbx theme, and suddenly now goes to a black and white screen with
familiar text but no theme, it is because you were using a theme which no longer exists in the latest version of the
code. If this happens to you navigate to:
http://domain_or_ip/mod/users/usersupdate.php
Then scroll down to where it says Template and select one of the valid templates from the drop down list. Then
press Save. It will be fixed now and you can continue with the remaining steps below.
(Note that any users who have invalid templates selected will also have the same problem you did. You can fix them
from the user manager option in the accounts menu)
Step 4: Apply permissions and Restart Freeswitch
Make sure that the freeswitch directory has the correct permissions
chown -Rv www-data:www-data /usr/local/freeswitch/
Restart Freeswitch
systemctl restart freeswitch
Step 5: Menu
Needed if your menu disappeared.

v1 and v2
Now update the menu to the latest version.
http://domain_or_ip/core/menu/menu_restore_default.php
Press Restore Default on the top right.

v3
https://your.ip/core/menu/menu.php
click e next to the default menu
click the restore default button.
https://your.ip/logout.php
https://your.ip/login.php

Step 6: Re-generate Settings
Sometimes variable names changes. In rev 1877 v_config_cli.php variable names changed which caused no fax to
email emails or voicemail emails to be sent. Problem was the SMTP details did not exist.
Go to Advanced -> Settings and then click save. This will re-generate v_config_cli.php and any other needs config
files.
Version Upgrade
Version Upgrade can take several steps to perform. Below will show how to upgrade from specific versions.
Version 4.0 to 4.2
1. Update the source code. From the web interface go to the Menu -> Advanced > Upgrade page. Check the source
box and the press execute. If you see a red bar it indicates there was a git conflict and you will need to update from
console instead. If you dont see the source box then you will need to update from the console.
git stash
git pull
chown -R www-data:www-data /var/www/fusionpbx
2. If the page goes blank type in the url http://domain.com/logout.php This should bring you back to the login
screen.
3. Udate the Schema. Advanced -> Upgrade Check the Schema box and then then press execute. https://domain.com/
core/upgrade/index.php
4. Check the box for App Defaults and run execute.
4.7. Advanced 79
5. Check the box for Menu Defaults and run execute. This will update the menu to the default menu. The menu
should now look like this.
6. Check the box for Permission Defaults and run execute. Permissions are store in a session to get new permissions
logout and back in.
7. Goto Dialplan > Dialplan Manager and delete local_extension. Then goto Advanced > Upgrade and only
check box App Defaults and click execute. This will regenerate the new local_extension version.
8. Go to Applications > Conference profiles. Edit each profile and replace $${hold_music} with lo-
cal_stream://default
9. Goto Advanced > Variables hold_music. Make sure its value is set as local_stream://default
Check Applications > Music On Hold to see if music is listed properly.

You should see in red default for the category and the kHz sub categories should be
in blue.
If not, do the following
* Edit (Pencil icon on the right) the Category names to reflect default for 8, 16, 32,
and 48kHz.
* After you click the pencil icon choose at the bottom the domain for the rates and
click save.
* If the category is blank, you may have missed running Advanced > check box app
defaults > execute or you may not have renamed autoload_configs/local_stream.conf.
xml file to local_stream.conf.
* For custom music on hold check the path for the domain name and set select for the
domain name to match the domain used in the path.
10. Remove .xml from the end of the following file names

**Before**
autoload_configs/callcenter.conf.xml
autoload_configs/conference.conf.xml
autoload_configs/local_stream.conf.xml
**After**
autoload_configs/callcenter.conf
autoload_configs/conference.conf
autoload_configs/local_stream.conf
11. Edit autoload_configs/lua.conf.xml adding languages. Restart of FreeSWITCH is required.
<param name="xml-handler-bindings" value="configuration,dialplan,directory,languages"/

>
12. Update Time Conditions (Bug Fix)

Goto Advanced > Upgrades page. Check box Update Source, execute. Goto Advanced > Default settings
> Category > delete the category: time condition presets. Goto Advanced > Upgrade > check box App
Defaults, execute. Goto Advanced > Default settings. Click Reload at the top right. (This will get the
new presets)
Next steps are for existing Time Conditions Goto Apps > Time Conditions and edit the time conditions
remove all holidays and hit save. Select the holidays over again.
Many of the provisioning templates were updated. If you use custom provisioning templates you should
consider updating them with the new versions.
Version 3.8 to 4.0
4.7. Advanced 81
Remove the comments from the script-directory in /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

If using the FreesWITCH package then remove $${base_dir} and set the full path to the scripts directory.
before: 
after: <param name="script-directory" value="/usr/local/freeswitch/scripts/?.lua"/>
Rebooting FreeSWITCH is required for this to take effect.
Version 3.6 to 3.8
Note: Upgrading can get very complex. If the production system is critical or you are intimidated from these
upgrade instructions you may want FusionPBX paid support at http://www.fusionpbx.com/support.php
A standard upgrade procedure should always be followed:

(1. Make a Backup!, 2. Advanced > Upgrade steps, 3. Update switch scripts, 4. Restart FreeSWITCH).
Beyond the standard upgrade procedure just described, the following will also need to be performed:
uncomment: <param name="script-directory" value="$${base_dir}/scripts/?.lua"/>

in: /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml
* Rebuild all time conditions.

* After you edit a particular time condition, click the Dialplan button on the top right to see what was there originally.
* Delete the following dialplans from each domain then run Advanced -> Upgrade -> App Defaults. If using XML
handler for the dialplan flush memcache. If using dialplans XML on the file system resave one of the dialplans to
have FusionPBX rewrite the XML files.
* user_exists - call_timeout variable was added
* extension-intercom - It has been renamed to page-extension
* eavesdrop - Change *88[ext] to *33[ext] so that it doesnt conflict with page-extension at *8[ext]
* user_status - Has been renamed to agent_status
* page - Dialplan has been simplified.
* valet_park_out - Changed regex variable from $1 to $2
* local_extension - failure handler was added to support call forward on busy and no answer
* If using call center feature code *22 edit each agent and add an agent id and password (pin number)
* Delete any dialplan with the features context. These have been moved into the dialplan domain contexts.

* If using App -> XMPP, Content Manager, or Schema they have been moved dev -> branches -> apps directory need
to pull files from there if you want to use any of them.
* For single tenant systems default context is no longer used by default.
* Easiest way to update your system is go to Advanced -> Domains and edit your domain.
* Copy your current domain name then change the name to default then save the change.
* Now edit the domain name again and paste your original domain name or IP address whatever the domain
originally was and save the changes
* Go to accounts extensions and save one extension. (not needed if using the XML handler)
* Go to Dialplan Manager and save one of the dialplans. (not needed if using the XML handler)
* FAX ( may require adjusting the paths and web server user account to match your server www-data is used in this
example)
* Delete all previous FAX dialplans
* Resave each fax server in the GUI.
* cd /var/www/fusionpbx/app/fax
* wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/fax_import.php
* chown -R www-data:www-data fax_import.php
* Login into the GUI and use this path in your browser http://<domain-or-ip>/app/fax/fax_import.php
* rm /var/www/fusionpbx/app/fax/fax_import.php
* Groups and Permissions
If you go to Advanced Group Manager -> And you see what looks like duplicates of user, admin and superadmin
groups then you need do the following instructions.
Remove permissions associated with all domain groups with names that match default global groups...
Use the Advanced -> SQL Query tool to do the following.
delete from v_group_permissions where domain_uuid is not null

and (
group_name = 'user'
or group_name = 'admin'
or group_name = 'superadmin'
or group_name = 'agent'
or group_name = 'public'
)
Remove all domain groups having the same names as the default global groups
(retains any custom domain groups)...
delete from v_groups where

domain_uuid is not null
and (
group_name = 'user'
)
4.7. Advanced 83
Empty the group_uuid field for any group user with a group_name value having
the same name as the default global groups (retains user assignments to custom domain
groups)...
update v_group_users set group_uuid = null where

group_name = 'user'
For group users with a null group_uuid, insert the group_uuid of the global group that matches the group_name
value...
Run this code from Advanced -> Command -> PHP Command.
$sql = "select group_user_uuid, group_name ";

$sql .= "from v_group_users where group_uuid is null";
$prep_statement = $db->prepare(check_sql($sql));
$prep_statement->execute();
$result = $prep_statement->fetchAll(PDO::FETCH_NAMED);
$result_count = count($result);
unset($prep_statement);
if ($result_count > 0) {
foreach($result as $field) {
//note group user uuid
$group_user_uuid = $field['group_user_uuid'];
$group_name = $field['group_name'];
//get global group uuid
$sql = "select group_uuid from v_groups ";
$sql .= "where domain_uuid is null ";
$sql .= "and group_name = '".$group_name."' ";
$prep_statement = $db->prepare($sql);
$prep_statement->execute();
$sub_result = $prep_statement->fetch(PDO::FETCH_ASSOC);
$sub_result_count = count($sub_result);
unset ($prep_statement);
//set group uuid
if ($sub_result_count > 0) {
$sql = "update v_group_users ";
$sql .= "set group_uuid = '".$sub_result['group_uuid']."' ";
$sql .= "where group_user_uuid = '".$group_user_uuid."' ";
$count = $db->exec(check_sql($sql));
unset($sql);
}
}
}
Apps menu disappeared

If your apps menu disappeared check that it wasnt set to protected in the menu manager.
(advanced -> menu manager). If protected is true, it wont show up.
Version 3.5 to 3.6
When running Upgrade -> Schema

If you see ALTER TABLE v_xml_cdr ADD json json; every time you run the upgrade schema then you likely have
an old version of Postgres. To fix this either upgrade to the latest Postgres server or run the following SQL statement
from advanced -> sql query.
ALTER TABLE v_xml_cdr ADD json text;
See https://github.com/fusionpbx/fusionpbx/issues/655 for more details.
Potential issue with call recording after upgrading/switch to latest 3.6 stable.
After upgrading to 3.6 stable from 3.5 dev I noticed that calls were no longer being recorded. This was due to the file
extension being missing from the recording path. If this is happening to you it is an easy fix.
Go to Advanced -> variables -> category default and add the variable record_ext and set it to either wav or mp3.
Choosing mp3 depends upon whether or not you have mod_shout installed and enabled.
Version 3.4 to 3.5
Gateways now use the gateway_uuid as the name that is used when interacting with FreeSWITCH. This script is
needed to help change the gateway names used in the outbound routes. You may need to remove the old gateway file
names from the conf/sip_profiles/external directory.
wget http://fusionpbx.googlecode.com/svn/branches/dev/scripts/upgrade/gateway_uuid.php
http://x.x.x.x/gateway_uuid.php
rm gateway_uuid.php
* Go To Advanced -> Default Settings -> Switch Category -> Sub category gateways change to sip_profiles
Permissions Issues - (access denied errors)

Due to changes which improve consistency throughout the product, some Users have had problem with superadmin
receiving access denied errors after the upgrade.
4.7. Advanced 85
* Go To Advanced -> Group Manager

* On superadmin click Permissions and then Restore Default
You may need to execute this operation for each group.
Default Settings
In the switch category change gateways to sip_profiles
Version 3.3 to 3.4
Update the source as described on this page, menu manager restore default, group manager edit a group restore
default, advanced -> upgrade schema.
FusionPBX 3.4 hunt groups have been deprecated. Use the following script run it only one time to move existing
hunt groups to ring groups.
wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/hunt_group_
export.php
http://x.x.x.x/hunt_group_export.php
rm -r hunt_group_export.php
Ring groups were expanded to add ability to call external numbers and match other missing hunt group features. A
new table was created to accomodate this.

wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/ring_group_
extensions.php
http://x.x.x.x/ring_group_extensions.php
rm ring_group_extensions.php
Version 3.2 to 3.3
FreeSWITCH changed the syntax to connect to the database so numerous LUA scripts had to be updated. If you
customized any of the lua scripts make a backup of the FreeSWITCH scripts directory. Then remove the contents of
the freeswitch/scripts directory and then run advanced -> upgrade schema (which will detect the missing scripts
and replace them).
Version 3.1.4 to 3.2
Ubuntu/Debian
git pull
Advanced -> Upgrade Schema
Menu
If you cant see the menu after upgrading try the following in your browser replace x.x.x.x with your ip or domain
name.
x.x.x.x/core/menu/menu.php
Edit the menu make sure the language is set to en-us.
Press **Restore Default**
Default settings
x.x.x.x/core/default_settings/default_settings.php
category: language
type: code
value: en-us
4.7. Advanced 87
Email
Migrating email to the new FusionPBX native voicemail.
wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/voicemail_
export.php
Run from the browser it will take the voicemail data from the FreeSWITCH database and copy the information into
the FusionPBX database.
http://x.x.x.x/voicemail_export.php
Remove the export file
rm voicemail_export.php
Call Forward / Follow Me
No longer using hunt groups. So the backend has changed so keep in mind that you need to reset call forward and
follow me settings. They are still listed in app -> hunt groups. After updating the info in call forward, follow me
you should delete the hunt group.
Version 2 to 3.0
LESS than or EQUAL to revision 1877, use the migration tool.

https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade
If greater than revision 1877, use latest.
| When upgrading from previous versions, you may encounter the following issues:
Changes to your dial plan or extensions dont take effect

* Go to the Advanced -> Default Settings page
* Remove /default from the end of your dialplan and extensions directories
Missing menus
* Go to hxxps://yourdomain.com/core/menu/menu.php
* Click the edit (e) button beside default

* Click the Restore Default button

* Check that all the entries in the list are accessible by the appropriate groups
Emails not being sent for voicemail or fax

* Double check the SMTP settings on the System -> Settings page
* Save it, even if you havent changed anything
Release Revisions
r0001 is 1.0 release - 6 Nov 2009
r2523 is 3.0 release - 3 May 2012
r2585 is 3.0.4 release - 24 May 2012
r2757 is 3.1 release - 18 Aug 2012
r2777 is 3.1.1 release - 26 Aug 2012
r2827 is 3.1.2 release - 12 Sep 2012
r2897 is 3.1.3 release - 26 Sep 2012
r2907 is 3.1.4 release - 27 Sep 2012
r3694 is 3.2 release - 19 Jan 2013
r3978 is 3.3 release - 1 May 2013
r4605 is 3.4 release - 28 Sep 2013
r6747 is 3.6.1 release - 22 Aug 2014
r8481 is 3.8.3 release - 11 May 2014
r793d386 is 4.0 release - Aug 2015
r4fdb6e9 is 4.1 release - Dec 2015
rxxxxxxx is 4.2 release - xxx 2016
SQLite
SQLite is the FreeSWITCH default. Databases are located in the freeswitch/db directory.
ODBC
http://wiki.freeswitch.org/wiki/ODBC
Postgres
Postgres native support will be in FreeSWITCH 1.2.4 but has been available in the Main GIT branch.
4.7. Advanced 89
Dependencies
libpq and the associated dev packages are required
Configure
To enable PostgresSQL as a native client in FreeSWITCH you must enable it during the build when running configure.
** ./configure enable-core-pgsql-support **
switch.conf.xml
Under the Settings area insert the following line

<param name=core-db-dsn value=pgsql;hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch
password= options=-c client_min_messages=NOTICE application_name=freeswitch />
Additional Information
Fail2Ban
For information about Fail2Ban on FreeSWITCH, http://wiki.freeswitch.org/wiki/Fail2ban see their wiki.
Logs
This will log FusionPBX authentication failures to syslog (AUTH_LOG). This file can be in different places
depending on how rsyslog, or syslog is configured.
Ubuntu
/var/log/auth.log
Examples
GUI Login
incorrect username
Feb 1 11:35:11 your_hostname FusionPBX: [w.x.y.z] authentication failed for login_

username
incorrect password
Feb 1 12:07:27 your_hostname FusionPBX: [w.x.y.z] authentication failed for

superadmin

Provisioning
Created from the code in /fusionpbx/mod/provision/index.php Please doublecheck this!
Feb 1 12:07:27 your_hostname FusionPBX: [w.x.y.z] provision attempt bad password for
AA:BB:CC:DD:EE:FF
Setting up Fail2Ban
RegEx
You can test the regex with fail2ban-regex
'[hostname] FusionPBX: \[<HOST>\] authentication failed'
Configuration
Jail Options
Every jail can be customized by tuning the following options:
Name Default Description

filter Camp- Name of the filter to be used by the jail to detect matches. Each single match by a filter
ground increments the counter within the jail
log- /var/log/messages
Path to the log file which is provided to the filter
path
maxretry 3 Number of matches (i.e. value of the counter) which triggers ban action on the IP.
find- 600 sec The counter is set to zero if no match is found within findtime seconds.
time
ban- 600 sec Duration (in seconds) for IP to be banned for.
time
Filter Rules
vim /etc/fail2ban/filter.d/fusionpbx.conf
# Fail2Ban configuration file

#
# Author: soapee01
#
[Definition]
4.8. Additional Information 91

# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named "host". The tag "<HOST>" can
# be used for standard IP/hostname matching and is only an alias for
# (?:::f{4,6}:)?(?P<host>[\w\-.^_]+)
# Values: TEXT
#
#failregex = [hostname] FusionPBX: \[<HOST>\] authentication failed
#[hostname] variable doesn't seem to work in every case. Do this instead:
failregex = .* FusionPBX: \[<HOST>\] authentication failed for
= .* FusionPBX: \[<HOST>\] provision attempt bad password for
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
add the following to /etc/fail2ban/jail.local
[fusionpbx]
enabled = true
port = 80,443
protocol = tcp
filter = fusionpbx
logpath = /var/log/auth.log
action = iptables-allports[name=fusionpbx, protocol=all]
# sendmail-whois[name=FusionPBX, dest=root, sender=fail2ban@example.org] #no
smtp server installed
Add /etc/fail2ban/filter.d/freeswitch.conf with the contents:
# Fail2Ban configuration file

#
# Author: Rupa SChomaker (first two regex)
[Definition]
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named "host". The tag "<HOST>" can
# be used for standard IP/hostname matching and is only an alias for
# (?:::f{4,6}:)?(?P<host>[\w\-.^_]+)
# Values: TEXT
#
failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure $REGISTER$ on sofia
profile \'\w+\' for \[.*\] from ip <HOST>
\[WARNING\] sofia_reg.c:\d+ SIP auth failure $INVITE$ on sofia profile \

'\w+\' for \[.*\] from ip <HOST>
\[WARNING\] sofia_reg.c:\d+ SIP auth challenge $REGISTER$ on sofia

profile \'\w+\' for \[.*\] from ip <HOST>

# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
Modify /etc/fail2ban/jail.conf. Add the following make sure the freeswitch.log file path is correct.
[freeswitch-tcp]
enabled = true
port = 5060,5061,5080,5081
protocol = tcp
filter = freeswitch
logpath = /usr/local/freeswitch/log/freeswitch.log
action = iptables-allports[name=freeswitch-tcp, protocol=all]
sendmail-whois[name=FreeSwitch, dest=root, sender=fail2ban@example.org]
[freeswitch-udp]
enabled = true
port = 5060,5061,5080,5081
protocol = udp
filter = freeswitch
logpath = /usr/local/freeswitch/log/freeswitch/freeswitch.log
action = iptables-allports[name=freeswitch-udp, protocol=all]
sendmail-whois[name=FreeSwitch, dest=root, sender=fail2ban@example.org]
/var/log/fail2ban.log will log this after 3 missed logins.
2011-02-01 12:32:18,151 fail2ban.actions: WARNING [fusionpbx] Ban 192.168.100.1
hostname # iptables -n -L fail2ban-fusionpbx
Chain fail2ban-fusionpbx (1 referecnes)

target prot opt source destination
DROP all -- 192.168.100.1 anywhere
RETURN all -- anywhere anywhere
Important
You can easily ban yourself, including current active ssh connections.
To unban:

hostname # iptables -n -D fail2ban-fusionpbx 1
Keep yourself from getting banned.

add to /etc/fail2ban/jail.local
[DEFAULT]
# "ignoreip" can be an IP address, a CIDR mask or a DNS host

ignoreip = 127.0.0.1 192.168.0.99
bantime = 600
maxretry = 3
Errors
If youre seeing something like this in your fail2ban logfile:
2011-02-27 14:11:42,326 fail2ban.actions.action: ERROR iptables -N fail2ban-

freeswitch-tcp
add the time.sleep(0.1) to /usr/bin/fail2ban-client
def __processCmd(self, cmd, showRet = True):

beautifier = Beautifier()
for c in cmd:
'''time.sleep(0.1)'''
beautifier.setInputCmd(c)
or
sed -i -e s,beautifier\.setInputCmd$c$,'time.sleep$0\.1$\n\t\t\tbeautifier.
setInputCmd$c$', /usr/bin/fail2ban-client
http://www.fail2ban.org/wiki/index.php/Fail2ban_talk:
Community_Portal#fail2ban.action.action_ERROR_on_startup
Freeswitch install
Upgrade Move Source

mv /usr/src/freeswitch freeswitch-version
Git Release
cd /usr/src
git clone -b v1.4 https://freeswitch.org/stash/scm/fs/freeswitch.git
cd freeswitch
./bootstrap.sh
or
Git Head
cd /usr/src
git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
cd freeswitch
./bootstrap.sh
or
files.freeswitch.org
cd /usr/src
wget http://files.freeswitch.org/freeswitch-1.4.26.zip
unzip freeswitch-1.4.26.zip
cd freeswitch-1.4.26
1.4.x is considered EOL use the steps below for 1.6.x
cd /usr/src
wget http://files.freeswitch.org/freeswitch-1.6.6.zip
unzip freeswitch-1.6.6.zip
cd freeswitch-1.6.6
Ubuntu Dependencies
apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev

libncurses5-dev libtool make python-dev gawk pkg-config libtiff-dev libperl-dev
libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-
dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev memcached
libmemcached-dev

Debian Dependencies
apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev

libncurses5-dev libtool libtool-bin make python-dev gawk pkg-config libtiff5-dev
libperl-dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-
dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev
memcached libmemcached-dev
CentOS
yum install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-
devel libjpeg-devel openssl-devel e2fsprogs-devel sqlite-devel libcurl-devel pcre-
devel speex-devel ldns-devel libedit-devel libmemcached-devel
Configure services to auto start
chkconfig --add memcached && chkconfig --levels 33 memcached on

chkconfig --add freeswitch && chkconfig --levels 35 freeswitch on
modules.conf
uncomment the FreeSWITCH modules that are needed.
mod_avmd
mod_callcenter
mod_memcache
mod_cidlookup
mod_curl
Used for MP3 support
mod_shout
Postgres driver
./configure --enable-core-pgsql-support
Run Make

make
Remove FreeSWITCH files
This step is only needed for a FreeSWITCH upgrade.

Once it has been confirmed that the compile was successful then remove files from previous version of FreeSWITCH
rm -rf /usr/local/freeswitch/{lib,mod,bin}/*
Install
make install
File Permissions
Set the file permissions instructions may vary based on the OS and install directory.
Debian and Ubuntu
chown -R www-data:www-data /usr/local/freeswitch
CentOS or Other Unix operating systems

(need make sure that the web server has access to IVR recordings, Fax, and Voicemail)
adduser --disabled-password --quiet --system --home /usr/local/freeswitch --gecos

"FreeSWITCH Voice Platform" --ingroup daemon freeswitch
chown -R freeswitch:daemon /usr/local/freeswitch/

chmod -R o-rwx /usr/local/freeswitch/
Install Sound Files
Run this on new installs.

cd /usr/src/freeswitch
make sounds-install moh-install
make hd-sounds-install hd-moh-install
make cd-sounds-install cd-moh-install
Startup Script
Run on new install only. Create the file /etc/init.d/freeswitch with the following code:
#!/bin/bash
### BEGIN INIT INFO
# Provides: freeswitch
# Required-Start: $local_fs $remote_fs
# Required-Stop: $local_fs $remote_fs
# Default-Start: 2 3 4 5
# Default-Stop: 0 1 6
# Description: Freeswitch debian init script.
# Author: Matthew Williams
#
### END INIT INFO
# Do NOT "set -e"
# PATH should only include /usr/* if it runs after the mountnfs.sh script
PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/local/bin
DESC="Freeswitch"
NAME=freeswitch
DAEMON=/usr/local/freeswitch/bin/$NAME
DAEMON_ARGS="-nc -nonat -reincarnate"
PIDFILE=/usr/local/freeswitch/run/$NAME.pid
SCRIPTNAME=/etc/init.d/$NAME
FS_USER=www-data #freeswitch
FS_GROUP=www-data #daemon
# Exit if the package is not installed

[ -x "$DAEMON" ] || exit 0
# Read configuration variable file if it is present

[ -r /etc/default/$NAME ] && . /etc/default/$NAME
# Load the VERBOSE setting and other rcS variables

. /lib/init/vars.sh
# Define LSB log_* functions.

# Depend on lsb-base (>= 3.0-6) to ensure that this file is present.
. /lib/lsb/init-functions
#
# Function that sets ulimit values for the daemon
#
do_setlimits() {
ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited

ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited
return 0
}
#
# Function that starts the daemon/service
#
do_start()
{
# Set user to run as
if [ $FS_USER ] ; then
DAEMON_ARGS="ècho $DAEMON_ARGS` -u $FS_USER"
fi
# Set group to run as
if [ $FS_GROUP ] ; then
DAEMON_ARGS="ècho $DAEMON_ARGS` -g $FS_GROUP"
fi
# Return
# 0 if daemon has been started
# 1 if daemon was already running
# 2 if daemon could not be started
start-stop-daemon --start --quiet --pidfile $PIDFILE --exec $DAEMON --test > /
dev/null -- \
|| return 1
do_setlimits
start-stop-daemon --start --quiet --pidfile $PIDFILE --exec $DAEMON --
background -- \
$DAEMON_ARGS \
|| return 2
# Add code here, if necessary, that waits for the process to be ready
# to handle requests from services started subsequently which depend
# on this one. As a last resort, sleep for some time.
}
#
# Function that stops the daemon/service
#
do_stop()
{
# Return
# 0 if daemon has been stopped
# 1 if daemon was already stopped
# 2 if daemon could not be stopped
# other if a failure occurred
start-stop-daemon --stop --quiet --retry=TERM/30/KILL/5 --pidfile $PIDFILE --
name $NAME
RETVAL="$?"
[ "$RETVAL" = 2 ] && return 2
# Wait for children to finish too if this is a daemon that forks
# and if the daemon is only ever run from this initscript.
# If the above conditions are not satisfied then add some other code

# that waits for the process to drop all resources that could be
# needed by services started subsequently. A last resort is to
# sleep for some time.
start-stop-daemon --stop --quiet --oknodo --retry=0/30/KILL/5 --exec $DAEMON
[ "$?" = 2 ] && return 2
# Many daemons don't delete their pidfiles when they exit.
rm -f $PIDFILE
return "$RETVAL"
}
#
# Function that sends a SIGHUP to the daemon/service
#
do_reload() {
#
# If the daemon can reload its configuration without
# restarting (for example, when it is sent a SIGHUP),
# then implement that here.
#
start-stop-daemon --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME
return 0
}
case "$1" in
start)
[ "$VERBOSE" != no ] && log_daemon_msg "Starting $DESC" "$NAME"
do_start
case "$?" in
0|1) [ "$VERBOSE" != no ] && log_end_msg 0 ;;
2) [ "$VERBOSE" != no ] && log_end_msg 1 ;;
esac
;;
stop)
[ "$VERBOSE" != no ] && log_daemon_msg "Stopping $DESC" "$NAME"
do_stop
case "$?" in
0|1) [ "$VERBOSE" != no ] && log_end_msg 0 ;;
2) [ "$VERBOSE" != no ] && log_end_msg 1 ;;
esac
;;
status)
status_of_proc -p $PIDFILE $DAEMON $NAME && exit 0 || exit $?
;;
#reload|force-reload)
#
# If do_reload() is not implemented then leave this commented out
# and leave 'force-reload' as an alias for 'restart'.
#
#log_daemon_msg "Reloading $DESC" "$NAME"
#do_reload
#log_end_msg $?
#;;
restart|force-reload)
#
# If the "reload" option is implemented then remove the
# 'force-reload' alias
#
log_daemon_msg "Restarting $DESC" "$NAME"

do_stop
case "$?" in
0|1)
do_start
case "$?" in
0) log_end_msg 0 ;;
1) log_end_msg 1 ;; # Old process is still running
*) log_end_msg 1 ;; # Failed to start
esac
;;
*)
# Failed to stop
log_end_msg 1
;;
esac
;;
*)
#echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2
echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2
exit 3
;;
esac
exit 0
Make the script executable and make it auto start on system boot:
chmod +x /etc/init.d/freeswitch
update-rc.d freeswitch defaults
Testimonials
Businesses of all sizes use FusionPBX daily. We love to see folks happy saving money using FusionPBX. Here are
some of the testimonials we received.
Just want to give a thankful shout out to everyone at FusionPBX that have helped in education, contribution and
support. The FusionPBX team have developed a leading product. Its been a joy from day one joining this community
and I look forward to the road ahead.
-Kissvoice

We have been using FusionPBX for many of our clients and just want to express our gratitude to Mark and the team for
not only providing a great product, but being extremely helpful in bringing out new features and helping us maintain
the service. Every new release amazes us with the work and development put into it.
-Kloudphone
SureVoIP have been using FusionPBX since 2010. SureVoIP sponsored the first versions of multi-tenant domains and
hot desking. SureVoIP sponsors and contributes fixes and features when possible.
Because of FusionPBXs highly configurable nature, responsive support team and sane design, SureVoIP have been
able to win many large customers because proprietry systems are so rigid and slow to innovate. We have been proud
to support and deploy FusionPBX for 7 years.
-SureVoIP
Winner of the Best Business ITSP (Medium Enterprise) 2016! http://www.surevoip.co.uk/2016-best-provider
I would like to tell everyone there that I have been trying to get an open source PBX to work for me for over three
months now and since I am not a linux guy, I havent been able to get any of them working the way I wanted.
FusionPBX installation script installed ALL required packages and libraries in one go and it was up and running
in 10 mins. Once I followed the youtube videos <https://www.youtube.com/fusionpbx>_ It took me no time to setup
and migrate my clients to FusionPBX. One of the best features I love in FusionPBX is the automatic dialplan expression
as I have always struggled with remembering the expression syntax. The user interface and the way all the features
are grouped is awesome.
Again, Thanks for the effort
-BareVOIP Limited
We would love to hear from you! Please reach out to us at http://fusionpbx.com/support.php if you would like to be
featured on this page.
Password Reset
The steps below are outdated but useful for older installations. Click here for the new youtube video on password
recovery.
Here some rough steps to change the password of the database. The password can only be changed and not recovered.
The database contains a table called v_users which contains the username, password and salt. The password is the
md5 hash of the password and the salt.

Password Hash
Use the following commands to generate the password hash. Dont forget to provide your own salt and password.
echo '<?php $salt = "random-salt-goes-here";$password = "put your password here";

echo md5($salt.$password)."\n"; ?>' > /tmp /test.php
Run the php file from command line.
php /tmp/test.php
SQLite
Install sqlite3 which can be be used to modify the database fusionpbx.db. Then open the database with the following:
sqlite3 fusionpbx.db
PostgreSQL
Connect to the PostgreSQL database. Once you are running psql you can use:
\l to list the databases.

\c to connect to one of them.
After running the SQL Query then use q to quit.
su postgres
psql
\c fusionpbx
Change the Password
The hashed password and the salt can be updated using the command:
update v_users set password = 'replace-with-password-hash-from-php-script', salt =

'replace-with-your-random-salt' where username = 'superadmin';
Feature Codes
Below are the * codes used with FusionPBX. You can also create more as needed. If you do be sure to pick ones
that are not currently in use.

Basic
Feature Code Name Detail

Call Transfer Transfer a call to another extension
*1
Record Active Call
*2
Attended Call Transfer Attended call transfer to another extension. After extension
*4
number press #
Directory
*411 *DIR to dial by name.
DISA
*3472 *DISA followed by Administrative PIN to receive a dialtone and
call out
Call Privacy Activate call privacy
*67<phone
number>
Call Return Call back the last incoming number
*69
Record
*732 *REC followed by Administrative PIN to record a message
Extension Intercom Page a specific extension.
*8[ext]
Redial Redial a number
*870
Talking Date Current server date
*9171
Talking Time Current server time
*9170
Talking Date & Time Current server data & time
*9172
Wakeup Call Schedule a wakeup call
*925
Enable DND Enable Do Not Disturb
*78
Disable DND Disable Do Not Disturb
*79
FreeSWITCH Connects to Cluecon Weekly
*9888
Conference
Speed Dial Speed dial an extension
*0[ext]
Follow Me Set the Follow Me number
*21
Enable Call Forward Enables Call Forward
*72
Disable Call Forward Disables Call Forward
*73
Call Forward Toggle Call Forward enable/disable
*74
Administrative PIN (Recordings pin) can be found here Administrative PIN page
Call Parking

Valet Park Attended Transfer (park). The park extension will be played back to you
*5900
Valet Un-Park Retrieve a Valet Parked call
*5901-5999

Advanced

Extension Intercom Page a specific extension
*8[ext]
Eavesdrop Listen to the call. Press 1 for remote, 2 for local, 3 for full conversation
*33 <ext>
** <ext> Intercept an Intercept a specific extension
extension
Voicemail
Feature Name Detail

Code
Voicemail The system detects the extension, and will prompt for your password
*97
Check any Voicemail The system will prompt for both your id (extension number) and
*98
box password
Check any Voicemail The system will prompt for both your id (extension number) and
*4000
box password
Send to Voicemail Send a call directly to voicemail
*99<exten-
sion>
Miscellaneous

Info Sends information to the console
*9192
Video Record Record Video
*9193
Video Playback Playback Video
*9194
Delay Echo Audio is played back after a slight delay
*9195
Echo Test Echo Test
*9196
Milliwatt Tone Tone Playback
*9197
Test MoH Test Music on Hold
*9664
*You can also add extra feature codes
Toll Allow
Toll Allow is a variable that can be set per extension. It allows you to limit who can make what type of calls. Note
that although the variable is provided in the extension configuration, the default dialplan DOES NOT make use of it.
Therefore if you want to use it you need to add statements to the dialplan to enable it.
An example for the contents of the toll_allow variable would be:
You can then add information to your dialplan to process this variable. In the example XML below, if the valid allow
value isnt present then an extension shouldnt be able to dial out. However extension -> extension should still work.

The following code are example XML for standard outbound routes (Dialplan->OutboundRoutes). Effectively you are
applying an additional condition to EACH outbound route that you want to limit. So in the FusionPBX GUI select an
outbound route and add
condition, type "${toll_allow}", data "local".

Order is important, this should be the FIRST condition of your outbound route.
Youll need to do that for all of your outbound routes, tag them local, domestic, or interna-
tional depending on what they are. On some installations this example file will be present in
/usr/local/freeswitch/conf/dialplan/default/01_example.com.xml:
PERMIT TOLL CALLS
This example assumes all calls are bad (except internal) unless they are flagged as good by the value of the toll_allow
variable.
<include>
<extension name="local.example.com">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{7})$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_
number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}

"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1${default_

areacode}$1"/>
</condition>
</extension>
<extension name="domestic.example.com">
<condition field="${toll_allow}" expression="domestic"/>
<condition field="destination_number" expression="^(\d{11})$">
number}"/>

"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>

</condition>
</extension>
<extension name="international.example.com">
<condition field="${toll_allow}" expression="international"/>
<condition field="destination_number" expression="^(011\d+)$">
number}"/>

"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>

</condition>
</extension>
</include>

PREVENT TOLL CALLS
This example takes the opposite approach and is how to PREVENT toll calls. The below example takes the opposite
approach. It assumes that all calls are good unless they are flagged as bad.
Put this in your advanced dialplan. In the toll allow of whatever extension you wanted to restrict put the value local.
This example restricts from calling 10 or 11 digit numbers.
<extension name="localcalls" >

<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="(^\d{10}$|^\d{11}$)">
<action application="hangup"/>
</condition>
</extension>
TFTP
Several models of phone out there that still only use TFTP for provisioning. Even though they have reached end of
life, some of the popular ones are the Cisco 7960 and 7940.
Install TFTPD
apt-get install tftpd

service xinetd
Change the configuration
edit the /etc/xinetd.d/tftp
Enable TFTP in FusionPBX Gui

Goto Advanced > Default Settings > Provision
Set Enabled to True and define the path to where the TFTP files will be.

Test TFTP
tftp x.x.x.x
get 000000000000.cnf
See the file getting requested for tftp

tail -f /var/log/syslog | grep tftp
Network Address Translation
NAT is Network Address Translation. When your FusionPBX and/or FreeSWITCH are inside NAT then then you may
experience one way audio or no audio in either direction the following information can help you get audio working in
both directions.
Default config
The external_rtp_ip and external_sip_ip are set to $${local_ip_v4} in Advanced -> Variables by default. The lo-
cal_ip_v4 variable is auto detected by FreeSWITCH. The variable can be also be overidden as a preset variable before
it is used if you want to control the IP address that it represents.
This works good when the server has a public IP address.
It also works well when all phones are inside the same network and nothing needs to traverse the NAT. For
example if you are using a SIP to TDM gateway and all your phones are in the same network.
SIP ALG
A SIP Application Layer Gateway (ALG) is a tool designed to help SIP traverse NAT. While the SIP ALG is good in
theory it often causes more problems than it solves. Because of this its usually best to disable the SIP ALG on your
firewall. An alternative way to disable it is to move SIP to a non standard port.

Static IP
FusionPBX is behind NAT and you have a static public IP address and you have phones on the same network and/or
outside the network.
Set external_rtp_ip to autonat:xxx.xxx.xxx.xxx
Set external_sip_ip to autonat:xxx.xxx.xxx.xxx
If you dont register a gateway to the carrier you may need to port forward SIP and RTP.
UPnP or PMP
FusionPBX is behind NAT and you dont have a static ip address. You do have a firewall that is capable of UPnP or
PMP.
Enable UPnP or PMP in your firewall
In Debian OS /etc/default/freeswitch remove -nonat
Make systemd aware of the changes. systemctl daemon-reload
Set external_rtp_ip to auto-nat
Set external_sip_ip to auto-nat
Restart FreeSWITCH. service freeswitch restart
Contributing
There are many ways to help the FusionPBX project.

What We Need:
1. Developers
2. Technical Writers
3. Translators
4. Quality Assurance Testers
5. Documentors
Note: If you are planning to contribute to any of our github repos we require that you sign the FusionPBX Contributor
License Agreement. This mainly protects FusionPBX and our users read: you from code that could be inserted that
might pose a legal problem. It does this by verifying that the code you are contributing is yours to give and the you
give it freely and irrevocably to the project.
How to Get Started:

1. Watch the FusionPBX Pull Requests with Github Youtube Video https://youtu.be/SPUe7S4Z6ms
2. If you have a good handle on PHP, Lua or SQL Development Might be the thing for you head over to the
Development Manual
3. Are you a FusionPBX power user and do you possible love to write? Check out the Documentation Guide or
the Testing Guide
4.9. Contributing 109

4. Would you like to see FreePBX in your native language and have the time to commit to staying on top of
translations for releases? Check out the translation section to learn how to use our translation server.
Note: Contributing Code or Documentation requires knowledge of Git, Github and how to create pull requests on
Github. This is not as bad as it sounds and if you are willing to learn we will help you through it.
Contributing Code
Note to External Contributors
Hello, External Person!

We at FusionPBX are eager to work with you.
In particular, in order for us to accept any patches from you, you will have to electronically sign the contributor license
agreement [Signing the CLA]/en/latest/contributing/signing_the_cla.html
Thanks, FusionPBX!
Signing the CLA
https://github.com/fusionpbx/open-source
Open Source at FusionPBX
This repository serves as the umbrella project to represent the various open source efforts of Mark J. Crane(https:
//fusionpbx.com). Come here to get an overview of the various projects, to learn how to contribute to them, and to
sign up as a contributor.
Table Of Contents
[Sign the CLA](https://github.com/Fusionpbx/opensource/blob/master/sign-cla.md) - Step 1 [Required]

[CLA Rationale](https://github.com/Fusionpbx/opensource/blob/master/cla-rationale.md) - [Optional if Curi-
ous]
[Contributors](https://github.com/Fusionpbx/opensource/blob/master/contributors) - [List of Contributors]
[Contributor License Agreement 2.0](https://github.com/Fusionpbx/opensource/blob/master/cla-2.0.md) - [The
Actual CLA]

In particular, in order for us to accept any patches from you, you will have to electronically sign a statement that
indicates two things:

You are willingly licensing your contributions under the terms of the open source license of the project that
youre contributing to.
You are legally able to license your contributions as stated.
The reason we do this is to ensure, to the extent possible, that we dont taint the projects we manage with contribu-
tions that turn out to be improper. This protects everyone who wants to use the projects, including you! If you want a
longer explanation, then you can check out the [CLA Rationale page](https://github.com/Fusionpbx/opensource/blob/
master/cla-rationale.md).
Once you sign the Contributor License Agreement (the CLA), we will then be able to merge your contributions with
a clear conscience and with only the friction that results from the usual technical back-and-forth of a vibrant open
source project.
To get started with this process, visit the [Sign the CLA](https://github.com/Fusionpbx/opensource/blob/master/
sign-cla.md) page.
Thanks, FusionPBX!
List of Projects
[FusionPBX](https://github.com/Fusionpbx/fusionpbx) The Official FusionPBX Repo.

[FusionPBX Apps](https://github.com/Fusionpbx/fusionpbx-apps) Applications for FusionPBX.
[FusionPBX Scripts](https://github.com/Fusionpbx/fusionpbx-install.sh) Install and Upgrade Scripts for Fu-
sionPBX.
[FusionPBX Documents](https://github.com/Fusionpbx/fusionpbx-docs) This site.
[Open Source Umbrella Project](https://github.com/Fusionpbx/opensource) Signed contributor licenses agree-
ments.
Contributing Documentation

Thanks, FusionPBX!
Contributing Translations

Thanks, FusionPBX!

Quality Assurance Testing

Thanks, FusionPBX!
Rebasing a branch with GitHub
Welcome to the rebasing a branch with GitHub guide.

1. Open the GitHub client application and make sure you are on the branch you want to rebase
Select the setting icon and choose Open on Git Shell

You will be presented with a new shell

2. Next execute the following two commands and notepad will appear
git fetch --all

git rebase --ignore-date --interactive fusionpbx/master
change the first commit to reword and the following commit(s) to fixup (similar to screenshot below)
Close and save the text, next it will pop up another notepad for the commit message. Enter the commit title on the first
line, leave a line blank and enter the commit message (similar to screenshot below)

Close and save the text again.

3. Switch back to the github client and switch between history/changes to make it update and check it has done what
you want.

4. If the changes are correct switch back to the git shell and execute this to push the changes
git push --force-with-lease

exit
All done!
Coding Standards
FusionPBX Best practice coding practices

Command (example):
# contributing/directory_structure

Documentation Guide
This page shows an nice overview of the reStructuredText syntax. This is not a comprehensive list of everything you
can do, but should be enough to get you up and running to contribute some really nice documentation. It is based on
resources found at Sphinx .
To get your own local documentation repository running, simply
Introduction
The reStructuredText (RST) syntax provides an easy-to-read, what-you-see-is-what-you-get plaintext markup syntax
and parser system. However, you need to be very precise and stick to some strict rules:
like Python, RST syntax is sensitive to indentation !
RST requires blank lines between paragraphs
This entire document is written with the RST syntax. In the right sidebar, you should find a link Edit on Github,
which will show each page in reStructuredText raw text format.
Contents
Documentation Guide
Introduction
Getting Started
* Getting Git Right

* Setting up the Docs Locally
Text Formatting
* Inline markup and special characters (e.g., bold, italic, verbatim)

* Headings
* Internal and External Links
* List and bullets
What are directives
Code and Literal blocks
* How to include simple code

* code-block directive
* Include code with the literalinclude directive
Tables
* Simple tables
* Multicells tables, first method
* Multicells table, second method
* The tabularcolumns directive
* The csv-table directive
4.10. Documentation Guide 117

The toctree directive

Images and figures
* Include Images
* Include a Figure
Boxes
* Colored boxes: note, seealso, todo and warnings

* Topic directive
* Sidebar directive
Others
* Comments
* Substitutions
* glossary, centered, index, download and field list
* Footnote
* Citations
* More about aliases
* Intersphinx
* file-wide metadata
* metainformation
* contents directives
Useful extensions
* pngmath: Maths and Equations with LaTeX

* TODO extension
Getting Started
Getting Git Right
Learn Git in 15 Minutes Git Tutorial that will help you get started if you prefer. There is also awesome Git Tutorials
on the Atlassian Git site. Here is the link on installing Git if you dont have it yet Git Install
Setting up the Docs Locally
One of the great things about Git and documentation is that all people who contribute are encouraged to setup their
own local copy of the docs for off-line editing. This by default will ensure that many backups of the documents exist
and there is never any concern about losing them.
Assuming you have Python already, install Sphinx locally:
$ pip install sphinx sphinx-autobuild
Clone the FusionPBX Github documentation repository:

$ cd /path/to/where_you_want_the_docs
$ git clone https://github.com/fusionpbx/fusionpbx-docs.git
$ cd fusionpbx-docs
Edit files or add new ones then build your changes:
$ make html
Open index.html with your web browser and check your changes:
fusionpbx-docs/build/html/index.html
Edit your files and rebuild until you like what you see, then commit your changes and push to the public repository.
Assuming the file you changed is called myfile.rst:
$ git add myfile.rst

$ git commit -m 'your commit message'
$ git push -u origin master
Text Formatting
Inline markup and special characters (e.g., bold, italic, verbatim)
There are a few special characters used to format text. The special character * is used to defined bold and italic text
as shown in the table below. The backquote character ` is another special character used to create links to internal or
external web pages as you will see in section Internal and External Links.
usage syntax HTML rendering
italic *italic* italic
bold **bold** bold
link `python <www.python.org>`__ python
verbatim ``*`` *
The double backquote is used to enter in verbatim mode, which can be used as the escaping character. There are some
restrictions about the * and `` syntax. They
cannot not be nested,
content may not start or end with whitespace: * text* is wrong,
it must be separated from surrounding text by non-word characters like a space.
The use of backslash is a work around to second previous restrictions about whitespaces in the following case:
this is a *longish* paragraph is correct and gives longish.
this is a long*ish* paragraph is not interpreted as expected. You should use this is a
long\ *ish* paragraph to obtain longish paragraph
In Python docstrings it will be necessary to escape any backslash characters so that they actually reach reStructured-
Text. The simplest way to do this is to use raw strings by adding the letter r in front of the docstring.
Python string Typical result
r"""\*escape* \`with` "\\"""" *escape* `with` "\"
"""\\*escape* \\`with` "\\\\"""" *escape* `with` "\"
"""\*escape* \`with` "\\"""" escape with ""

Headings
In order to write a title, you can either underline it or under and overline it. The following examples are correct titles.
*****
Title
*****
subtitle
########
subsubtitle
**********************
and so on
Two rules:
If under and overline are used, their length must be identical
The length of the underline must be at least as long as the title itself
Normally, there are no heading levels assigned to certain characters as the structure is determined from the succession
of headings. However, it is better to stick to the same convention throughout a project. For instance:
# with overline, for parts
* with overline, for chapters
=, for sections
-, for subsections
^, for subsubsections
, for paragraphs
Internal and External Links
In Sphinx, you have 3 type of links:

1. External links (http-like)
2. Implicit links to title
3. Explicit links to user-defined label (e.g., to refer to external titles).
External links
If you want to create a link to a website, the syntax is
`<http://www.python.org/>`_
which appear as http://www.python.org/ . Note the underscore after the final single quote. Since the full name of the
link is not always simple or meaningful, you can specify a label (note the space between the label and link name):
`Python <http://www.python.org/>`_
The rendering is now: Python.

Note: If you have an underscore within the label/name, you got to escape it with a \ character.
Implicit Links to Titles
All titles are considered as hyperlinks. A link to a title is just its name within quotes and a final underscore:
Ìnternal and External links`_
This syntax works only if the title and link are within the same RST file. If this is not the case, then you need to create
a label before the title and refer to this new link explicitly, as explained in Explicit Links section.
Explicit Links
You can create explicit links within your RST files. For instance, this document has a label at the top called
rst_tutorial, which is specified by typing:
.. _rst_tutorial:
You can refer to this label using two different methods. The first one is:
rst_tutorial_
The second method use the ref role as follows:
:ref:`rst_tutorial`
With the first method, the link appears as rst_tutorial, whereas the second method use the first titles name found after
the link. Here, the second method would appear as Documentation Guide.
Note: Note that if you use the ref role, the final underscore is not required anymore.
List and bullets
The following code:
* This is a bulleted list.

* It has two items, the second
item uses two lines. (note the indentation)
1. This is a numbered list.

2. It has two items too.
#. This is a numbered list.

#. It has two items too.
gives:
This is a bulleted list.
It has two items, the second item uses two lines. (note the indentation)


Note: if two lists are separated by a blanck line only, then the two lists are not differentiated as you can see above.
What are directives
Sphinx and the RST syntax provides directives to include formatted text. As an example, let us consider the code-block
syntax. It allows to insert code (here HTML) within your document:
.. code-block:: html
:linenos:
<h1>code block example</h1>
Its rendering is:
1 <h1>code block example</h1>
Here, code-block is the name of the directive. html is an argument telling that the code is in HTML format, lineos is
an option telling to insert line number and finally after a blank line is the text to include.
Note that options are tabulated.
Code and Literal blocks
How to include simple code
This easiest way to insert literal code blocks is to end a paragraph with the special marker made of a double coulumn
::. Then, the literal block must be indented:
This is a simple example::
import math
print 'import done'
or:
This is a simple example:

::
import math
print 'import done'
gives:
This is a simple example:
import math
print 'import done'

code-block directive
By default the syntax of the language is Python, but you can specify the language using the code-block directive as
follows:
.. code-block:: html
:linenos:
<h1>code block example</h1>
produces
1 <h1>code block example</h1>
Include code with the literalinclude directive
Then, it is also possible to include the contents of a file as follows:
.. literalinclude:: filename
:linenos:
:language: python
:lines: 1, 3-5
:start-after: 3
:end-before: 5
Tables
There are several ways to write tables. Use standard reStructuredText tables as explained here. They work fine in
HTML output, however, there are some gotchas when using tables for LaTeX output.
The rendering of the table depends on the CSS/HTML style, not on sphinx itself.
Simple tables
Simple tables can be written as follows:
+---------+---------+-----------+
| 1 | 2 | 3 |
+---------+---------+-----------+
which gives:
1 2 3
Size of the cells can be adjusted as follows:
+---------------------+---------+---+
|1 | 2| 3 |
+---------------------+---------+---+
renders as follows:
1 2 3
This syntax is quite limited, especially for multi cells/columns.

Multicells tables, first method
A first method is the following syntax:
+------------+------------+-----------+
| Header 1 | Header 2 | Header 3 |
+============+============+===========+
| body row 1 | column 2 | column 3 |
+------------+------------+-----------+
| body row 2 | Cells may span columns.|
+------------+------------+-----------+
| body row 3 | Cells may | - Cells |
+------------+ span rows. | - contain |
| body row 4 | | - blocks. |
+------------+------------+-----------+
gives:
Header 1 Header 2 Header 3
body row 1 column 2 column 3
body row 2 Cells may span columns.
body row 3
Cells may span rows.
body row 4
Cells
Multicells table, second method

contain
The previous syntax can be simplified:
===== ===== ====== blocks.

Inputs Output
------------ ------
A B A or B
===== ===== ======
False False False
True False True
===== ===== ======
gives:
Inputs Output
A B A or B
False False False
True False True
Note: table and latex documents are not yet compatible in sphinx, and you should therefore precede them with the a
special directive (.. htmlonly::)
The tabularcolumns directive
The previous examples work fine in HTML output, however there are some gotchas when using tables in LaTeX: the
column width is hard to determine correctly automatically. For this reason, the following directive exists:
.. tabularcolumns:: column spec

This directive gives a Ccolumn specC for the next table occurring in the source file. It can have values like:
|l|l|l|
which means three left-adjusted (LaTeX syntax). By default, Sphinx uses a table layout with L for every column. This
code:
.. tabularcolumns:: |l|c|p{5cm}|
+--------------+---+-----------+
| simple text | 2 | 3 |
+--------------+---+-----------+
gives
title
simple text 2 3
The csv-table directive
Finally, a convenient way to create table is the usage of CSV-like syntax:
.. csv-table:: a title
:header: "name", "firstname", "age"
:widths: 20, 20, 10
"Smith", "John", 40
"Smith", "John, Junior", 20
that is rendered as follows:
Table 4.1: a title

name firstname age
Smith John 40
Smith John, Junior 20
The toctree directive
Sooner or later you will want to structure your project documentation by having several RST files. The toctree
directive allows you to insert other files within a RST file. The reason to use this directive is that RST does not have
facilities to interconnect several documents, or split documents into multiple output files. The toctree directive looks
like
.. toctree::
:maxdepth: 2
:numbered:
:titlesonly:
:glob:
:hidden:
intro.rst
chapter1.rst
chapter2.rst

It includes 3 RST files and shows a TOC that includes the title found in the RST documents.
Here are a few notes about the different options
maxdepth is used to indicates the depth of the tree.
numbered adds relevant section numbers.
titlesonly adds only the main title of each document
glob can be used to indicate that * and ? characters are used to indicate patterns.
hidden hides the toctree. It can be used to include files that do not need to be shown (e.g. a bibliography).
The glob option works as follows:
.. toctree::
:glob:
intro*
recipe/*
*
Note also that the title that appear in the toctree are the files title. You may want to change this behaviour by changing
the toctree as follows:
.. toctree::
:glob:
Chapter1 description <chapter1>
So that the title of this section is more meaningful.
Images and figures
Include Images
Use:
.. image:: _static/images/logo.png
:width: 200px
:align: center
:height: 100px
:alt: alternate text
to put an image

Include a Figure
.. figure:: _static/images/logo.png
:width: 200px
:align: center
:height: 100px
:alt: alternate text
:figclass: align-center
figure are like images but with a caption
and whatever else youwish to add
.. code-block:: python
import image
gives
Fig. 4.1: figure are like images but with a caption

and whatever else youwish to add
import image
The option figclass is a CSS class that can be tuned for the final HTML rendering.
Boxes
Colored boxes: note, seealso, todo and warnings
There are simple directives like seealso that creates nice colored boxes:
See also:
This is a simple seealso note.
created using:
.. seealso:: This is a simple **seealso** note.
You have also the note directive:
Note: This is a note box.
with

.. note:: This is a **note** box.
and the warning directive:
Warning: note the space between the directive and the text
generated with:
.. warning:: note the space between the directive and the text
There is another todo directive but requires an extension. See Useful extensions
Topic directive
A Topic directive allows to write a title and a text together within a box similarly to the note directive.
This code:
.. topic:: Your Topic Title
Subsequent indented lines comprise

the body of the topic, and are
interpreted as body elements.
gives
Your Topic Title
Subsequent indented lines comprise the body of the topic, and are interpreted as body elements.
Sidebar directive
It is possible to create sidebar using the following code:
.. sidebar:: Sidebar Title

:subtitle: Optional Sidebar Subtitle
Subsequent indented lines comprise

the body of the sidebar, and are
interpreted as body elements.
Sidebar Title
Optional Sidebar Subtitle
Subsequent indented lines comprise the body of the sidebar, and are interpreted as body elements.

Others
Comments
Comments can be made by adding two dots at the beginning of a line as follows:
.. comments
Substitutions
Substitutions are defined as follows:

.. _Python: http://www.python.org/
and to refer to it, use the same syntax as for the internal links: just insert the alias in the text (e.g., Python_, which
appears as Python ).
A second method is as follows:
.. |longtext| replace:: this is a very very long text to include
and then insert |longtext| wherever required.
glossary, centered, index, download and field list
Field list
Whatever this is handy to create new field and the following text is indented
:Whatever: this is handy to create new field
glossary
.. glossary::
apical
at the top of the plant.
gives
apical at the top of the plant.
index
.. index::
download
:download:`download samplet.py <_downloads/sample.py>`
gives download sample.py

hlist directive
hlist can be use to set a list on several columns.

.. hlist::
.. hlist::
:columns: 3
* first item
* second item
* 3d item
* 4th item
* 5th item
first item
second item
3d item
4th item
5th item
Footnote
For footnotes, use [#name]_ to mark the footnote location, and add the footnote body at the bottom of the document
after a CFootnotesC rubric heading, like so:
Some text that requires a footnote [#f1]_ .
.. rubric:: Footnotes
.. [#f1] Text of the first footnote.
You can also explicitly number the footnotes ([1]_) or use auto-numbered footnotes without names ([#]_). Here is
an example1 .
Citations
Citation references, like [CIT2002] may be defined at the bottom of the page:
.. [CIT2002] A citation
(as often used in journals).
and called as follows:
[CIT2002]_
More about aliases
Directives can be used within aliases:

1 this is a footnote aimed at illustrating the footnote capability.

.. |logo| image:: _static/images/logo.png

:width: 20pt
:height: 20pt
Using this image alias, you can insert it easily in the text |logo|, like this . This is especially useful when dealing
with complicated code. For instance, in order to include 2 images within a table do as follows:
+---------+---------+-----------+
| |logo| | |logo| | |longtext||
+---------+---------+-----------+
this is a longish text to include within a table and which is longer than the width of the column.
Note: Not easy to get exactly what you want though.
Intersphinx
When you create a project, Sphinx generates a file containing an index to all the possible links (title, classes, functions,
...).
You can refer to those index only if Sphinx knowns where to find this index. THis is possible thanks to the intersphinx
option in your configuration file.
For instance, Python provides such a file, by default Sphinx knows about it. The following code can be found at the
end of a typical Sphinx configuration file. Complete it to your needds:
# Example configuration for intersphinx: refer to the Python standard library.

intersphinx_mapping = {'http://docs.python.org/': None, }
file-wide metadata
when using the following syntax:
:fieldname: some contents
some special keywords are recognised. For instance, orphan, nocomments, tocdepth.
An example of rendering is the toctree of top of this page.
orphan
Sometimes, you have an rst file, that is not included in any rst files (when using include for instance). Yet, there are
warnings. If you want to supprresse the warnings, include this code in the file:
:orphan:
There is also tocdepth and nocomments metadata. See Sphinx homepage.

metainformation
.. sectionauthor:: name <email>

Specifies the author of the current section.:
.. sectionauthor:: John Smith <js@python.org>
By default, this markup isn Ct reflected in the output in any way, but you can set the configuration value
show_authors to True to make them produce a paragraph in the output.
contents directives
.. contents::
.. contents:: a title for the contents

:depth: 2
depth indicates the max section depth to be shown in the contents
Useful extensions
In the special file called conf.py, there is a variable called extensions. You can add extension in this variable. For
instance:
extensions = [-
'easydev.copybutton',
'sphinx.ext.autodoc',
'sphinx.ext.autosummary',
'sphinx.ext.coverage',
'sphinx.ext.graphviz',
'sphinx.ext.doctest',
'sphinx.ext.intersphinx',
'sphinx.ext.todo',
'sphinx.ext.coverage',
'sphinx.ext.pngmath',
'sphinx.ext.ifconfig',
'matplotlib.sphinxext.only_directives',
'matplotlib.sphinxext.plot_directive',
]
pngmath: Maths and Equations with LaTeX
The extension to be added is the pngmath from sphinx:
extensions.append('sphinx.ext.pngmath')
In order to include equations or simple Latex code in the text (e.g., ) use the following code:
:math:`\alpha > \beta`

Warning: The math markup can be used within RST files (to be parsed by Sphinx) but within your pythons
docstring, the slashes need to be escaped ! :math:`\alpha` should therefore be written :math:`\\alpha`
or put an r before the docstring
Note also, that you can easily include more complex mathematical expressions using the math directive:
.. math::
n_{\mathrm{offset}} = \sum_{k=0}^{N-1} s_k n_k
Here is another:

1
offset =
=0
It seems that there is no limitations to LaTeX usage:

1

1
column
= , row
= .
=0 =+1
TODO extension
Similarly to the note directive, one can include todo boxes but it requires the sphinx.ext.todo extension to be added in
the conf.py file by adding two lines of code:
extensions.append('sphinx.ext.todo')
todo_include_todos=True
Todo
a todo box
Bibliography


Bibliography
[CIT2002] A citation (as often used in journals).
135
136 Bibliography
Index
A
apical, 129
C
contents (directive), 132
H
hlist (directive), 130
S
sectionauthor (directive), 132
137

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FusionPBX Documentation

Jul 30, 2017

1. Adding extra functionality to the incredibly robust FreeSWITCH VoIP Platform.

6 Chapter 2. Benefits of FusionPBX

Call Block Call Broadcast Call Flows Call Center

8 Chapter 3. FusionPBX Features

Welcome to the FUSIONPBX installation guide.

apt-get update && apt-get upgrade -y --force-yes

apt-get update && apt-get upgrade

10 Chapter 4. Our Ecosystem

Installation has completed.

Use a web browser to login.

Official FusionPBX Training

4.1. Getting Started 11

If you need help upgrading safely please consider paid support.

12 Chapter 4. Our Ecosystem

Command line settings

4.1. Getting Started 13

#delete freeswitch logs older 3 days

echo "Backup Complete";

From the Gui.

Goto Advanced > Default Settings.

path array /var/backups/fusionpbx/postgresql True

path array /usr/share/freeswitch/scripts True scripts

path array /var/lib/freeswitch/storage True storage

14 Chapter 4. Our Ecosystem

path array /etc/freeswitch/conf True conf

Click "Reload" at the top of the page.

FreeSWITCH Source install paths.

Settings for FreeSWITCH source backup paths.

path array /var/backups/fusionpbx/postgresql True postgresql

Click "Reload" at the top of the page.

4.1. Getting Started 15

FreeSWITCH Package install paths.

16 Chapter 4. Our Ecosystem

Basic ports used

Iptables are used in the Debian install script.

iptables -A INPUT -i lo -j ACCEPT

OPENVPN: iptables -A INPUT -p udp --dport 1194 -j ACCEPT

Rules to block not so friendly scanner

iptables -I INPUT -j DROP -p tcp --dport 5060 -m string --string

4.1. Getting Started 17

iptables -I INPUT -j DROP -p tcp --dport 5080 -m string --string

iptables -I INPUT -j DROP -p tcp --dport 5060 -m string--string

Show iptable rules

Show line numbers

iptables -I INPUT -s 62.210.245.132 -j DROP

apt-get install iptables-persistent

18 Chapter 4. Our Ecosystem

Packet Filter is used in the FreeBSD setup script.

set skip on lo0

antispoof for lo0

pass out quick all

Firewall device settings that help with SIP connections.

20 Chapter 4. Our Ecosystem

22 Chapter 4. Our Ecosystem

Accounts -> XMPP manager.

on the right to create a profile.

Profile Name: gtalk

Two approaches can be used for the next part.