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.

Call Recording
Inbound Call Recording Hierarchy
Call Recording Call Flow instance
The Call recording module allows you insert call recording control at arbitrary points in the call flow then
continue on to any module. It functions exactly like the choice with Inbound Routes at the point it is
called but it comes after. A Yes or No previously set on the call would beat a Yes or No here but a Force
or Never here would override any Yes, No, Force or Never setting from any previous module.

Call Recording
Ring Groups, Queues and Conference Rooms
Ring Group, Queues and Conference Rooms
These modules also allow you to directly inside the module set your Call Recording options
but they come after any Inbound Route so just like the Call Recording Module A Yes or No
previously set on the call would beat a Yes or No here but a Force or Never here would
override any Yes, No, Force or Never setting from any previous module.

Call Recording
Extension Rules
Extensions
Call recording rules can be applied to record calls between extensions and to and from extensions
through trunks. Extensions are the last destination on a inbound call so they are the final stop.
– Inbound External- A call that came from outside the PBX
– Outbound External- A call made from an extension out through an Outbound Route
– Inbound Internal- An inbound call made directly from another extension dialing direct, not through a
Ring Group, Queue, etc.
– Outbound Internal- An outbound call made directly to another extension dialing direct, not through a
Ring Group, Queue, etc.
• A Yes or No previously set on the call would beat a Yes or No here but a Force or Never
here would override any Yes, No, Force or Never setting from any previous module.

Call Recording
Extension Rules
• On Demand- Whether to allow this extension to use the feature code to
start a recording on demand.
– Enable
– Disable
– Override
• When your call is in the status of 'Yes', 'No' or 'Don't Care', if the extension
is enabled for On Demand, they can start and stop the recording by
dialing the feature code.
• If the call is in 'Never' or 'Force', users can not stop or start recordings,
unless they have the 'Override' permission.

Call Recording
Extension Rules
Extension Recording Conflicts
When there is a conflict between two extensions calling each other as inbound calls, you must choose
‘who’ wins the decision whether or not to record the call. This is handled with the Record Priority Policy:
• Record Priority Policy- The extension with the highest priority dictates the recording
decision in a conflict.
• Handling Ties- If the priorities are equal, then we default to the Call Recording Policy as
specified in Advanced settings, allowing either the caller or callee to always win ties.

Call Recording
Outbound Call Recording Hierarchy
• Outbound routes can force recordings in the same way previously
discussed for inbound routes.
• The extension is the only other place a outbound call can be told to start.
If the Extension has set Yes or No it will beat the outbound route of Yes or
No.
• If the Extension has Force or Never set the Outbound Route setting a
Force or Never will beat the extension since it comes after the Extension
in the call flow.

Call Recording
Outbound Call Recording Hierarchy
• All Call Recordings are saved in /var/spool/asterisk/monitor directory on your system. Although this can be
changed in advanced settings, not all FreePBX recording features will continue to operate correctly.
• The recordings are then stored each day in a different folder organized as: year/month/day.
• There is a ‘loose’ naming convention for recorded calls that is based on how the call was initiated and
contains initial information about the callers involved. The actual call file name is stored in the
• CDR record.
• An example queue call is:
q-4600-6787890629-20130118-163430-1358534040.27732.wav
This indicates Queue 4600 recorded the call from phone number
678-789-0629 on January 18 2013 at 16:34:30. The unique asterisk ID is
1358534040.27732

Call Recording
How to View and Listen to Call Recordings
• You can view all call recordings in the CDR Reports module. Any call that was recorded will have a
download link in the column marked Recording.
• If the call was recorded by an extension direct they can view the Call recordings in the UCP under
Call History

Call Recording
How to View and Listen to Call Recordings
• There is a commercial module called Call Recording Reports, that lets an administrator view and
listen to all call recordings and setup auto archiving. This allows you to manage and archive older
recordings

Call Recording
LAB: Call Recordings
Create the following Call Recording rules
1. Record all queue calls
2. Record all Outbound Calls from your primary Extension only
3. Record all Emergency Calls
4. Go view your Call Recordings from the Call Recording
Report module and setup Archive of Call Recordings

Call Recording
LAB: 1- Record all Queue Calls
Record all Queue Calls
• We must change this in the Queue directly.
– As previously described, this can’t be done prior to the call flow, it must be
done in the Queue module itself.
• Set the Record Call option to “wav”

Call Recording
LAB: 2- Outbound Calls from Primary Extension
Record all Outbound Calls from your Primary Extension only
• Use the Extension Settings to control this since it is a per-
extension configuration.
• ! Now anytime we make a external outbound call from that
extension the call will be recorded.

Call Recording
LAB: 3- record all Emegency Calls
Record all Emergency Calls
• This can be achieved by recording all calls on a given outbound route, in this case
the Emergency Route.
• This is another example of why you may want to have multiple routes that use
identical trunks to another route. (e.g. record all International calls…)
• Set the record option on the Emergency Route to be Record Immediately or
Record on Answer

Call Recording
LAB: 4- View Call recording Reports
• View Call Recording Reports under the Reports Section
• From here you can see the Source, Date, Time and Duration of Recorded
Call. You can also listen to and delete any call recording.

Call Recording
LAB: 4- Archive Call Recordings
• Set Archive of Call recordings inside the Call Recordings Report module.
• Set how long to hold Call Recordings in Months. For example if we set this to 3 months on the first of each
month we will archive up all call recordings older then 3 months and send a email with a link to download
the archive. All archives are saved for 1 month before the archived voicemails are deleted if you do not
move them first.

Conference Rooms
Overview
Conferences:
• Asterisk Applications (controlled in Advanced Settings):
– app_meetme
– app_confbridge (This is the preferred application)
• Can be optionally controlled by a PIN code
– Conference Number: number to dial the conference bridge
– Conference Name: friendly name
– User PIN: Identifies a conference user
– Admin PIN: Identifies the caller as an admin (Leader) and
provides them will additional admin features when they dial “*”

Conference Rooms
Overview
Options controlling overall configurations
• Talker Optimization:
Asterisk detects silent users and ‘mutes’ them, reducing mixing load and avoiding background noise, useful on large conferences
• Talker Detection:
Tells Asterisk to include manager events identifying the current talker. Needed by external applications that require talker
identification or to see the talker in the CLI calls.
• Music on Hold (and Class):
Whether music should be played if waiting to join a conference and which music class to play.
• Allow Menu:
When yes, a menu of options will be presented to the user when they press “*” while in a conference.
• Record Conference:
Whether or not the conference should be reported.
• Maximum Participants:
Limit the number of conference users allowed into a conference

Conference Rooms
Overview
Options effecting joining and leaving a conference
• Join Message:
Play a message to user before entering conference
• Leader Wait:
If yes, users who enter before the Leader (Admin User) will not be able to talk to each other, usually hearing music, until
the Leader Joins.
• Quite Mode:
If yes, sounds will not be played when users enter and leave the conference
• User Count:
Announces the number of conference participants upon joining
• User join/leave:
Announces to everyone in the conference each time a user joins or leaves the conference
• Mute on Join:
Yes will mute each user when they join the conference.

Conference Rooms
LAB
Create a conference room:
• Require an Admin to join before others can hear
• Enable Music while waiting to enter
• Limit the attendees to 3 users
• Add the ability to dial into the conference from the IVR by direct dialing an access code, but
do this with the Directory Module
• Point your Inbound route to the IVR
• Now you should be able to call into the conference room, test to make sure that the user
who calls in gets music until the Admin calls in, and test what happens when a 4th user tries
to call into the conference.

Parking Pro
Overview
• Add Multiple Parking Lots to FreePBX
Park Pro adds the ability to add multiple parking lots within FreePBX, useful for companies
running multiple locations off the same server, or companies that need multiple parking lots
or have many internal departments.
• Park & Announce Feature
Automatically Announces Parked Calls to a Page Group. The Park and Announce feature
of Parking Pro allows you to set up and define & automatically announce when a call is
parked by paging a group of phones. Parked calls can be picked up by anyone on a
system. As an option this module allows the caller to leave a brief message to be played
during the page announcement. This can be useful for announcing the callers name or
other information requested from your caller. Parked Calls will then be announced to the a
paging group allowing anyone to pick-up the call at the announced extension.

Parking Pro
Discussion Park and Announce
• Configure new parking lot
• Create page group
• Park and Announce module configuration
• Setup destination on IVR to go to Park and
Announce

Parking Pro
LAB: Configure New Parking Lot

Parking Pro
LAB: Configure New Page Group

Parking Pro
LAB: Configure Park and Announce

Caller ID Management
Overview
• The CallerID Management module allows
you to change the outbound Caller ID on an
extension basis by dialing a feature code
that is setup to change the Caller ID inside
the Caller ID Management module.

Features
• Default is the Caller ID will only be used for that call.
• If Persistence is checked it will change the extensions caller ID permanently until a new feature code is used or you change
it in the extension GUI for that user.
• Dynamic Caller ID allows you to use a * as the Caller ID Num and now when you use the feature code you would dial
featurecode CallerID #NumbertoDial so it might look like *2399208869999#9209829999.
– Feature Code of *239
– Caller ID to be used id 9208869999
– Number to call is 9209829999

LAB
• Configure caller ID management entry and place outbound call to your
mobile device ( persistent unchecked )
• Make another call and the caller ID will return to your default Extension
Caller ID
• Change caller ID management by checking persistent – make outbound
call to mobile and it should be your new Caller ID
– Now log into the extension page in FreePBX for that device and the extension
Caller ID field should be updated to show the new Caller ID
• Setup a dynamic Caller ID and set your cell as the Caller ID.

Fax Pro
Overview
• Faxing in FreePBX allows users to
have inbound faxes sent to email. All
fax settings for a user are in the
extension module.
– This will be moving in FreePBX 13 to
be in User Management
• Allow users to send Faxes from UCP
• Allow users to view their received
and sent faxes in UCP
• Global Coversheet Template with
End User Overrides

Fax Pro
Lab Global Settings
• Set some global settings
– Fax Header- The name of the Fax
machine that is printed at the top of
all outgoing faxes
– Local Station Identifier- The Fax
Number that is printed at the top of all
outgoing faxes
– Outgoing Email- What emails
address inbound faxes should appear
to be sending from.
• Fax Transport Options
– Error Correction Mode- Set to No
when using VoIP
– Max Transfer Rate- Set to 9600 when
using VoIP
– Min Transfer Rate- Set to 2400

Fax Pro
Lab Global Settings
• Define Fax Coversheet info
such as
– Logo, Name, Address, Phone
Number and Footer
• Set Prefix of *323 to tell
SIPStation you are sending a
T38 Fax

Fax Pro
Lab Global Settings
• Set Retry attempts for failed Faxes
• Set email notifications on when to receive emails of fax attempts.

Fax Pro
Lab Outbound T38 Route
• Go create a new outbound for T38 Faxing. Call the trunk Fax.
– Add 2 dial rules of below to allow the *323 fax prefix plus any 10 digit or 11 digit number.
• *3231NXXNXXXXXX
• *323NXXNXXXXXX

Fax Pro
Lab Enable Fax for Extension
• Go edit your extension in
FreePBX for your Lab phone
– Enable Fax for this user
– Set email where inbound faxes
should point to
– Store the Fax Locally in UCP for
viewing also
– Optionally set override for
Header and Station ID. If not set
Globals will be used.
– Set Coversheet Name, Number
and Email address

Fax Pro
Lab Inbound Route Fax Detect
• Go edit your Inbound
Route.
– Detect Faxes set to Yes
– Set Fax Detect Type to
SIP
– Set Detection Time
somewhere between 5-6
seconds. This is the time
to try and detect a Fax
tone on new incoming call.
If Fax is detected Send
Fax to destination below.
– Pick Fax Destination and
pick your Lab Extension

Fax Pro
Make sure system is setup for T38
• Go edit Asterisk SIP Settings module
• Click on Chan SIP Option on the right menu
• Make sure T38 Pass-Through is set to Yes

Fax Pro
Recap
• Later on Today we will setup and use UCP to actually view,
send and manage our Fax for our user. The purpose of this
lab was to get everything for Faxing setup.
• Currently if you send a inbound fax to your DID the system
should detect the Fax and convert it to PDF and email it to
you and also store a local copy in the UCP.

Changing Your Menu Layout
freepbx_menu.conf
• Allows you to change the layout of the top menu bar.
• Allows for you to define custom layout the way you want display pages to
be categorized and displayed.
• In this file you can do the following things.
– Change the name of how the module pages are displayed.
– Change which category the page is displayed under
– Remove the page from being displayed in the menu
– Add a Favorite Category that is always displayed on the top left and pick
which pages are in this category
– Create a new Category and put the pages you want in this category

freepbx_menu.conf
• There is an example config file for this located at /etc/asterisk/ freepbx_menu.conf.template
to show you some examples of what can be done.
• Make sure that you enabled the Advanced Settings module option to enable supporting
freepbx_menu.conf or anything you do in this file will be ignored by FreePBX
• Create a file called freepbx_menu.conf in /etc/asterisk

freepbx_menu.conf
• The modules are located under
/var/www/html/admin/modules
• Most modules use the same page name as the
module’s directory name, but some have
multiple pages. You can see the page name of
a display page in the URL, for example
Outbound Routes:
• http://10.0.1.100/admin/config.php?display=ro
uting&extdisplay=3
• The page name will always be
display=pagename in the URL

freepbx_menu.conf
• You can find all the pages displayed by a
module, if any, in their module.xml file.
• Example core:
• Extensions: extensions
• Users: users
• Devices: devices
• Inbound Routes: did
• Outbound Routes: routing
• Trunks: trunks
• Advanced Settings: advancedsettings
• Administrators: ampusers
• FreePBX Support: wiki

LAB: freepbx_menu.conf layout change
• Change Asterisk SIP Settings to be SIP Settings.
– The display name is sipsettings so we will define that and then define what name we wan to show
up.
• Change which category the Languages page is displayed in from Applications to
the Settings category

• Remove DAHDIconfig page from being displayed since you have only SIP trunks
• Add a Favorite Category that is always displayed on the top left menu and move
Extensions and Follow Me there

• Create a Media Category and move Announcements, Music on Hold, Recordings to this
category
• Put dashboard into a new category called Status with nothing else in that category. When
there is only a single display page in a category, the top level menu item becomes it’s own
button.

• Once done the configuration file
should look similar to this and your
header bar in FreePBX should look
completely different once you
refresh your browser.

Contact Manager
Overview
• Allows you to create groups of contacts and share them with users.
• Users can then view the contacts from the Contacts Phone App and User Control Panel
• Create contacts of internal users or external contacts such as customers or vendors.
– Internal Group- Groups of internal users. Manually add any users into this group. All contact
information for each user added will be pulled from the users User Management settings. This is
used mainly to create groups of users like Support, Sales and such.
– External Group- Groups of external contacts such as customers and vendors.
– Default User Man Group- Like Internal Groups but this is a single group and will include all
users in this group that are told through User Management to be added to this group. Think of it
as a default group of all your internal users.

Contact Manager
LAB: Create a new Default User Manager Group
• We are going to first create a default User Manager group of all of our Users. Users are pulled
from the User Management module which is used for logging into UCP and other applications.
– The FreePBX Admin can set information on the internal user such as Name and different phone numbers
like home, mobile, work and such from User Management Modules.
– End Users can also change and set these numbers from inside UCP which we will go over later.
• Pick User Manager type
• Give it a name such as All Users and press Submit

Contact Manager
LAB: Create a new Default User Manager Group
• Now go to User Management and make sure your users are setup to be part of the Default User Manager
Contact group
• The difference between a User Manager group and Internal group is when you create a new User account
checking the “Show Contact in Contact manager” will auto add them to any User Manager Group in
Contact Manager.
• Creating a Internal only group in Contact Manager allows you to pick within Contact Manager which
contacts are part of that group so a good example would be creating a Support and Sales Group and only
adding Sales and Support Users to those group.

Contact Manager
LAB: Create a new External Contact Group
• You can also create external contact groups and include external contacts. For our lab
we will create a new Group called Vendors and add one or more Vendors into our
contact group.
• Press Add Entry option
• Add contact information

Contact Manager
LAB: Restrict which Contact Groups users can see
• Using Class of Service module you can restrict which Contact groups and
user can see in UCP or their Contacts Phone App.
• Now press the Contact button that we setup on your phone earlier and you
should be able to view and search contacts.
• Later in the UCP lab we will also view and manage contacts. In UCP users
can also create custom groups of contacts that only they will see.

Phone Apps (RestApps)
Overview
Phone Apps are a suite of phone applications that integrate directly with
FreePBX and our commercial End Point Manager.
Phone Apps allow users to control functions and settings directly from
the screen of their phone.
• Call Flow Control
• Call Forward
• Conference Rooms
• DND
• Follow Me
• Login/Logout
• Parking
• Presence
• Queues
• Queue Agents
• Time Conditions
• Transfer to Voicemail
• Visual Voicemail
• Contacts

EPM Setup
Go into EPM and under your template create XML-API button types and pick the
different Apps

Rest API Setup
• Make sure your End User is
setup to allow access to all
Modules for the FreePBX
API module in the User
Management module in
FreePBX under the Rest API
section

Rest API Setup
View our WiKi on how to use each of the app at
http://wiki.freepbx.org/display/FCM/RESTful+Phone+Apps-Apps+UserGuides
…Lets take 20 minute and let you play with the Apps now!!

Backup and Restore
Overview
• Discussion on options
• Create a local full backup
• Create a second backup to login to your partner’s system and
perform the backup and scp the backup to your system and
restore it. This is how you would setup a warm standby of
your primary system.

Backup and Restore
Discussion
• In FreePBX 2.10 the backup module was completely rewritten to make it more
reliable and flexible. It was designed to be very modular and has added some
confusion and complexity but more flexibility.
• There are 4 main sections to the new Backup Module
– Backups- These are your actual backup schedules that you setup.
– Restore- Here you can restore a backup from any of the servers where you have
backups stored as part of your PBX. You can also upload a backup file from your local
computer to restore from.
– Templates- These are actual templates of what files, directories and mysql tables to
include in your backup. We have defined some standard templates in the backup
module but you can create your own custom ones or edit the default ones included.
………Continues 

Backup and Restore
Discussion
– Servers- This option can cause confusion for people. Servers are either
servers that can be backed up or servers that backups can be stored on.
• Email- This is where you define an email address that you would send a backup to once it
completes.
• FTP- You can define an FTP server that backups are sent to once complete. You can also
restore from the FTP server direct from the Restore option now.
• Local- This is a local directory where backups will be stored and restored from.
• MySQL- This is where you define tables that you want to include in your backup file. This
would be used if you were going to add custom MySQL tables that are not part of normal
FreePBX that you want to include in your backup.
• SSH- This is where you would define another server to ssh to store your backup on or
another server such as a different PBX that you want to login to and perform a backup on.

Backup and Restore
Lab: Create a new backup
• Go create a new backup
• Give it a unique name- Daily
backup

Backup and Restore
Lab: Create backup
• Drag the Full Backup,
System Audio and Voice Mail
templates under the Backup
Items section
• You should see all items that
will be included in the backup
set

Backup and Restore
Lab: Create backup
• Make sure we are picking this server to backup
• Pick where to store the backup. By default it should show Local Storage and Legacy
Storage. Use local storage which is located in /var/spool/asterisk/backup. We could pick
more than one location to store the backup.

Backup and Restore
Lab: Create backup
• Pick how often to run this backup.
• You can pick the Custom setting to define more specific schedules.

Backup and Restore
Lab: Create backup
• Pick how long to keep the backup for.
• Now save the backup. Go scroll to the bottom and press the run now option to have the
system do a backup now.

Backup and Restore
Lab: Create backup
• SSH to your PBX now.
• Go into the directory your backups are stored in.
– Hint: if you picked Local Storage this location is: /var/spool/asterisk/backup/
• Verify the backup is there

Backup and Restore
Lab: Warm Standby
• We are now going to walk through having your system login to your partner’s
PBX each day to do a backup on their system and restore it to yours. This
would allow your system to be a warm standby system to your production
server.
• Determine which system is the Primary Server and which system is the Backup
Server with your lab partner.
• First thing we need to do is setup share keys for the asterisk user to be able to
access the primary server.
• SSH to your Backup System and use the following command to generate a
share key that will be transferred to the Primary Server (partner system) so it
can login to their system to do the backup.

Backup and Restore
Lab: Warm Standby
• sudo -u asterisk ssh-keygen
– You will see the following output. You will need to press Enter 3 times during this process.
• Next we will copy the key to the Primary server.(Partner Server)
– sudo -u asterisk ssh-copy-id -i /home/asterisk/.ssh/id_rsa.pub root@192.168.1.186. Replace the
IP with your partner’s IP.

Backup and Restore
Lab: Warm Standby
• We will now check to make sure the asterisk user can get into your primary server (partner
system)
– ssh -i /home/asterisk/.ssh/id_rsa root@192.168.1.186
• Replace 192.168.1.186 with your partner’s IP
• Type exit to go back into your box.
• We will now setup the backup on the Backup Server to go into the Primary Server (partner
system) and backup their settings and restore them on the Backup Server. (We will not
actually do the restore as we don’t want to lose all your settings of your PBX)

Backup and Restore
Lab: Warm Standby Backup
• First we need to add your partner’s
server as a SSH server in the backup
module under servers.
• Make sure to enter the path to your
private key so the system knows how
to load the key to your partners
system to login.

Backup and Restore
• Go create a new backup
• Give it a unique name- Warm Stanby

Backup and Restore
• Drag one of the templates to the Backup Items Area. Use the Safe Remote Restore Option
• You should see all items that will be included in the backup set
– Notice we are including just the config options which is all we need but we are excluding backup
settings as we would not want to restore the backup settings from the primary server or it would
override all the settings we have stored here.

Backup and Restore
• Make sure we are picking the Primary
Server we just setup as a Server. (For this
example we will not pick the Restore Here
option that way when it runs it won’t do the
actual restore but just move over the
backup file.
• Pick where to store the backup. We will
store the backup on our Backup Server
after it performs the backup on our Primary
Server. It will than scp the file here.

Backup and Restore
• Pick how often to run this backup.
• You can pick the custom setting to define a more specific schedule

Backup and Restore
• Pick how long to keep the backup for.
• Now save the backup. Go scroll to the bottom and press the run now option to
have the system do a backup now.

Backup and Restore
• We should now see it go through the backup and show a status

Linux System Admin-PBX Related
Overview
• Software Raid
• Notification of Asterisk Crashes
• Kernel Panic Auto Reboot
• Notification of Trunk Failures
• DNS Explained as it relates to Asterisk and SIP trunks
• Changing Asterisk Versions on your FreePBX Distro
• Logging into GUI without knowing Admin user or password.

Software Raid
• Disk Raid is an important factor to be considered when setting up a
PBX
• There are 2 options for Raid
– Hardware Raid
• Handled by your hardware and managed by the hardware
• Most efficient
• Most costly
– Software Raid
• Handled by the Linux operating system
• Not as efficient
• Part of Linux – no cost

Software Raid
• When setting up the FreePBX Distro, the first option under each Asterisk
version will attempt to setup software raid if multiple disks are detected.

Software Raid
• To Verify you have a software raid you would type of following command from the
Linux CLI.
– cat /proc/mdstat
• The output will be similar to the following if software raid is present, blank otherwise.
• In this example we can see all 3 partitions of the hard drive md0, md1 and md3 are
setup in a raid with disk sdb and sda

Software Raid
• If the Raid Array was in a down state meaning 1 or more hard drives had failed you would see something
similar to below.
• You will notice the [U_] showing that the second drive, sda in this example, has failed.
• To rebuild the raid array follow the guide at
http://wiki.freepbx.org/display/L1/Rebuilding+Software+Raid

Software Raid
• To setup email
notification to be
notified of a raid failure
log into the System
Admin module in
FreePBX and under
Storage you can define
the email address.

Notifications of Asterisk Crashes
• Since FreePBX Distro is setup to start asterisk with the safe_asterisk scripts we can tell
safe_asterisk to notify us anytime Asterisk crashes via email.
• We set this up in the /usr/sbin/safe_asterisk file.
– nano /usr/sbin/safe_asterisk
• Modify the NOTIFY section.
– Remove the # comment
– Replace the email address with your own

Notifications of Asterisk Crashes
• If you want to test this we can force asterisk to stop and have safe_asterisk kick in.
• We need to find the asterisk process ID so we can kill it.
– ps -ef| grep asterisk
– Find the process ID which in our example is 16093 and kill the process
– You should now get a email of the asterisk crash.

Kernel Panic Auto Reboot
• At times your Linux box may kernel panic. This is a rare occurrence and is almost always related to a
hardware issue. A kernel panic is like a Blue Screen of Death in the Windows world.
• The downside to a kernel panic is your box is locked up and the only way to recover is to reboot by cycling
power.
– This is not ideal if you are not close to where the box is located.
• We prefer to setup our boxes to auto reboot on a kernel panic. This is really important if you are doing
upgrades of the kernel as at time the box may kernel panic and you do not want to be locked out.
• We need to modify the following file
– /etc/sysctl.conf
• Add the following 2 lines to the bottom of the file

Notification of Trunk Failures
• FreePBX has the ability to call an external script to monitor a trunk. You can define the script on a per
trunk basis in the GUI of your trunks. (This field is hidden by default and exposed in Advanced Settings.)
• We have included a sample Trunk Monitor script that we will now install.
• SSH to your PBX and follow these steps
– wget -P /var/lib/asterisk/agi-bin/ -N http://ottstrunk.freepbx.org/otts/trunk-alert.agi
– amportal chown
• To make sure the ownership of the script is owned by the asterisk user and not root
• Now you can go into the GUI of your SIP trunk and define this custom monitor script:
– /var/lib/asterisk/agi-bin/trunk-alert.agi

DNS Explained as it relates to Asterisk
• FQDN with Asterisk and SIP trunks have always been issues for users.
– Asterisk has a long outstanding “BUG” in how its DNS Manager works related to it being a blocking
protocol.
– What happens is each time chan_sip needs to resolve a FQDN that is used anywhere with your SIP
setup it does a DNS lookup.
– Until this lookup resolves no other SIP packets are processed on the system.
– When this happens, Asterisk as a whole ‘bogs down’ and goes into somewhat of a ‘melt-down’ mode
where the system appears sluggish or locked up. Even Analog and PRI lines will be unusable on the
system.
• We highly recommend not using FQDN in your trunks or extensions or any peers in Asterisk.
If you must use a FQDN we recommend using a DNS server that you control on the local
network so if your internet goes down the PBX can still resolve DNS.

• The FreePBX Distro uses DNS Masq to handle this. Simply put 127.0.0.1
as the first DNS entry in /etc/resolv and that will tell the system to use the
local DNS server on the PBX.
• You can also set the 127.0.0.1 in the System Admin module in FreePBX
which will write out your /etc/resolv file.

• The FreePBX Distro has a script that lets you change between Asterisk 1.8, 10 and 11 at anytime
on the fly.
– When changing Asterisk versions Asterisk will be restarted for you and all active calls will be lost so only do
this when you have no calls.
– From the linux CLI type “asterisk-version-switch
– Pick which version of Asterisk you want to switch to.
– Once completed you should see a blank CLI screen.

Loggin into FreePBX GUI without password
• If you ever need to log into your FreePBX Admin GUI and either forgot the login username and
password you can unlock your login from the Linux CLI following these steps.
– Bring up the main login page to your FreePBX GUI.
– Do a “ctrl a” to highlight the whole page and look to the left side of the screen for some text. This is your
unique php session ID for your current session only. Copy this into your clipboard.

Loggin into FreePBX GUI without password
• Login into your Linux CLI and type of the following
commandreplacing the session ID with the one you copied from your
box.
• Go refresh your GUI login page and it should log you into FreePBX.
Now you can go reset the administrator password or create a new
user.

User Control Panel – UCP
Overview
• ARI replacement
• User centric
• Responsive design
• User Management Integration
– Login with the User Management User
and Password.

Features Overview
• Access to linked
extensions
• Designed to be the end
user interface to the
PBX and it’s features
• Desktop , Tablet , or
mobile device

LAB: Userman Setup
• Go into User Management module in FreePBX and make sure UCP is enabled for login.
• Make sure the user has XMPP set to yes
• It will make you set a UCP password when you enable XMPP. Set this password to be
anything you want. This is how you will login to UCP as this User.

LAB: Userman Setup
• Set the following Options on for UCP Permissions for this
User
– Allowed Settings
• Pick both your extensions
– Allowed CDRs
• Pick both your extensions
• Also allow for CDR Downloads and Playback of Recorded Calls.
– Allowed Conference Bridge
• Pick any conference rooms you want this user to view and manage from
UCP
– Allowed End Points
• Pick your Lab phone.
– Enable Presence
• Yes
– SIPStation SMS DID
• Make sure to pick a SIPStation DID for SMS so you can send and receive
SMS in UCP for your user.
– Allowed Voicemail
• Pick both your primary and softphone
– Enable WebRTC Phone
• Yes
– Allow Originating Calls
• Yes
• Login into UCP with the username and password

LAB: Gear
• Click on the Gear Icon in the top right menu
• From here we can pick to;
– Log Out of UCP
– Originate a call.
• This will place a call to any phone number or contact you
type in. It will call your primary extension and when you
answer it will place a call to the number or contact you
provided.
– Settings
• From here you can change your Userman Password and
change your contact information that is part of the internal
Contact Manager of FreePBX along with setting your
Language you want to view UCP in and enable Desktop
Notifications for things like new Voicemails, Faxes or SMS
messages.

LAB: Action Buttons
• Change status
– Your status change here will be linked
with the Presence Rest App on your
Lab phone
• Compose SMS for SIPStation
customers only
– Send SMS to your cell phone.

LAB: Action Buttons
• Make WebRTC Call
– Make a call to any phone number
• Send a XMPP Chat
– To any other user contected to the
XMPP Server of the PBX. This
could be any UCP user or any
user using a XMPP client

LAB: Call History
• ! Available for primary and any linked extensions

LAB: Conference Rooms
• Ability to see any of your Conference
Rooms you have permissions for.
• View callers in your Conference room
and Mute/Unmute or Kick any caller.

LAB: Conference Rooms
• Invite Callers into your Conference.
• You can type in any Contact Name or
External Phone Number. The number will
be dialed and when they answer they will
be transferred into your Conference Room.
If your Conference Room is setup with a
Pin Code invited callers will not need to
enter the Pin Code.

LAB: Contacts
• Contacts App will let you view any Contact Groups that your PBX admin has created and
given you permissions to view through the Contact Management Module and Class of
Service.
• You can also create your own Custom Contacts and Groups of Contacts.
• These same contacts can also be viewed on your Phone with the Contacts Phone App
• When sending a SMS, Fax, XMPP Chat, Originate Call or Inviting someone to a Conference
you can start typing any contact and it will give you a list of their numbers.

LAB: Device Manager
• If using EPM to manage your phones
you can allow your users through User
Management permissions to change the
button layout of their phones. For
example if they want to add or remove a
button you have setup in the EPM
Template they can change those settings
and it will only effect their phone not any
other users using the same template.
• Think of the settings as a user level
override.

LAB: Fax Pro Sending Faxes
• Send Faxes
• Type in any phone
number or name of
stored contact

• Optionally Include the system generated
cover sheet and define recipient
information in the cover sheet.
• My Name, My Telephone and My Email
are pulled from the Extension page for
this user. A user can change these before
sending the fax and also change the
stored information for them in the Settings
section of Fax in UCP.

• Upload 1 or more PDFs, or Tiff files and send the fax.

LAB: Fax Pro Reviewing Faxes
• Click on any of the Directories such as Incoming or Sent
• View or download the fax with the action icons.

LAB: Fax Pro Settings
• A user can at anytime change their Fax Settings by
clicking on the Settings tab in the Fax Section of
UCP
• These are the same settings that the Admin user in
FreePBX can set under the Extension for this user.

LAB: Fax Pro Desktop Notifications
• Desktop notification will be sent to the user if they have enabled them in the
Settings tab under the Gear in the top right corner of UCP if they are using Google
Chrome as their browser

LAB: Presence State Controls
• Status updates on login and logout/close
• Actions linked to status changes. Anytime your change
your status the action for that status will be set. For each
status you can tell the PBX to
– Not Change Anything
– Enable your Follow Me
– Enable Do Not Disturb.

LAB: Extension Settings
• Available for primary and
linked extensions in User
Management
• User control of options such as
Follow Me, Call Forward, DND,
Call Waiting. If you enable,
DND, Follow Me or Call
Forward here it should light up
the buttons on your phones if
you are using the Phone Apps.

LAB: SMS
• Enabled via Sipstation options
• Desktop notifications
• Persistent Chat box with history
– Clicking on the To Name will bring up the full SMS history until you delete it.

LAB: Voicemail Management
• Available for primary and any linked extension
• Desktop notifications

LAB: Voicemail Management
• Manage voicemail options
• Drag and drop greetings

VM Notify
Overview
• This module allows an individual or a group to be notified and optionally accept
responsibility for a voicemail. Additionally notifications can be sent out when
someone claims responsibility for the voicemail.
• Used primarily for After Hours Support or Emergency Notifications.
• Call a list of numbers in specific order like queue agents.
• When someone answers the call they will be notified a new voicemail has been
left in the Emergency Box or whatever message you record.
• The caller will be prompted to press 1 to listen to the voicemail.
• While listening to the voicemail the caller can press 1 to take responsibility for the
voicemail and no other users will be called about the voicemail.
• Email report will be mailed out showing the results of the notification upon a user
accepting responsibility or failed attempt to reach any user and include the
voicemail.

VM Notify
LAB: Setup VM Notify
• Navigate to the VM Notification module in FreePBX and create a new Notification.
• Pick your desktop phone as the voicemail box we want to link this notification with
and verify the notification is set to be enabled.

VM Notify
• Provide a list of numbers or extensions we want to dial. We can group numbers together into
groups just like queue agents by putting a penalty number at the end of each number.
• The VM notification system will first attempt to call all numbers with a penalty of 0, then 1 and so
forth. All numbers with the same priority number will be dialed at the same time so if you have 2
numbers with a penalty of 0, both of those numbers will be dialed at the same time.
• We can also setup Caller ID information to be used when dialing numbers.

VM Notify
• The default Initial Greeting informs the caller that a new voicemail has been left in mailbox XXX with XXX being the name
recorded on the voicemail box. The default instructions prompt the caller they can press 1 while listening to the voicemail to
take responsibility for the voicemail.
• Retry Count is how many times we should loop through the Caller List if nobody accepts responsibility for the voicemail.
• Retry Delay is now many minutes to wait after calling all the numbers in the Caller List before calling them again if nobody
took responsibility for the voicemail.
• Priority Delay is the amount of time in minutes to wait before trying the next group of Callers as defined by their penalty
number of 0 thru 99.

VM Notify
• Upon a caller taking responsibility for a voicemail or after all numbers have been called including the Retry Count without a
caller taking responsibility for a voicemail a email notification will be sent out based on the setting below.
– Email From- From address the email should be sent from.
– Email Success- Is the email address we should send the notification to upon a caller taking responsibility for a voicemail.
– Email Fail- Is the email address we should send the notification to upon calling all numbers including the Retry Count and no
caller took responsibility for a voicemail.
– Email Attach- When set to yes a copy of the voicemail file will be included in the email at a attachment.

VM Notify
• Email Subject- This is the subject line of the email that will be sent out.
• Email Body- This is the body of the email that will be sent out.
• Both of these fields include variables that are pulled in for each Voicemail
Notification including a full log of all numbers dialed and if they answered the call,
listened to the voicemail, took responsibility for the voicemail

DAHDI 101
Overview
• DAHDI is what Asterisk uses to connect to the legacy PSTN such
as analog or PRI lines.
• They are generally PSTN cards but can also be USB driven
• There are 2 main manufactures of cards.
– Digium Cards- These are cards manufactured by Digium. There are
also many knockoffs that copy their cards. Be careful with these
knockoffs.
– Sangoma Cards- They manufacture their own cards and provide to
you lots of additional tools with their wanpipe software.
• Wanpipe sits between the PSTN and DAHDI so it allows you to troubleshoot and see what is
happening with the PSTN signaling in front of DADHI and gives you more tools.

DAHDI 101
Overview
• DAHDI config files are stored in 2 primary locations.
– /etc/DAHDI/: This is where the configurations for DAHDI as it appears to the
outside world or PSTN lines
– /etc/asterisk/: This is where all configurations for asterisk and how it talks to
DAHDI are stored.
• When using a DAHDI card FreePBX expects one of two contexts:
– from-analog: This is used for any analog lines.
– from-digital: This is used mainly for PRI’s, BRI’s or other digital trunks that
send a DID.

DAHDI 101
Overview
• There are 2 main types of cards in the Market
– Digital.
• T1- Which can be for straight T1 or PRI style T1’s (North America)
• E1- Most of the Latin American and European Markets E1/PRI
ETSI/ISDN and MFCR2
– Analog- with either FXO or FXS Ports
• FXO Ports- Are used to connect to the incoming Phone Line
• FXS Ports- Are used to connect an analog phone as a extension into
your PBX

DAHDI 101
Overview
• DAHDI Groups. Everything with DAHDI trunks belong to a group.
– F or example we have have a 2 port PRI card. Port 1 we can setup as group 0 and port 2 can also be
group 0 or it could be its own group such as group 1.
– A group is simple way of grouping channels and PRI ports together into a shared pool used for
outbound calling.
– When you create a trunk in FreePBX you generally will pick which group the trunk sends calls out.
– A group could be a mix of PRI and Analog ports all in the same group if you wanted to, though that is
not typical.
– A common practice would be to have your PRI/T1 setup with Group 0 and your 2 Analog failover
lines as Group 1. Then in the trunk module of FreePBX you would create a PRI trunk that maps to
Group 0 and a Analog trunk that maps to Group 1.
– That way when creating outbound routes you first try Group 0 trunk then try Group 1 trunk.

DAHDI 101
FreePBX DAHDI Module
• Easiest way to setup your DAHDI cards is with the new FreePBX DAHDI module.
• This module currently supports Analog and T1/E1 cards from the following manufactures
– Digium
– Sangoma
– Allo
– OpenVox
– Rhino
• Since some of the DAHDI settings require a restart of the DAHDI service which is a linux level permission
there are times the module will inform you to reboot your system after changes.
– Instead of rebooting you can do the following commands
• amportal stop- Stops Asterisk
• service DAHDI restart- restart DAHDI. If you are using sangoma cards you need to do a service wanrouter restart which will also
restart DAHDI.
• amportal start- Starts Asterisk

DAHDI 101
• Since most of the systems in our lab do not have DAHDI
cards we will do this lab interactively.
– Global Settings- These are all the settings that go into /etc/
asterisk/chan_DAHDI.conf and apply globally to DAHDI from the
Asterisk side of things.
– Modprobe Settings- These are kernel driver specific settings for
each module driver DAHDI uses. Making changes here will
prompt you to reboot the system.
– Sangoma Settings- These are settings that are specific to
Sangoma Cards only.

DAHDI 101
• Digital Card Settings- Settings for the individual ports of a T1/E1 card.
• FXO Port Settings- Settings for each FXO port of an analog card.
• FXS Port Settings- Settings for each FXS port of an analog card.

DAHDI 101
• Creating DAHDI Trunk-
– Once we’ve setup a PRI or FXO port in the DAHDI module we can go to the trunk module in
FreePBX and create a DAHDI Trunk.
– At the bottom of the page we will see a drop down and it will let us pick from any group such as G0 or
individual analog ports on an analog card.
– In our example we can pick Group 0 Ascending or Group 0 Descending
– If we pick Ascending it will start with channel 1 and work up for each concurrent outbound call.
– If we pick Descending it will start with the highest channel which in our example would be 23 and
work down.

DAHDI 101
• Creating a DAHDI Extension
– Once we have setup 1 ore more FXS ports in the DAHDI module we can create a
DAHDI extension.
– Under Device Options choose the desired Channel to map this extension to the given
DAHDI channel.

Asterisk Log Files Settings
Overview
• Starting with FreePBX 2.11 you can set and control the log file setting for Asterisk.
• You can also view all the different Asterisk Log files from the GUI.
• Its important to understand how to read log files but we could spend 2 days just on
Asterisk Log Files.
– The point to this talk is to talk about how to setup and verify you have logs being written
and what the different options mean.
– Teach you how to setup a custom log file for specific things like DTMF debugging.
• Everything that happens in Asterisk can be logged to a log file.
• All logs are stored in /var/log/asterisk/ by default

Log Files Settings
• In the Asterisk Log Settings module in FreePBX you can set the following options.
– Date Format- This should generally not be changed or it will break things like Fail2ban. This is just how the Date
Timestamp is displayed in the log files.
– Log Rotation- When asterisk rotates the logs which by default is daily how do we want the file names of the rotated
logs to be displayed. By default your logs will rotate nightly and keep 7 days worth.
• Sequential- When rotating make the newest log file have the highest number such as full.7
• Rotate- When rotating make the oldest log file have the highest number such as full.7 This is the normal behavior in linux.
• Timestamp- Save the file with the date/time as the file name.
– Log Queues- Do you want Asterisk to create a queues log file. Lots of Queue reporting software's use this to generate
queue reports.

Log Files Settings
• Log Files
– By default you will see your system has a log file called full and you will see which options have been
enabled to log to the full log file.
• Debug- Used for debugging and most of the time can be ignored as they are for debugging purposes and will
generate a lot of noise.
• DTMF- Used to log every DTMF entry asterisk receives. This is useful if debugging DTMF issues to see if
Asterisk is receiving the DTMF or not.
• Error- Possible issues with dialplan created but not critical.
• Fax- Errors related to res_fax_asterisk to help debug fax issues.
• Notice- Message of a action like a Call being completed or a phone registering. Just informing you of actions.
• Verbose- Step by Step of the call flow. Used to debug or watch calls as they navigate your dialplan.
• Warning- Critical errors or issues usually and the most important one when debugging issues with Asterisk.
• Security- Security related events such as failed login attempts

Creating a DTMF Log File
• We are going to setup a log file just to capture all the DTMF that Asterisk receives
into its own log file.
• Create a new log file called DTMF and the only type of events we want to capture
is DTMF. Your setup should look like this.
• Press Save when done and Apply Configs.
• Go make a call to your IVR and then direct dial one of your extensions from the
IVR so we can get some DTMF logs generated.

Viewing Log Files
• To view the new logfile we just created go to the module in FreePBX under Reports called
Asterisk LogFiles
• At the top you will see a drop down of all the log files in Asterisk. Pick DTMF and 500 for the
last 500 log entries and press the Show button
• You should now see the entries from the call you just made. In my example We dialed 4002

Viewing Log Files
• We will see the DTM begin meaning it’s the beginning of the tone.
– Begin ‘4’ received on
• Next since we are just going into a IVR and not bridging channels it will
ignore the DTMF meaning not pass it to another channel
– Ignored ‘4’ on
• Then it will end the receiving of the DTM and finally end the pass-through.
– End ‘4’ received on end passthrough ‘4’
• Generally anything under 60MS will have issues with decoding the DTMF
the average DTMF is 80-300MS.

FreePBX High-Availability
HA Enables Automatic
Mirroring and Failover
Between 2 FreePBX Systems

High Availability (HA)
• Direct cost
• Additional work hours
• Lost work hours
• Lost revenue
• Regulatory compliance & risk
management

Why HA?
• Business Continuity

Historical Problems
• HA systems were built by hand
• Required the use of very
expensive sysadmins to design,
implement & constantly maintain
• Difficult to keep up-to-date
• Almost impossible to replicate or
support

What makes up FreePBX HA?
• FreePBX Distro
• Integrate DRBD, Cluster Manager
• Pacemaker

Automatic Mirroring and Failover
FreePBX HA Enables
Automatic
Mirroring & Failover
Between Two
FreePBX Systems
Automatic Mirroring
and Failover

Easy to Install

Select HA Install Option

Enter Location Specific Settings

Initial Configuration of FreePBX

Rinse and Repeat

HA hardware

Recommended Virtual Environments
• KVM
• Solus VM
• FreePBX Hosting

HA Installation

HA How it Works Overview

HA Management

Up time & Reliability
• Run different services
on both PBXs
• More resilient
• Quickly migrate
services when there’s
an issue

Astrisk is Down
• Displays an error
• Fail and attempt to
restart a set number
of times
• Before migrating the
Impacted services to
the other PBX

• Active node changed to
freepbx-b
• Calls are now being processed
by the secondary node
• Mysql and other FreePBX
services are still utilizing the
primary server

Knowing there is a problem
• SNMP Alerts
• SMTP Alerts (email notifications)

Keeping your System Updated

Todo lo lo que necesita saber para implementar FreePBX

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