Todo lo lo que necesita saber para implementar FreePBX

Todo lo que debe saber
para empezar a implantar
FreePBX

Agenda
• Introduccion
• El Portal de Miembros
• Opciones disponibles de Instalación
• Instalando el Distro
• Registrando y Activando su implementación
• Agregando funcionalidades y módulos comerciales
• ZULU!!! Anuncio
• Preguntas y Respuesta
© 2015 Sangoma Technologies 2

Rápida revisión del Proyecto
• Open Source
– GPLv3 +
• Base Instalada
– Base instalada estimada en mas de 2,000,000
– Crecimiento estimado de 50 mil por mes
• Estabilidad probada con alta madurez
– 10/14/2004 – 1.1 (AMP)
– 03/17/2006 – 2.0 (FreePBX)
– 05/16/2006 – 2.1
– 01/05/2007 – 2.2
– 08/25/2007 – 2.3
– 02/10/2008 – 2.4
– 09/19/2008 – 2.5
– 2.6, 2.7, 2.8. 2.9, 2.10, 2.11, 12
• …. … he perdido la cuenta ☺

FreePBX Distros
Algunas palabras sobre los
Distros
• Distros populares
– FreePBX Distro
– AsteriskNOW
– PBX-in-a-Flash
– Elastix
– TrixboxCE
• Appliances de Sangoma
– FreePBX
– PBXAct
Para esta sesión:
FreePBX Distro
• Módulos Comerciales
• Plataformas
soportadas
• Actualizaciones

Regístrese en el Portal
Objetivos del Portal:
• Proveer a usuarios y canales un
punto de acceso para registrar
cada instalación de FreePBX
• Tener acceso a recursos de
soporte
• Tener acceso a software y
hardware para adquirir online
• Abrir tickets y dar segumiento a
casos de soporte
• Etc… (cambios ocurriendo….)

Acceda el Portal
• Toda implementación de
FreePBX tiene asociado un
Deployment ID
• El Deployment ID es una
identificación única y se
asocia a un usuario del portal
• El Deployment ID se obtiene
mediante la activación de la
instalación
…..Veremos los beneficios de
esto en un momento

Anunciando Zulu!!
FreePBX Zulu UC
• Integración Outlook y Browser…
• Desarrollado por el mismo team que
esta detrás de la PBX mas popular y
usada del mundo de Open Source.
Permite fácil integración con aplicación
que la gente usa diariamente.
• Precio de promoción pre-reléase de
US$ 199.00 usuarios ilimitados. Valido
hasta Octubre 31.

Obteniendo el Distro

Creando su Instalación
• Inicie su computador o VM usando el
ISO/USB
• Efectúe un “full Install”
• Apóyense en wiki.freepbx.org
• Full Install: Automáticamente activa
RAID 1 si hay dos discos presentes
• No RAID: no activa el RAID 1 a pesar
que existan discos.
• Advanced: Permite particinamiento
manual y definición de los discos, RAID,
etc.
• HA: Para instalar una maquina que va a
formar parte de una configuración HA

Configurado la instancia
• Pueden usar los valores
por defecto
• O seleccionar la
configuración IP que les
convenga

Configurando la Instancia
• Seleccione la zona horaria
correspondiente
• Asigne el Password del “root”

Iniciando la Instancia
• Al final se produce un reboot y se inicia la
nueva maquina
• Es deseable que la maquina tenga
conectividad internet para completar la
actualización de módulos

Proceso de Instalación
• El proceso puede
tomar nos 15 minutos

Primer boot
• Estamos listos para el primer
Login
• Observen la IP asignada por
el DHCP de su red o
predefinida por ustedes en
forma estatica

Primeros pasos GUI
• Acceda via http la IP indicada en el Login
• Defina un usuario y password para la cuenta
del administrador

Login como Administrador
• Hagamos Login con la cuenta de
administrador recién creada

Inicio en el dashboard
• Estado del Sistema
• Observar si existe alguna
alarma que deba tener
nuestra atención
– Modulos sin actualizar
– Fallas en Asterisk o inicio de
algun modulo
– Acividades pendientes de
registro
– Etc…

Algunos ajustes iniciales
• System Admin

Algunos Ajustes Iniciales
• Activacion

The FreePBX Distro
Commercial Modules
• Purchase Commercial Add on modules to expand and
enhance your FreePBX System.
• We will spend a small amount of time talking about this in
more detail later on in the class and in the Sales and
Marketing and Reseller Section.
• Helps fund the project and allows us to keep the lights on.
• Don’t worry we are not here to pitch you for 3 days on buying
modules or anything else.

The FreePBX Distro
Support
• Purchase Support Contracts from the FreePBX Project
with SLA contracts.
• Purchase Hourly Support credits
• You can view support options at
http://www.freepbx.org/support-and-professional-services
• Like commercial modules this helps fund the project and
allows us to keep the lights on.

The FreePBX Distro
Upgrades
• FreePBX GUI updates can never touch anything at the operating system level
such as kernel, Asterisk or any of the other 500 packages used to make up your
phone system.
• To update the Distro we publish upgrade scripts for each track version of the
Distro. We create a new track for each OS or FreePBX GUI major release and
inside a track will have multiple versions. For more information on Versions and
Tracks view our wiki at
http://wiki.freepbx.org/display/FD/Updating+FreePBX+Official+Distro
• Our current tracks are;
– 6.12.65 which is EL 6.5 and FreePBX 12- STABLE
– 5.211.65 which is EL 6.5 and FreePBX 2.11- EOL
• For the class we are using the Stable track of 6.12.65.

The FreePBX Distro
Upgrades
• To view which track your system is on you can go into the sysadmin module
• You can also view the version from the Linux CLI with the following command.

The FreePBX Distro
Upgrades
• Upgrades can be done 3 different ways.
– You can find which version you are on and download each upgrade script based on the wiki for your version. Then run
each script in order to get to the version you want.
http://wiki.freepbx.org/display/FD/Updating+FreePBX+Official+Distro
– You can execute the following script from the linux CLI and it will update you to the latest version in your track for you.
Please note this will only work if you have registered your PBX with our License Server which we will do in a few
moment.

The FreePBX Distro
Upgrades
– If you have the Commercial
Sysadmin Pro you can pick
which version your want to
upgrade to and also set
automated schedules on how
often to look for new upgrades
and install them automatically
for you.

Initial Setup of FreePBX
System
• OBE
• Register Deployment
• Install SSH Client
• Update Distro
• Sysadmin Module
– Set Time Zone.
– Set harddrive failure/fillup notification
– Intrussion Detection Settings
– Add Email and Setup whitelist for the whole subnet of 192.168.101.0/24
• Initialize several Advanced Settings

OBE (Out of Box Experience)
• This is our initial screen after first install.
The purpose of this is to allow you to
setup your GUI username and password
for FreePBX. Instead of having a standard
predefined username and password we
make you set one.
• The main purpose for this is so there are
not default usernames and password
floating around the web that hackers can
use to try and hack into your system since
we know most users never change these
as history shows.

Portal and Registering your Deployment
• You should already have a Portal
Account and be able to login.
• A portal account is required:
– If you ever want to engage FreePBX
Professional Support and Services
– To license free and paid commercial
modules
– To order any other services or products
from FreePBX
• Since we will be using different
commercial module, you will register
today's deployment.
https://schmoozecom.com/oss-registration.php?view=register

• We will now walk through how to Register your PBX
to a deployment. There are 3 ways of doing this.
1. One is to login to the portal.schmoozecom.com
and create a deployment. You would then paste
the deployment ID into the System Admin
module under License to link this PBX with the
Deployment Number.
2. The other option is to go into the System Admin
module and under license have the system reach
into our portal.schmoozecom.com for you and
auto create a deployment and link this PBX to
that Deployment Number it created.
3. A new option is when purchasing a module within
module admin - a deployment ID will
automatically be created and assigned at
checkout out
• We will use option 2 as we are using a Discount
code to get a bunch of modules for use in the class.

Advanced Settings, Default Menu Items,
etc.
• /etc/freepbx.conf
– Minimal configuration: Database
Credentials and bootstrap path
• Advanced Setting
– Everything Else

(Advanced Settings)
• Many Purposes
– Change defaults for page configurations
– Expose hidden menu items
– Enable advanced/less used features
– Change file system configuration and related
– Allow Branding/Styling Changes
– Modify Asterisk Manager Credentials
– Control logs and logging levels
– Enable Developer Features
– View some internal system settings
• Read Only
– Some settings are “read only” because of
they are more ‘dangerous’ and should be
changed with caution.
– Display Read Only Settings will show these
– Override Read Only Settings will allow
changes

(Advanced Settings - examples)

(Registering your Deployment)
• Log into the System Admin module and click on License
– Would you like to register this deployment now.
• click Yes
– Do you have a Deployment ID that is not tied to another
Hardware System. This would be no since we did not create
a Deployment in the portal for this system ahead of time
• click No

(Registering your Deployment)
– Do you have a Portal account.
• Click Yes as you already have an account on the
portal.schmoozecom.com site for the FreePBX
Store/Portal.
• Put in your email address that is used with the portal
and click the Register button.
– Location Name- Here you define a friendly name for this
PBX that is easy for you to identify this location or PBX.
We will use OTTS Lab as our location name.
– Click Register when done.
– Your PBX has now been registered with the
License Server and a Deployment ID has been
created. You can now obtain commercial
modules for this system which we will be doing
in a bit..

ssh install
• Windows systems should download Putty if not
already installed:
– You can download it from:
http://downloads.freepbxdistro.org/OTTS/
– Put it on your desktop or somewhere convenient,
we will be using it a lot during the next 3 days
• For Mac:
– Applications -> Utilities -> Terminal
(put it in your menu bar, you will need it a lot)

• Type in the username of root
and press enter
• Type your root password of
freepbx and press enter.

Distro Releases and Upgrades
• We will use the simple script
option that we discussed earlier
to make sure our systems are
upgraded to the latest version of
the Distro
• You can also change the
Asterisk version at anytime by
using the built in asterisk version
switch script: asterisk-version-
switch. For the Class we will be
using Asterisk 11. PLEASE DO
NOT CHANGE IT.

Setup System Admin module
• Setup the following options
in the System Admin
module in the FreePBX
GUI.
– Timezone Change
– Set harddrive failure/fill-up
notification
– Intrusion Detection Settings
– Add email and setup whitelist
for the whole subnet of
192.168.101.0/24 (or whatever
your local subnet will be)

Advanced Settings
Setup some settings under the
Advanced Settings module in FreePBX
GUI
– Developer and Customization:
– System Setup
– Expose all Device Setting when
adding extension
– Send P Asserted Identity and
manage NAT

Overview - Lab
• Create 2 SIP Extensions
• Download Xlite and Setup
• Dial Echo Test
• Purchase and Install OTTS Bundle of Commercial
Modules.
– Discount Code (Code TOBEDEFINED)
• Setup Desk Phone with EPM
• Dial between 2 Extensions

LAB: Setup Extensions
• Create 2 SIP Extensions
– Create Ext Number
• See your welcome sheet on what extensions
to use.
– Give a display name
– Enable voicemail and set a voicemail
password
– Set any other options you would like to
change or set
• TIP- Notice the defaults on ext for things
like sendrpid is the Passerted Identity
since earlier in advanced setting we told
FreePBX to default that to PAI.

LAB: Setup X-Lite -Windows
• Download Xlite if you do not have it
already
– Setup Xlite with one of the extension
you created on your PBX.
– Display name, User ID, Authorization
name:
• => your extension number
– Password => your SIP Secret
– Domain => PBX IP Address

LAB: Setup X-Lite –MAC/OSX
• Download Xlite if you do not have it
already
– Setup Xlite with one of the extension
you created on your PBX.
– Display name, User ID, Authorization
name:
• => your extension number
– Password => your SIP Secret
– Domain => PBX IP Address

LAB: Simple Dialing
• Verify calling ability and assure 2
way audio:
– Dial your Echo Test feature
code.
• Hint: look at the feature code
module to get the “echo
test”feature code.
– Setup your voicemail box.
• Hint: look at the feature code
module to get the “check
voicemail” feature code.

LAB: Obtaining Licenses
• We will now login to the portal and purchase a few modules we need.
– OTTS Bundle
• Use Discount Code TOBEDEFINED which will allow you to get these modules for free.
• Appointment Reminder
• Broadcast
• Call Recoding Report
• Caller ID Management
• Class of Service
• Conference Pro
• Extension Routing
• End Point Manager
• Fax Pro
• Outbound Call Limits
• Park Pro
• Pinset Pro
• Q Xact Reports
• Restful Phone Apps
• System Admin Pro
• UCP for EPM
• VM Notify
• Voicemail Reports
• VQ Plus
• Web Call Me
• XMPP

• Login to http://portal.schmoozecom.com
– Email Address=
– Password=

• Click on the Store at the top. Then click on
FreePBX Software
• On the left side click on Software Bundles
• Go find the following modules and Press
the Add Icon to add the item to your
Shopping Cart:
– FreePBX-CM-OTTS Bundle

• Press the Checkout Button on the right side
once you have the bundle
• Pick the deployment number from the drop
down next bundle item. This should be the
deployment number that was generated
earlier today. Use the Discount code of
OTTS2015 to purchase the products with no
need for payment and press the REDEEM
button

• To finish the checkout process Pick Check as your payment type
and Agree to the Terms and Conditions and press the “Process
Order” button

• Go back into the System Admin module
under license and press the Update
License button. You should now see a
license and expiration date for your new
modules.

LAB: Using EPM
• We are now going to go into EPM module
to setup our Desk Phone.
– EPM is a Commercial module used to
setup and manage phone configs for
over 20 manufactures and supports
over 250 devices
• We are going to just go over the basics
here as a interactive lab and get your
Desk Phone setup.
• Navigate to your End Point Manager
module under Settings section

LAB: Using EPM
• Click on Global Settings at the top.
– Here we need to define some
global settings.
– Define the IP address of your PBX
under the Internal IP section and
press the submit button when done.
• Take note of the HTTP provision
port defined here as we will need it
later when we tell the phone how to
reach the server for config files.
• Press the Submit button when done

LAB: Using EPM
• Click on the firmware management tab on the
right hand side and then click on Brand of
Phones you are going to use
• Drag the latest firmware to slot 1 and press
submit – wait a minute or two and refresh the
page to make sure its been downloaded as
shown.

LAB: Using EPM
• Select the Phone Brand to create a template

LAB: Using EPM
• Fill in the following fields for our new
template
– Template Name- Friendly name for
this template
– Destination Address- Pick Internal
which will pull the IP Address we
defined earlier in Global Settings or
you can just type in any IP or FQDN
– Time Zone Settings
– Provision Server Address should be
the same IP as the Destination
Address above
– Provision Server Protocol should be
HTTP.
– Firmware Version should be slot 1
• Save Template when done

LAB: Using EPM
• We are now going to setup some buttons on
our phone (using SNOM 870) by checking
the S-870 phone.
• For Line Key 1 we are going to set it up as
out ext
– Type-Line
– Account-Account1
• For button 2 lets setup a BLF to monitor our
X-lite phone.
– Type- BLF
– Label- Name of softphone
– Value- Extension number we want to
monitor which is the soft phone.
– Account-Account1
• Press the Save Model button

LAB: Using EPM
• Lastly we need to map our Desk Phone to
a extension on our system and what
template we want it to use. Under
Extension Mapping section of EPM we will
do the following setup.
– Extension- What extension on the system
we want to map
– Brand- What phone brand will this extension
be using
– Template- What template would we like to
build the phone config off of.
– MAC Address- MAC Address of this phone
– Model- What model number is this phone.
• Press the Save button at the bottom of the
page and EPM will write out the
configuration files for this device.

LAB: Using EPM
• Now we just need to point our Desk
Phone to the PBX IP address and
HTTP or FTP/TFTP port that is shown
in the Global Settings, making sure
the phone supports such protocol for
provisioning.
• Press the Settings button
• At this point you should program your
phone for provisioning and test!!!

Initial trunking to PSTN
Overview
Initial Trunk and PSTN Setup
• Setup a SIP trunk with SIPStation
• Configure Inbound route for your assigned DID
• Create a simple 10 digit Outbound Route
– Test calls between your classmates
– Test calls from your cell phones
– Test calls to your cell phones

Sip Trunk Setup
Go to
https://sipstation.schmoozecom.com/
• Login with your same username and password you used for the portal earlier to
buy your commercial modules.

Sip Trunk Setup
• SIPStation allows you to create more then 1 location under the same account.
Make sure you are using the location at the top of OTTS LAB. As this location was
added to your account to allow you to buy the trunks and DIDs we need free of
charge for the class.

Sip Trunk Setup
• Add 2 call trunks to your account. Each
trunk purchased allows 1 inbound or
outbound call with unlimited normal
business usage calling
• Pick a DID for your account
• Click the Checkout Button. Your shopping
cart should show 2 Trunks and the DID.

Sip Trunk Setup
• Agree to the terms and
pick the Confirm Order
& Charge my Card
button.
– No Credit Card is
needed since we
marked your account
as free for the class.

Sip Trunk Setup
• Now we can go get our key code for easy
setup in FreePBX. Click on My Account
tab at the top.
• Provide a valid e911 address.
• Enable SMS and T38 Faxing for a future
lab exercise

Sip Trunk Setup
• Copy the FreePBX Module keycode into your clipboard.

Sip Trunk Setup
• Go back to your PBX and open up the SIPStation module under
Connectivity tab and paste in the keycode and press Add Key

Sip Trunk Setup
• This will pull in all of our information from SIPStation and present to us our DIDs and let
us setup routing of this DID. Lets route this DID to our main extension.
– Give the Route a Description Name such as Main DID
– Optionally set the failover number. In the event we can not reach your PBX when
someone calls this DID we will failover to the number provided here such as your Cell
Phone. This can also be managed from inside the SIPStation Store.
– Pick your main phone extension on where to route calls on this DID to.
– Press Update DID Configuration when done.
• Now Press the Apply Config button to write out all your changes.

Sip Trunk Setup
• Our trunks should now show Green and Registered in the module after we refresh
the page.

Sip Trunk Setup
• So why use the SIPStation module.
– What this just did was setup 2 trunks to our trunk1 and trunk2 for
redundancy in FreePBX.
– Setup your DID as a inbound route in FreePBX.
– Setup outbound routes for calling outbound in FreePBX.
• All this was done from the SIPStation module without you
having to go into 4 modules and waste 20 mins to create all
this like every other provider.

Recap

Recap
• From within the SIPStation module in FreePBX you can do some basic account management such as
– Setup Failover Number.
• This can be a global failover that anytime we can not reach your PBX with a call we will forward to this Phone Number or IP Address.
• You can also set a per DID failover. If set we will use the DID failover instead of the Global Failover number.
– Manager e911
• Manage your Default e911 address and what DID it is associated with.
• Setup additional DIDs with their own e911 addresses. A per month charge does apply for this.
• We can also see a few basic items like
– Are our trunks Registered.
– Do we have International Calling enabled
– Have we enabled outbound T38 Faxing
– Have we enabled SMS inbound and outbound faxing on our account.
• ! Lastly you can manage your full account as if you were logged into our website all from the SIPStation
module without ever leaving our PBX.

Recap
• We have now created a SIP trunk to a provider
• We have configured a simple inbound route to one extension
• We have configured a simple 10 digit outbound route to the PSTN
• We should be able to make calls between ourselves and to the real PSTN
• FreePBX allows easy setup of SIP, IAX or DAHDI trunks.
– SIP- Open Standard for VOIP trunking that most providers use
– IAX- Open Standard developed by Digium and is a Asterisk only trunking protocol
• Inter-Asterisk eXchange
– DAHDI- This is the open source software driver that allows you to connect asterisk to
the standard PSTN of Analog, T1 such as PRI or E1 and BRI’s
• Digium Asterisk Hardware Device Interface

Inbound Routes Other options
Inbound Routes – Other
Options
• Alert Info and CID name
prefix
• Privacy Manager
• CID Lookup Sources
• Channel Language
• Call Recording
• Fax Detection and
Routing
– What to do IF a Fax signal is
detected

Call Routing
So far we’ve got
• Extensions/Mailboxes
• Inbound Routes
FreePBX call flow:
• Building blocks need to be created
• Call flow construction then needs to be
constructed working “backwards”
Next Pieces:
• Call Distribution
– Ring Groups
– Queues
• IVR
Company DID
8004522233
75
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Recordings
• Announcements
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Recordings
• Announcements
• Ringgroups, Queues, Destinations
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Recordings
• Announcements
• IVRs
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Recordings
• Announcements
• IVRs
• Time Conditions
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Recordings
• Announcements
• IVRs
• Time Conditions
• Call Flow Controls
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Company DID
8004522233
Call Routing
Working Backwards
• Recordings
• Announcements
• IVRs
• Time Conditions
• Call Flow Controls
• Inbound Routes
• Analog Channel DID
IVR
1-Sales
2-Support
3-Directions
Call Flow
Control
Sales
Ringing
Support
Queue
Support
Manager
Sales
Manager
After Hr
Msg
John’s VM

Call Routing
Working Backwards
• Just Got Easier!
– Requires >= 2.11
• Destination popOvers:
– Destination Modal Box
– One “generation” deep
– Supports Most
Destinations

Distributed Calling
Overview
Distributed Calling
• Finding someone to answer the phone
– Distributed Calling: when FreePBX sends calls to multiple extensions or
otherwise seeks to find someone to answer the phone
• Most commonly achieved with:
– Ring Groups
(more people to answer than callers coming in)
– Queues (ACD)
(more callers coming in than people to answer)
• Advanced Call routing and escalation rules

Distributed Calling
ACD/Ring Groups
© 2015 Sangoma Technologies
86
Ring Groups: Each Agent will receive all 3
calls at the same time
Queue
If Line Busy
Will try next
agent
If Line Busy
Will try next
agent
Agent 1
Agent 2
Agent 3
Agent 2 Agent 3Agent 1
A
A
A
A
B C
Calls B and C
remain on
Queue until A
is assigned
Queues No Autofill
Ring Groups

Distributed Calling
ACD/Ring Groups
• Queues, autofill
(default 1.4+)
– Each call distributed
to an available
agent, any
additional calls
queued
Caller A
Caller B
Caller C
Agent 1
Agent 2

Distributed Calling
Lab: Ring Group Setup
• Create a Ring Group
– Add both of your extensions
– Add your Cell Phone or Office
Number
– Hard Set a Caller ID to be
used when calling external
numbers
– Set Failover Destination to be
a voicemail box

Distributed Calling
Lab: Queue Setup
• Create a Queue
– Add 1 extension as
dynamic and 1 as static
• Log in Dynamic member
– Hint look in feature
codes module for
how to do this
• Set Failover Destination to
be a voicemail box
• Call into the queue.

Distributed Calling
Lab: Extra Credit!!
• Create a special voicemail greeting to be used by the
queue and ring group when going to your voicemail box
as the failover destination. This greeting should only be
used by the queue and ring group and your normal
greeting played for all other callers.
GO FOR IT!!!!!

IVR & Misc Applications
Overview
• Discussion IVR
• Discussion System Recordings
• Discussion on Call Flow Toggle
• Setup a System Recording for your IVR
• Setup IVR
• Setup a Misc App to IVR
• Setup Call Flow Toggle

IVR Discussion
• IVRs (Auto Attendants) are used to route calls to different areas of your PBX by giving the
caller options.
– Simple Example:
• “Press 1 for sales, 2 to reach an operator”
• It’s typically bad practice to provide more than 3-4 IVR options at one level of an IVR.
• If you have more options, “nest” the IVRs:
– “For sales press 1, for support press 2, for the operator press 3, to hear our hours of operations, fax
number, or directions to our store press 4. Option 4 would route to another IVR.
– When nesting, always provide option to “Return” back up
– Always set a “timeout destination” where calls will go if the user doesn’t choose an option.
– Always set an “invalid destination” if a caller presses an invalid option too many time.
– Set the loop count to allow multiple attempts before being directed to the “timeout” or “invalid”
destinations.

System Recording Discussion
System Recordings – a digression
• Before getting started, we need to make sure we can make recordings for our next few labs
• System Recordings allow you to create greetings/recordings from any phone on the PBX or upload audio
files.
• The recordings can then be used in other modules such as the Queue or IVR Greeting to guide callers
through your system.
• These recordings are used throughout the system.
• To make it easy to re-record a specific recording, enable a feature code that can be dialed from any phone
and it will walk you through how to re-record that recording. Nice way of changing these without going into
the GUI.

Call Flow Toggle Discussion
• A Call Flow Toggle is simply a on and off light switch or diversion.
– Route a Call through the Call Flow Toggle
– Pick a ON and OFF destination. In Call Flow we call them Normal and Override.
• When in Override mode if a BLF button is set to monitor the Call Flow it will turn on or red.
• Used to divert calls by a manual process only.
• Usually used where you will never have a automated schedule were a Time
Condition would work better. We will talk about Time Conditions later on.

Lab: System Recordings Setup
• On the initial screen of the
System Recordings module
you can either:
– pick an extension to use to
dial into the recording
system to make your
recording. We will be
using this option today.
– upload a pre-recorded file
in the proper format

• Type in your Extension Number of your phone and
press the Go button
• It will now instruct you to dial *77 to record your
greeting and will walk you though the review
process. Now dial the *77 and create a simple
recording that we will change later after the IVR is
setup. Say something like “thanks for calling.”
• When done give the recording a name. Make sure
not to put in any spaced in the name or it will
complain. Press Save when done.

• On the right side we should see a list of
recordings that have been made. Click
on the recording you just created.
• Enable the Feature Code option. This
will allow you to record your message
with the designated feature code (*293
here).
• When we create an IVR later we will use
this code to re-record a proper message
inline with the IVR you will be creating.

Lab: IVR Setup
• Create an IVR with the following Options:
– Add announcement you just created.
– Ring Group
– Queue
– Remote Voicemail Access
– TimeOut Destination
– Invalid Destination
• Dial the feature code for your
announcement you created earlier and re-
record it with the options above.
– When doing your recording for the IVR don't
announce what to press to check voicemail as this
would be a hidden option for your users to check
their voicemail while out of the office. Make it a hard
option that someone wont dial by accident like 6712

Lab: IVR Setup
• Create a Misc App to call your IVR from
any internal phone
– Great way to test your IVR without calling
in from a trunk
• Point your inbound route for your DID to
your IVR now and call from the outside
world.

Lab: Call Flow Toggle
• Create a call flow toggle between your IVR
and a general Voicemail Box.
• Point your DID to this Toggle so the call can
flow through it.
• Call your DID and it should ring your IVR.
• Dial your Toggle. Default feature code is
*28 plus the index so our example would be
*280.
• Call your DID and it should ring to your
Override Destination in our example this is
a Voicemail Box.
• Setup a BLF button on your phone to be
the feature code. Our example this is *280.
(Hint go into EPM and update the template
then reboot the phone so it will get a new
config)

Outbound Call Flow
Discussion
Outbound Call Flow
• Configure Routes and Trunks
– Setup EMERGENCY Route
• Make sure it is first!
– Add 7 digit dialing to our Local Route but have it auto add the area code of your DID since that is your local
area code
– Setup 11 Digit route for Long Distance Calls
– Setup a 411 Route for Information
– Set up Extension CID to be 414-888-8888 and Extension Emergency CID to be your DID number
– Test Outbound Calls
• Review and discussion – what did we just do?
– Route and Trunk Call Flow
• Route Dial Patterns
• Trunk Number Manipulation Rules
– Outbound CallerID Choice

Outbound Call Flow
Discussion
The order of the Routes on the right side is
important to determine the outcome of what route
will be used. When you make a outbound call it
starts with the top rule and works down until it
finds the first match. Once a match is found it does
not continue on looking for a “better” match.
This means if you have a route called Long
Distance that has 1NXXNXXXXXX and you also
have a route of Toll Free which has
18XXNXXXXXX you need to have the Toll Free
route above the Long Distance Route or the Long
Distance route will match when dialing a
1800XXXXXXX number and a call will never use
the Toll Free route.

Outbound Call Flow
Routes, Trunks, Dial Patterns
Outbound Route
Patterns
• Pattern Chooses the
route
• Route Choice is “final”
– Subsequent routes that
also match the number
will never be tried
• Pattern can remove
leading digits or add
new ones
– Common example,
emulate old style PBX
– 9|1NXXNXXXXXX
Tip: Always make your first route your Emergency Route
to handle E911 and other related
calls. This assures that numbers matching your
emergency patterns will ALWAYS go down
the Emergency Route.
Route 1 (Emergency)
Route 2
Strip
Disgits if
configured
Bad
Number
Context
Next
trunk
Next
trunk
Normal
Congestion
or Optional
Destination
Route 3
Match?
Busy,
Answer, No
Answer
Busy,
Answer, No
Answer
Congested, Channel
Unavailable
Congested, Channel
Unavailable
Match?
Match?

Outbound Call Flow
Lab: Emergency Route
CallerID & E911
• Setup Emergency CIDs based
on physical device location
• Use Emergency Routes
– Put Emergency Route first
– They will use Emergency CID
over all others
• Check with your telco if you
have unusual circumstances
– remote extensions
– emergency CID not part of
your DID blocks
– other
Extension or Device
Route with Digitally Enabled Trunks
Always verify that
your trunks can
transmit your
required
CIDS if it is part of
an E911
configuration

Outbound Call Flow
Lab: Emergency Route
EMERGENCY
• Make sure it is FIRST
• Make sure “Route Type”
Emergency is checked
• Drag and Drop it above
the Local route created
earlier
• MAKE SURE FOR LAB
TO USE 933 not 911

Outbound Call Flow
Lab: Local Route
Local

Outbound Call Flow
Lab: Long Distance Route
Long Distance

Outbound Call Flow
Lab: Information Route
Information

Outbound Call Flow
Lab: Extra Credit!!
Extra Credit
• Create a Route for Toll Free calls.
• Modify your information Route so that when some one
dial 411 it actually dials a free 411 Information Server
like 1800-FREE411
• Modify Long Distance route so only your Desk Phone
can dial it but your softphone can not.
– Hint try using the CallerID field in the dial patterns section of
outbound routes.

Outbound Call Flow
Route Patterns + Trunk Rules – putting it together
• Example: Lazy Dial an International Number
• (07031) 278325
Exten 222 dials: 0 7031 278325 ext 222
Sent to Trunks: +49 7031 278325
Sent to this trunk: 011 7031 278325

Outbound Call Flow
Route Patterns + Trunk Rules
• Which do I use?
• Trunk Rules:
– Usually trunk specific
• Carrier requires specific format
• Carrier doesn’t accept 10 digit local calls
• Route Patterns:
– Applying a consistent dialplan
• Standardize area code for 7 digit dialing
• CallerID specific rules
• Route patterns generate much more
efficient dialplan than trunk rules
CallerID Handling == confusing
• Follow-Me and RingGroups Can manipulate
CallerID
• Outbound Routes Can Add CallerID
• Trunks Have more options to control
CallerID

Outbound Call Flow
CallerID when you make a call
CallerID Hierarchy – Not an exact science!
• Emergency CallerID Overrides All
– If the call is going down an Emergency CID Route, including Trunks set to force their CallerID always.
• Extension CallerID
– If the route or trunk followed isn’t configured to explicitly override this
• Route CallerID
– If the trunk followed isn’t configured to explicitly override this
• Trunk CallerID
– If the trunk followed isn’t configured to explicitly override this
– If you have a carrier that rejects calls with CNAM, you can choose to remove all CNAMs on the trunk here, but still use the above
CallerIDs
• Always set a trunk CallerID on a digital trunk
– The above hierarchy means it will only be used if no other CallerID is provided, unless you force otherwise on the trunk
– Not sending a proper CallerID can result in rejected calls by many carriers, the trunk CallerID makes sure you send something.

Outbound Call Flow
CallerID when you “forward” a call
Call Forwarding, Follow-Me, Ring Groups
(how their CallerID is handled)
• Follow-Me or Ring Groups
– Leave the original CallerID as it was (same as Call Forward)
– Set a “fixed” CallerID (for all calls, or just outside calls)
– Set it to the inbound DID that was dialed (from outside calls only)
• Route CallerID
– NEVER APPLIES for Follow Me, Call Forward or Ring Groups
• Trunk CallerID
– Will normally pass the original CallerID, or in the above cases, the modified CallerID if present.
– Exceptions:
• Block Foreign CallerID: will block all but the “fixed” CallerIDs you can set above. Will allow the “DID” replacements from above
if they were set to be forced.
• Force Trunk CallerID: will always use the Trunk’s CallerID, period.
• If no CallerID is present, the Trunk CallerID will be used.

Outbound Call Flow
CallerID Examples
Extension
4002
Extension CallerID settings
Outbound Route CallerID settings
Trunk CallerID settings
In this example the final Caller ID that is used is the
Extensions Caller ID 9208868132
since the outbound route was set to not override
Extension and the Trunk is set to
Allow ANY CID
Extension
4002
In this example the final Caller ID that is used is the
Extensions Caller ID 9208868132
since the outbound route was set to not override
Extension even though it had a Caller
ID set and the Trunk is set to Allow ANY CID

Outbound Call Flow
CallerID Examples
Extension
4002
In this example the final Caller ID that is used is the Route
Caller ID 9208868130
since the outbound route was set to override Extension
and the Trunk is set to Allow
ANY CID
Extension
4002
In this example the Caller ID that is used is the Trunk
Caller ID 4256540156 even
though the outbound route was set to override Extension
but since the Trunk is that last
item to control Caller ID and it was set to Force Trunk it
wins out.

Outbound Call Flow
CallerID Examples & Tips
TIPS:
• Caller ID Name Displayed on Phones
• If using a phone that supports rpid such as the Digium and
Aastra phones you will notice when you make a external
call the Caller ID number that was used on the call you
placed will be displayed as the “CID Name” portion of the
CID field on the phone since the Name Field is not used for
outbound calling otherwise.
• This is handy way to know what Caller ID was used when
making any call.
• This can be turned off in advanced settings module.
Extension
4002
In this example the final Caller ID that is used is the Trunk
Caller ID 4256540156
since the Extension and Outbound Route had no Caller
ID set and the Trunk is set to
4256540156.

Follow Me
Overview
Extension and Follow Me configurations:
• Extensions can optionally have a Follow Me
– Not all Extensions have to have a Follow Me
– Advanced settings allow for a Follow Me to be automatically created when an extension is created, or you can
manually create if not enabled.
• Follow Me enable/disable
– When a Follow Me is configured, it can be enabled or disabled
– Each Follow Me has a BLF generated to see the enabled/disabled state and toggle the state:
*21XXXX where XXXX is the extension number
• Follow Me REST App can also be used to enable/disable a Follow Me and control many other features, as
can UCP.

Follow Me
Overview
Extension and Follow Me configurations:
• Call routing with Follow Me:
– When a Follow Me is enabled, all calls to that extension will go to the Follow
Me, as well as all call flows that send a call to that extension as a destination.
• Except as members of Ring Groups or Queues depending on your Ring Group and Queue
Settings
– When a Follow Me is disabled, the calls will only go to the extension.
– Follow Me can not be chosen as a module destination, EXCEPT:
• VmX Locater can choose its own ‘disabled’ Follow Me
• An Extension destination can choose its own ‘disabled’ Follow Me

Follow Me
Overview
Extension Destinations:
• Provide more control on how to handle unanswered calls:
• Can control behavior for:
– UNAVAILABLE
– BUSY
– UNREACHABLE (CHANUNAVAIL, e.g. phone is offline)
• Options:
– Voicemail if available
– Any standard destination (including other extension’s voicemail)
– This extension’s Follow-Me (even if disabled)

Follow Me
Lab: Softphone Follow-Me when offline
Softphone Follow-Me
• Configure for ringallv2
• Disable – ONLY use if Softphone is offline
• Add your cell phone and desk phone extensions
• Make sure to use confirmation
• Send calls to voicemail of your desk phone to have a unified voicemail
box.

Follow Me
Lab: Softphone Follow-Me when offline
Go to extension page for your soft phone.
• Go to Optional Destinations
• Not Reachable:
– Force Follow Me
• Try the CID prefix
• Now disable the softphone and test a
call

VmX Locater
Overview
VmX Locater (Voicemail Extension)
• Provides a ‘personal IVR’
• Uses the Voicemail Greetings
• Provides 3 configurable options
• Option 0:
Can use default ‘0 out’ operator configuration of FreePBX or override with a specific internal or external number.
• Option 1:
Can force a call to Follow-Me which is otherwise currently disabled, or can send a call to a specific internal or external number.
• Option 2:
Can send a call to an internal or external number.
• VmX can be engaged for an unanswered call, busy call, or both. If not enabled for one mode, then the call will go
straight to voicemail in that mode.

VmX Locater
Overview
Change Hardphone Voicemail:
• Unavailable Greeting
– “I am not currently available, you can leave me a message or if it’s urgent you can press 1 and the system will try to
find me”
• Busy Greeting
– “Unfortunately I am not currently reachable, please leave me a message and I will get back to you as soon as I
receive it”
• Now when a caller calls:
– If you are on the phone and don’t answer the second call, they will get your busy message and the only option will be
to leave voicemail.
– If you are not on a call and the phone is not answered, they will get your unavailable greeting. They will be given the
option to leave a message, or to have the system find you, which will then engage your Follow-Me.
• You can also have option 2 with a phone or internal number as an option
• You can override option 0 from the normal system default
– If Follow-Me fails to find you, they will be directed to your Busy Voicemail message and since VmX was disabled for
Busy, they will not be able to continue engaging your voicemail and will only be left with the option of leaving
voicemail.

VmX Locater
Lab: Hardphone VmX plus Follow Me
• Go to the Extension page in
FreePBX for your desk phone and
configure your VMX Locator
• Enable VmX Locater
• Set Option 1 to go to your Follow
Me
• Configure a BLF for your follow
me through EPM and reboot your
phone.
– BLF is: *21XXXX (XXXX == your
extension)

VmX Locater
Lab: Hardphone VmX plus Follow Me
Hardphone Follow-Me
• Configure for ringallv2-prim
– Initial ring of about one ring
• Disable initially – but use with
VmX
• Add your softphone and cell
phone
• Make sure to use confirmation
• Choose Hardphone BUSY
voicemail

Follow Me Advanced
Ring Strategy for Follow Me
Is Primary
Ext
Occupied? Yes
Ring
Primary
Only for
Ring Time
Answer
NoAnswer
To Destination, if No Answer
Continue call flow as specified
in module config
Follow Me / Ring Groups
-prim mode Ring Strategy

Follow Me Advanced
Is Primary
Ext
Occupied?
Ring
Strategy
Include
Primary
Answer
Go Through Ring
STrategy
No
Someone
Answers
in module config

Follow Me Advanced
Is Primary
Ext
Occupied? Yes
Ring
Primary
Only for
Ring Time
Answer
Go Through Ring
STrategy
No
Someone
Answers
in module config
No
Answer

Intra Office Trunking
Overview
Overview
• Setup Interoffice Trunk
– We will do this in multiple Configurations
• When Static IP addresses are available at all branches
• When a branch has a Dynamic IP addresses
• Setup Interoffice Route

Lab: Setup Static Trunk

Lab: Setup Static Trunk & check IAX status
Outgoing Settings:
Trunk Name: TO-TONY
PEER Details:
username=philippe
type=friend
trunk=yes
secret=notsecure
qualify=yes
insecure=port,invite
host=192.168.1.186
context=from-internal
auth=md5
requirecalltoken=no

Lab: Setup Test Route
• Setup Route to capture 4 digit
dialing
• Setup Route for remote Echo Test
– In this example, “943” is the remote
echo test
– We could have configured 9*43 but
some phones may not be setup to
transmit that sequence, so we show
an example of the use of prepending
and prefixes to transmit *43 to the
remote PBX when 943 is dialed.
• Now Try dialing the remote
extension

Discussion-Pitfalls and Next Steps
• At this point you can
– Echo test to each system
– Dial each systems extension
• Pitfalls
– If you mis-dial, you have a potential infinite loop
– Ways to Address this:
• Max Channels on the trunks
• More restrictive pattern matching then XXXX
• More restrictive pattern matching with the CallerID field
• Next Step
– One or both PBX’s do not have a Static IP
– Use registration to the one that is Static
• If both are dynamic, choose the more “stable” of the two, make it the static side, and use a FQDN coupled
with a Dynamic DNS Update service
– Next Example, PHILIPPE becomes the Dynamic PBX

LAB: Change to Dynamic Trunks

LAB: Accesing DID’s
• We can now call internal
extensions, next step DIDs
• Modify your routing and
setup patterns to dial the
remote DID though the new
trunk
– Add the DID of your parner
to your Route Dial Patterns
– Change the order so the
Intra-Company Trunk is
used first

LAB: Accesing DID’s
• Now try dialing
– ss-noservice …
– We have NOT exposed our EXTERNAL
facing numbers to this route since we are
configured with from-internal.
– We could change to from-pstn but then
our internal dialing ability will break. (Try
It)
• Configuring to access both
– Create from-branches in
/etc/asterisk/extensions_custom.conf from
the CLI using your SSH Client such as
putty
– Change the trunk to use the new context

Understanding from-branches
Example Dialpla Context Organization
Dialplan is split up into two main sections
• from-pstn
“WAN” Side of a firewall
• from-internal
“LAN” Side of a firewall
• Inbound Routes
– Similar to “IP port mapping” in a firewall
– Exposes Internally protected dialplan to the
“WAN”
– ext-did
Where FreePBX puts the inbound routes
“WAN ” / “LAN” Contexts in FreePBX
Calls from the “outside”
• [from-pstn]
• include => from-pstn-custom
• include => ext-did
• include => ext-did-catchall
Calls from the “inside” (simplified)
• [from-internal]
• include => from-internal-additional-custom
• include => app-xyz ;lots of them
• include => ext-group
• include => ext-findmefollow
• include => ext-local
• include => outbound-allroutes
Outbound Routes
• [outbound-allroutes]
• include => outbound-allroutes-custom
• include => outrt-001-Emergency
• include => outrt-002-dundi
• include => outrt-003-Local
• include => outrt-004-LongDistance
• include => outrt-005-International

Codecs
Codecs
• Linear Codecs
– ulaw (g.711u)
– alaw (g.711a)
– slin
• Compressed Free
– gsm
– g726
– adpcm
– lpc10
– speex
– ilbc
• Compress Licensed
– g729
– g723
MOS (Mean Opinion Score)
• Compressed codecs degrade quickly
with delay/packet loss
Tip: Use “allow=none” followed
by “allow=codec1&codec2” to
restrict a trunk or phone to
specific codecs.

Class of Service
Overview
• Overview Discussion
• Lab’s
– Create a new No Emergency Outbound Route
– Setup Class of Service to restrict emergency route from
soft phones.

Class of Service
Overview
• The Class of Service Administration module provides
granular control at the extension level to access and set
permissions of specific calling features of your PBX. These
features include Outbound Routes, Feature Codes, Ring
Groups, Queues, Conference Rooms, Voicemail Blast
Groups and Paging.
• The Class of Service module for FreePBX allows you to
restrict extensions from dialing most destinations of your
PBX.

Class of Service
Overview
• Outbound Routes
• Feature Codes
• Ring Groups
• Queues
• Conferences
• Page Groups
• Voicemail Blast

Class of Service
LAB: No E911 Route
Create Outbound Route to BLOCK callers from calling 933
• Route will go to an informative announcement
• Route should be positioned so human error will not easily allow normal phones to
use this route
• Steps:
– Create recording (try a built-in recording)
– Duplicate and rename Emergency Route
– Remove the trunk
– Reposition the route to be AFTER the real Emergency route
– Add a destination to play recording and pick your system recording
– Create Class of Service to restrict normal emergency route for Soft Phones
– Save and test

Class of Service
LAB: No E911 Route
Create the recording:
• Admin -> System
Recordings
• Choose option for
Built-in Recordings
and pick an
example:
• Create your
recording and save

Class of Service
LAB: No E911 Route
Create the announcement:
• Go to the Announcement
module and create a new
announcement
• Pick the system recording
you just created.
• Set the destination to be
Terminate Call> Hangup

Class of Service
LAB: No E911 Route
Create No-E911 Route:
• Go to Emergency Route and Duplicate it
• Rename it to No-E911
• Remove the trunk
• Change the Normal Congestion destination
to Announcement and pick your
Announcement we just created

Class of Service
LAB: No E911 Route
• Submit it, then submit your new route and move it after the Emergency
• route:
• Note: The placement of the second route was important. It will never be used if an
extension has access to the real Emergency Route. As a result, it is much more
likely to accidentally allow a soft phone to use the real emergency route then it is
to configure an internal extension that is blocked from it since you have to explicitly
disable the extension from the Emergency route when you create a new
extension. The default for new extensions is to have access to all routes.

Class of Service
LAB: Class of Service
Apply restrictions for the soft phone now.
• Go to the Class of Service module under admin and create a new Class of Service called No 911
• Add the softphone as a member
• Click on Routes and move the normal Emergency Route to the Deny list. Press save when done
and apply config.

Class of Service
LAB: Class of Service
Remember now that we have a single COS setup
for the Softphone anytime we add a new
destination like Ring Groups, Queues or anything
else the Softphone will not be able to dial those
new destinations until you modify COS to add the
items.

Paging & Paging Pro
Overview
• Overview Paging
• Overview of Paging Pro
• Setup Page Group
• Setup our 911 route to page our desk phone.

Paging & Paging Pro
Overview Paging
• Paging module allows you to setup a group of phones and when you dial
the group number all the phones will auto answer on speaker phone and
the pager can start talking.
• Only works on phones that support auto answer SIP signaling. Most soft
phones do not support this.
• Can pick if the phones being paged are on another call what to do;
– Skip- Don’t page the busy phone
– Force- Force the page as a new call to the phone
– Whisper- Barge in on the call and whisper to the phone the page. The active
caller will not hear the page as the whisper is to the phone only.

Paging & Paging Pro
Overview Paging
• By default the extensions in the page group can hear the
page but can not talk back to the person making the page.
• If you enable the Duplex mode all phones that are paged will
also not be muted so everyone in the page group can talk at
the same time to everyone else like a conference call.
• If Duplex is disabled any user who is paged can press *1 to
unmute themselves and start talking into the page group.
Pressing *1 again will re-mute them.

Paging & Paging Pro
Overview Paging Pro
• Outbound Notifications - Enables the ability to notify a group of phone(s) when a
user dials a specific number, ie: 911. Any page group can be linked in the
outbound routes module. When a call is placed a page will go out to the page
group notifying the page group of what number was dialed and what user dialed
the number. Any user of the page group can dial *1 to barge into the call and
speak.
• Prepend Recording - You can now have the page group weather normal or valet
style play a recorded message to all participants of the page group before the
pager can start speaking.

Paging & Paging Pro
Overview Paging Pro
• Valet Style Paging (Airport Style) - You can now choose to have your
pages recorded and when you hang up have it send the audio file to all
the devices that are part of the page group. This setting is done on a per
page group. You can also tell the system to only use Valet if someone
dials the page group and it is in use already.
– Do Nothing- If page group is busy play busy tone.
– Valet- If page group is in use and you dial the page group it will have you record your
page and when the other page group is done it will play the page to all the phones.
– Force Valet- Make all pages be valet all the time.

Paging & Paging Pro
Overview Paging Pro
• Scheduled Pages - Define
custom schedules to have the
system page a group of devices
and play a recording. This is a
great replacement for school
bell systems or lunch break
buzzers.

Paging
LAB: Page Group
• Create a new Page group.
– Extension Number- Something in your range assigned to
you
– Name- Page All
– Drag your Desk Phone into the Selected box
– Try the different busy phone options.

Paging Pro
LAB: Page Group
• PAGING PRO FEATURES
– Set busy option to be Force Valet and make a page.
– Record a system recording and have that played to the page group before
your page is played.
• Try this in both Force Valet and Do Nothing options
– Play with setting up a scheduled page.
• You can add more then 1 page schedule to a page group

Paging Pro
LAB: Outbound Notifications
• Create A NEW Page Group for outbound notification
– Provide a extension number
– Name- 911 Page
– Add your Desk Phone to the group by dragging it to the Select side
– Submit your changes

Paging Pro
LAB: Outbound Notifications
• Go to your emergency route and under notification pick your new page group you just
created.
• Submit your changes and apply config.
• Dial 911 from your softphone and your desk phone should auto answer and barge you
in on the call after playing info to you. You can press *1 to un-mute your deskphone
and start talking to both parties.

Company Directory
Overview
• Overview
• Setup a Directory
• Add option to IVR for Directory
• Lock IVR down for Direct Dial to the Directory
Entries only

Company Directory
Overview
• Company Directory is a list of extensions that a caller can enter the person’s first or last
name into the keypad of their phone to be connected to the extension.
• It is usually used as an IVR option.
• Starting in 2.8 you can now add custom entries into the Directory like remote extensions
from another system or even a Ring Group or Queue.
• You can also add entries for outside sales people with no extension that dials their cell
phone direct.
• We can use a directory as a “map” to determine what “direct dial” extensions can be called
from a given IVR and where they will be routed.
• You can create as many Directories as you want for specific departments or companies.

Company Directory
LAB: Creating a Company Directory
• Lets go to the Directory Module in FreePBX and click
Add New Directory

Company Directory
• General Settings
– Give the Directory a name like “All Users”
– Add a description to you remember what the directory is for
– Optionally set a CID Prepend that way when a call is sent to your extensions the Caller
ID will be prepended with DIR so you know the call came from the Directory

Company Directory
• Directory Options
– We need to set an Invalid Destination. This is
where calls will be routed to if they hit the
“Invalid Retries” threshold. In our example this
is 2 times.
• This means the caller can try and find a
match 2 times and if no match is found
they are sent to the invalid destination as
defined below.
– You can also change the default
announcement that is played to the caller to
something like “thanks for calling FreePBX.
Please start entering your parties first or last
name on your keypad.”
– “Return to IVR”, if enabled, will ignore the
“Invalid Destination” configured if the call
came from an IVR and route the caller back to
the IVR they came from instead.
– “Announce Extension”, if enabled, will
announce the extension number of the party
just prior to sending the call to them.

Company Directory
• We will now add actual entries to the Directory by pressing the green button at the bottom of
the page. Our options are:
– Picking an individual extension from the system to add to the directory
– Choosing all users to add to the Directory
– Adding a Custom Entry such as a Outside Phone Number, Ring Group or Queue or anything else
that has a number associated to it that the system can dial

Company Directory
• Add the Following to your Directory
– All of your extensions
– Your partner’s extensions
– Your cell phone

Company Directory
• Go to your IVR created earlier and
add an option for your newly created
Directory
• Lock the IVR Direct Dial by using this
Directory as a “map” to restrict only
those extensions as available for
Direct Dial.
– Now a caller can direct dial those
extensions but not others that may
be present on the system

Company Directory
Extra Credit!!!
• Make your Custom entries play the person’s name instead of Text To Speech
or Spelling the Name since Custom Extensions don’t have a voicemail name
to play like local extensions
• Override your Xlite extension so it dials your main Desk Phone if a caller
chooses your Xlite. This is great in a CEO/Assistance relationship where the
CEO’s name is in the directory but when they enter the CEO it rings their
assistant’s phone instead of theirs.
– Notice that the number is no longer grey-ed out. This is because this extension is no
longer linked to the user.
– By default, a local user that has not had the entry modified in the module will be
updated with any changes made in the extension page automatically. Once it has been
modified though it will no longer track what is there.

Time Conditions
Overview
• Overview
• Setup Time Group M-F 8-5 and Sat 9-4
• Setup Time Conditions
– Day-Ring Group
– Night-Night IVR
• Setup Feature Code for Override
• Dial Feature Code for overrides to test the override
• Point your DID to your new Time Condition
• Extra Credit

Time Conditions
Call Routing with Time Conditions
Time Conditions, Time Groups and Overrides
• Time Conditions + Time Groups == Flexible & Simpler Call Flows
– Time Groups: define your open hours
– Time Condition: define your destinations for open and closed
• Each Time Condition will have a Override Feature Code
– BLF hint generated, shows current open/closed state
– Feature Code allows current state to be toggled
– Normal flow automatically resumes in the subsequent time transition.
• Example, close early by pressing the BLF:
• BLF lights up, flow changes to ‘closed’
• At ‘5:00PM’ back to normal mode (BLF still lit since it’s now the normal ‘closed’ time.
• Next morning, normal flow resumes at 8:00am, BLF turns off
• Sticky Mode accessible through GUI only and now the RestApps
– Force ‘open’ or ‘closed’ – no auto-reset

Time Conditions
LAB: Call Routing with Time Conditions
• Create Time Group
– Monday-Friday 8:00AM – 5:00PM
– Saturday 9:00AM – 2:00PM

Time Conditions
• Create Time Condition choosing the above
Time Group
– Destination if time matches (‘open’) goes to
Ring Group
– All other times go to non-matching (‘closed’)
goes to IVR

Time Conditions
• Once you have created your Time
Condition you can now come back and
edit it.
– You will see the feature code override.
In our example it assigned *271. This
can also be programmed as a BLF
button on your phone.
• You can change the default assigned code if
desired from feature code admin module.
– You can optionally force the override
from the GUI dropdown.
• Now modify Inbound Route to point
from IVR to this Time Condition

Time Conditions
LAB: Extra Credit!!!
Extra Credit
• Brain Teaser:
– Add Thanksgiving Day into your Closed Hours
• Adjust the frequency that the background polling occurs
to keep the time conditions up to date
– Try turning it off and changing your times to confirm that it still
self adjusts even when off.

Asterisk BLF and Hints
Asterisk “state” Information
Understanding the relationships
• Understand some of the tools
– FreePBX GUI
– CLI commands
• Relationship with BLF keys
• Extension State (“hint”) Examples
– in FreePBX
– user created

Device states
• There are 3 “types” of devices/device states
– Channel states
• SIP/7134, SIP/PSTN, DAHDI/12, IAX2/tony, …
– Asterisk Generated “device states”
• Meetme:8000, park:72, ccss:SIP/7134, …
– FreePBX/custom user “device states”
• Created using DEVICE_STATE()
– Custom:DND7134
– Custom:DEVDND7134
– Custom:FOLLOWME7134
– Custom:DEVCF7134
• Device states are internal to Asterisk
– They are “Atomic”
– They can not be viewed directly

Extension states (hints)
“hints”
• An externally viewable “state”
• Made up of one or more device
states
• Must be defined in dialplan
• FreePBX examples:
The green numbers are the actual
hints you would subscribe your
phones BLF to monitor.

FreePBX generate hints
• FreePBX generates hints for the most commons feature codes
• The following example shows a limited number of these hints

LAB: Subscriptions (BLF)
• Subscriptions
– Once a hint is created you can view it from
the CLI
– Phones and other Endpoints can Subscribe
(with BLFs)
– CLI> core show hints
• You can see how many “watchers” have
subscribed
– CLI> sip show subscriptions
• You can see the specific subscriptions

Queues in Depth and Oddities
Overview
• Discussion
– The $$$$$ question. What is the behavior of penalties and why do most
people not use them.
• Edit queue with both dynamic and static agents.
• Create BLF button to log in and out of queues.
• Create BLF button to pause/unpause agents.
• Play with Queue Agents rest apps
• Play with ring strategies and penalties.
• Create a Virtual Queue to change some settings.

Discussion
• Static agents
• Dynamic Agents
• Penalties
• Agent Restrictions
• External Agents (note no # at end)
• Login methods
• Virtual Queues

Agents
• Static Agents (queue members)
– A static agent is an agent that can not log out of a queue
• They can be paused/unpaused with feature codes, auto-pause or third party Apps such Rest Apps.
– The agent can be set with a penalty
• Dynamic Agents
– Dynamic agents can log in and out of queues at any time
• Can also be paused as with static agents above.
– Auto generated BLF enabled login/out codes
• See feature codes panel for base code or look at dialplan generation
CAUTION:
• It’s possible to have the same agent configured as static and logged in as dynamic with
some third party apps. This can cause problematic behavior and should be avoided.

Agent Penalties
Agent Penalties
• Allows you to group skill sets of agents. Penalties start at 0 and go up, typical
range may be 0-100. The default and base penalty value is 0.
• When the Queue calls agents it starts with agents of penalty 0 and progresses up
from their by default.
• If you set a Ring Strategy of Ring All and have agents 101,0 102,0 103,4 and
109,99.
– The desired outcome would be to try agents 101 and 102 first. If unanswered call agent
103 next and finally call agent 109 if 103 fails to answer.
– Is that correct though????

Penalties
Penalties
• That would seem logical but that is not what always happens. If it calls agents 101
and 102 and both agents are logged in and available it will keep calling them and
never progress.
– An agent is considered available if they are not on the phone and are not in a paused
state.
– In order to progress to the next penalty, all agents at that level have to be unavailable,
busy or paused so the queue doesn’t attempt to ring them, not answering does not
make them unavailable. (See auto-pause if you want the system to automatically pause
them if not answering)
– Remember to log them out or pause them if the above described behavior is intended.

Debugging
• Debugging
– From the Asterisk CLI do
• Queue show 4501 which is our queue number
– We can see a list of all agents logged in.
– We can see which agents are dynamic
– See the penalty of each agent
– See if they are on a call. In Use or Not in use
• Enable DND and check your status
– List of callers waiting in queue
– Basic stats of the queue at the top

Agent Restrictions
Agent Restrictions
• Called As Dialed
– This will call the extension of the agent and honor all settings of this extension including Follow Me and Call Forward
settings.
– This can be dangerous resulting in calls being answered by a follow-me’s voicemail box destination or other
unexpected outcomes.
– This mode will not check to make sure the agent logged in is a valid extension of the system.
• No Follow Me or Call Forwarding
– This will dial the extension but not honor any Call Forwarding settings or Follow Me.
– Note: this only works for Server side Call Forwarding so phones like Polycoms that do device side Call Forwarding will
have no effect on this.
• Extensions Only
– This is the same as No Follow Me or Call Forwarding but will do validation checks and not send a call to any agent
that is not a valid extension number.

Allowing Follow Me or Call Forward to agents
• Allowing Follow Me or Call Forward to agents.
• Enable the Call Confirm option
This will force the queue to play a message to the agent it is calling if the agent is dialed
through an external phone number. It will play the message you select and tell them to
press 1 to take the call. This avoids issues such as a cell phone voicemail answering the
queue call.
– In your Queue Enable the Call Confirm Option
– Pick a recording to play to the caller. Usually something that tells them this is a call from
the sales queue press 1 to answer it.
– Hint: if an agent is configured through their follow-me AND has call confirm enabled, the
Queue’s message will override the Follow-Me’s message having the benefit of the agent
knowing the call is a queue call vs. a personal call.

Login Options
• Feature Code *45
– If dialed with no queue number at the end it will toggle your login state to either logged in or
logged out for all queues that agent is configured in the GUI as a dynamic agent.
– If you dial this as *45 + queue number, e.g. *455001 this will toggle your login state for queue
5001 only.
– BLF hints per agent per queue and for all queues)
• *45*4002 (BLF lit if agent 4002 is logged in any queue they are a member)
• *454002*5001 (BLF lit if agent 4002 is logged in queue 5001)
• Logging in/out through the CLI, example extension 4002 queue 5001
– queue add member Local/4002@from-queue/n to 5001 penalty 0 as “John Doe”
state_interface hint:4002@ext-local
– queue remove member Local/4002@from-queue/n from 5001

Pause Options
• Queue Pause and Auto Pause
– Agents can be paused, remaining in their Queue(s) but unavailable to take calls
– Agents can be auto-paused if they don’t answer a Queue call from the specific Queue or all Queues they are logged into.
• Feature Code *46
– Un-pause you in all queues you are a listed as a member (static or dynamic) if you are paused in any queue, otherwise pause you in
all queues agent is listed as a member of.
– If you dial this as *46 + queue number, e.g. *465001 this will toggle your paused state for queue 5001 only.
– BLF hints (requires FreePBX Distro or patch) per agent per queue and for all queues)
• *46*4002 (BLF lit if agent 4002 is paused in any queue they are a member)
• *46*4002*5001 (BLF lit if agent 4002 is paused in queue 5001)
• Pausing through the CLI
– queue {pause|unpause} member Local/4002@from-queue/n
– queue {pause}unpause} member Local/4002@from-queue/n queue 5001

Login/Pause Hints
• BLF Hint Key
– You can do a core show hints to see a hint code for each user with each queue. You can than program a BLF key to each of
those codes to log in and out of individual queues or pause agents.
• Asterisk CLI
– core show hints
• Create a BLF for one of them like *454002*4501
– *45 is the feature code to toggle login
– 4002 is your extension/agent number
– 4501 is your queue number.
• Do the same for pausing (*46), slight format change:
– *46*4002*4501

VQ Plus
• Build Dynamic Queue Penalty Rules that change the longer a caller waits in a queue.
– Example: Set a queue to only try agents with a penalty of 0-3 for the first 30 seconds, then only
try agents with a penalty of 2-5 for the next minute
• Virtual Queues to manage queue behavior and expand and customize caller destinations
for callers routed through the virtual queues
• Expanded Queue Destination Controls. The standard queue only allows you to send
unanswered calls to a single destination regardless of why the call was not answered. VQ
Plus gives you the ability to control destinations for additional reasons.
– Queue Fail Over on FULL Destination
– Queue Fail Over on JOINEMPTY Destination
– Queue Fail Over on LEAVEEMPTY Destination
– Queue Fail Over on JOINUNAVAIL Destination
– Queue Fail Over on LEAVEUNAVAIL Destination

VQ Plus
• Post Hangup Destinations
VQ Plus adds the ability to route both the agent and callers
to any destination on hangup of a queue call.
– Example: Route the inbound callers to a survey system
automatically when the agent hangs up the call, or send
the agents to a similar destination when the caller hangs
up the call.

Queue Priority
• By default anytime a caller enters a queue the caller has a priority
of 0. With VQ Plus you can change this caller priority before they
enter the queue to a different number much like agent penalties
except a caller with a higher number will get routed to a agent
before a caller with a lower priority number in the same queue.
– Example: we may setup a virtual queue for VIP customers that set
the Queue Priority for any caller to be 100. It then sends the call to
our normal support queue but with the priority of 100 so they will
automatically be moved above any callers with a lower priority and
their call will be answered first.

VQ Plus
• Create Virtual Queues. A virtual queue allows you to change the settings of a queue
before you route the call to your queue.
– Announcements- Played to the Agent informing them the caller is a VIP customer.
• Record the announcement in system recordings.
– Wait Time- Increase their wait time to be longer.
– Minimum and Maximum Agent Penalties- We will set the minimum agent penalty to be 1
and the max to be 100. This way the caller will not be routed to any of our tier 0 agents
since they are VIP we want them going to better qualified agents.
– Queue Priority- Set them to be 100 so the caller will be moved to the top of the queue.
– CID Prefixes- Set the Caller ID prefix to be VIP so the agents will see VIP on their Caller ID
when a call from this virtual queue is routed to them.
– Destination- Pick our normal queue as the queue to send the caller to. They will be sent to
the queue will all the above information set.

LAB!!
Lets now go setup some of the things we just talked about.
• Edit a Queue with both dynamic and static agents
– Soft Phone should be static agent with a penalty of 1
– Desk phone should be dynamic agent with a penalty of 0
– Call your DID and from IVR pick your queue option.
– Watch it call your Desk phone over and over and never call your Soft Phone.

LAB!!
• Create BLF button to log in and out of the Queue
– *45*EXT as BLF. Replace EXT withy your extension numbers.
• Create BLF button to pause/unpause a member
– *46*EXT as BLF. Replace EXT withy your extension numbers.
• Go to EPM and add the 2 BLFs above and reboot your phone
• Pause your Desk Phone by pressing the pause button.
– Inside asterisk CLI do the following command but replace 4501 with your queue
number. Verify it shows your extension as paused.
• Queue show 4501
• Call back in your DID and now you will see your soft phone will ring because we
made the Desk Phone be Not Available.

LAB!!
• Setup a virtual queue. Go to Applications>Virtual
Queues
– Name- VIP Customers
– CID Name Prefix- VIP-
– Queue Priority- This will put the caller into the queue at
a higher position. We will set position 100 so they will
be ahead of all other callers that have a lower priority.
Standard calls come in with a priority of 0. Not to be
confused with Agent Penalties.
• Change the min and max agent penalty to be 1 and
1. That way our
• VIP customer does not have to deal with tier 0
support.

LAB!!
• Route the virtual queue to your main queue.
• Point a option in your IVR to go to the new virtual queue. Option 20. Then you can
give out the secret code of 20 to your VIP customers that they can dial when they
get your IVR.

Call Park
Call Parking
Call Parking
• Pick the Parking Lot Extension that you
send calls to. Default 70
• Pick where the actual holding Lots start
at. Default is 71
• Pick the number of slots to enable. In our
example we have 8 set so they would be
71 thru 79
• Parking Timeout- This is the number of
seconds the caller will be parked before
going to the timeout destination defined
later.

Call Park
Call Parking
Call Parking
• CallerID Prepend- If a Parked Call is not picked up it will go to the destination defined below. When this
happens you can prepend the Caller ID with anything you want.
• Auto CallerID Prepend- Here you can pick some options to have auto prepended such as the slot
number the caller was parked at.

Call Park
Call Parking
Call Parking
• Come Back to Origin- If a parked call is not retrieved by someone and the timeout is met as defined
above do we want to send the Caller back to the extension that parked the caller.
Yes or No.
• Destination- If Come Back to Origin is set to No above then this is the destination we will send timed out
callers. Or if set to yes above but we are not able to send the caller back to the extension such as the
extension is now offline this is the destination that will be used.

Call Park
Call Parking
Call Parking Buttons
• Setup a BLF keys to view parked calls and retrieve them. For our example we will
setup 2.
– 71
– 72
• Go into EPM and modify your template and reboot your phone

Call Park
Call Parking
Call Parking
• In our example we will park callers by transferring them to “70”
• The caller will be put on hold and the slot number that the caller was transferred to
will be played back to you.
• In our example since no other caller is currently parked we would expect to hear
back slot 71 since that is our first starting slot number.
– We can now dial 71 to pickup the parked call.
– You can also setup a BLF to Monitor each slot number such as 71, 72 and anytime a
caller is in the slot the BLF light will show in use and you can press the BLF button to
retrieve the parked caller.

Call Park
LAB
Lab
• Point your inbound DID to your softphone
• Call your DID from your cell phone and answer the call
• Park the call from your softphone (##70)
• Your slot 71 BLF should turn red
• Pickup the call by pressing the BLF
– Note the Caller ID is the cell phone’s number, this is because the PAI configuration and not all
phones will do this
• You can also use the Parking REST App to pickup calls and view all the currently parked
calls with a single button. You still need a separate parking button to park a call even with the
REST App.

Call Pickup
Direct Call Pickup
Directed Call Pickup
• Allows anyone to answer a specific ringing phone
• Can be integrated with “smart” BLF phone keys
– When a phone is ringing and you have a BLF setup to it the BLF should blink. Pressing
the BLF on your phone should execute a Directed Call Pickup and let you answer that
call.
• Remote directed call pickup
– Because directed call pickup is an application in Asterisk dialplan, it is possible to
configure directed call pickup to a remote PBX through your intra-company route
– Extra Credit: Setup Remote Directed Call Pickup

Call Pickup
Experiment features
Directed Call Pickup
• Default is **XXXX (XXXX == extension)
• BLF buttons are often programmed to automatically implement a call pickup when pressed while the BLF is in the ringing
state (fast blinking).
– Note the “Reverse CID” behavior as with Parking
– Note the difference between the Softphone which does not implement the Reverse CID and the Hardphone which does.
• Try to configure your Intra-Company Trunk to pickup a ringing extension on the other PBX using ‘88’ as a prefix

Intercom
Discussion
Intercom
• Paging allows you to broadcast announcements to one or more devices simultaneously and
we went through the paging module earlier. Intercom allows you to page a specific users
• Intercom-ing is always to a User (not Device)
• Always duplex
• *80 is default feature code:
– *80XXXX
• Intercom must be enabled on the incoming extension and the phone must be capable of
auto-answering a call just like with paging. By default all extension are enabled for intercom.
– *54 – enable intercom
– *55 – disable intercom

Intercom
Lab: Auto-Answer local and Intra-branch
Intercom All Internal Extensions
• Can be set per phone or overall
• Requires Paging & Intercom
module installed and operational
• Set in Advanced Settings
– Now you can disable in the
extension and it will still auto-
answer
– The advanced setting overrides
the per extension settings

Intercom
Black and White listing Extensions
• This applies to intercom also which is what the auto-answer uses
• *557135 Disable extension 7135 from intercom to my extension.
– Make it so your softphone is not allowed to intercom your desk phone.
• Now trying calling, it will ring instead of auto-answer
– Now cancel it with *557135 again, and it should auto-answer again. Replace 7135 with your
softphone extension.
• Disable intercom for everyone on this phone *55, and try calling again, it rings instead of
auto-answer
• Now whitelist just your softphone so it can intercom your desk phone but other phones would
still not be able to *547135 and try calling. Replace 7135 with your softphone extension
number.
• Now cancel the whitelist *547135 and re-enable intercom to your desk phone for all device
by dialing *54

Intercom
Modify our intra-company trunk
• Dial prefix *80 for explicit intercom to remote extension. Now you can dial *80 and extension number on remote systems to
intercom
• Normal dial to always intercom remote extension
– Add a *80 prepend to the 4 digit intra-company route
– Intercoms the remote extension, their auto answer settings will not control the behavior

NAT
NAT and how to “trick” Asterisk
Enabling Remote NATed Phones
• How to overcome NAT with remote extensions
Step 1 - Behave as if no Firewall/NAT
– externip/externhost + localnet
– port forward required ports
Step 2 - Don’t believe phone’s RTP/SIP ports
– nat=yes
Step 3 - Keep SIP Port Open at Phone
– Enable Keep-Alive
or
– Set Registration timeout 30-60 seconds

NAT
Asterisk SIP Settings
• Usually Static IP or Dynamic IP
Router
• SIP and RTP port forwarding

NAT
Asterisk Extension/Device Settings
• nat: Yes
• Qualify=yes realities:
– Although this can help, if the far
side firewall pinhole closes up,
this will no longer be affective.
The Keep-Alive below is the
RIGHT solution
Phone
• Keep-Alive
• Short registration intervals if no
Keep-Alive available
Not required if keep-alive option
is available

DISA
DISA Overview
• Get you PBX dialtone from outside world.
– Call into DISA and get dialtone just like taking your desk phone receiver off the
hook.
– Dial Internal extensions or external numbers
– Outside calls show as coming from your PBX.
– Always set a password to protect your DISA.
– When would we use DISA
• Handy when you want to show a call from your PBX while traveling.
• Make normally expensive calls from your cell out your PBX so the PBX is making
the call not your Cell Phone such as calling international.

DISA
DISA Overview
DISA

DISA
Lab: DISA Setup
• Create a DISA with a Password.
• Go add a new hidden option in your IVR for DISA.
• Call into DISA from your Cell through your new IVR
option and make calls to other extensions and
outside numbers.
• Press ** to hangup a call and be presented with new
dialtone to start a new call.

Call Recording
Overview
• Complete re-write in FreePBX 12
• Allows much more granular control over Call Recordings
• Conflicts- Who Wins
• Call Recordings can be set at the following levels
– Inbound Route
– Outbound Route (in 2.11+)
– Extension
– Ring Group
– Conference Room
– Queue
– With the Call Recording Module anywhere in the Call Flow
– On demand with feature code in most scenarios
• How to view and listen to Call Recordings

Call Recording
Inbound Call Recording Hierarchy
Inbound Routes
• Inbound Routes have the highest level of priority for inbound calls. Options are:
– Force-
– Yes-
– Don’t Care-
– No-
– Never-
• 'Never' and 'Force' are overrides, and offer a higher priority than 'Yes' or 'No. Yes and No have an EQUAL
priority, and will not change what is already set, however 'Never' and 'Force' will always override what is
currently happening.
• If we set a yes here and later on in our call flow such as in a queue or extension we set a no the yes will
win since it was set first.
• If we set a Force here and later in the call flow we set No the Force would win but if we set a Never later in
the call flow the Never would win.

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