Cisco Spiad

Cisco SPIAD
Hands on Lab
Communications Manager
Express Version 10.5
Table of Contents
TASK 1:
LAB TOPOLOGY
TASK 2:
LAB OVERVIEW
TASK 3:
NETWORK SETUP
TASK 4:
CONFIGURING CCME AS IP PBX SYSTEM ..
13
TASK 5:
VOICEMAIL & AUTO ATTENDANT SETUP
..
33
TASK 6:
ADVANCED FEATURES CONFIGURATION
..
45
Disclaimer
This lab is primarily intended to be a learning tool. In order to convey specific information,
the lab may not necessarily follow best practice recommendation at all times. This exercise
is intended to demonstrate one way to configure the network, servers and applications to
meet specified requirements for the lab environment. There are various ways that this can
be accomplished, depending on the situation and the customers goals/requirements. Please
ensure that you consult all current official Cisco documentation before proceeding with a
production/lab design or installation. By enrolling in this class or having access to this
document you acknowledge you are aware of this disclaimer and its implications.
2015 Cisco and/or its affiliates. All rights reserved. This document is Cisco Confidential. For Partners only. Not for distribution
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TASK 1: LAB TOPOLOGY
In the lab document, xx refers to your Pod number. For example, if you are seated at
POD01, you would replace all instances of xx in the lab documents with 01.
In order to get PSTN access a SIP trunk was set up for each POD, this is to emulate a PSTN
connection for the lab environment. Normally, you would plug trunks from the PSTN into the
FXO ports or T1/E1 port.
Hardware and software requirements:
Cisco SPIAD running CME 10.5
LAN Switch (2900, 300 or 500 series, Meraki MS)
Serial Console Cable
Cisco Phone 78xx, 79xx or any other supported model
Windows PC
Cisco IP Communicator Client
SSH client and terminal emulator (putty client)
TFTP server software
FTP Server software
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WAN IP addressing, Interface GigabitEthernet 0/0:

WAN IP Address
WAN Subnet Mask
WAN Default Router
DHCP
DHCP
DHCP
LAN IP addressing Interface GigabitEthernet 0/1:

LAN_DATA IP Address
LAN_DATA Subnet Mask
VLAN_DATA
192.168.10.1
255.255.255.0
VLAN 1
LAN_VOICE IP Address
LAN_VOICE Subnet Mask
VLAN_VOICE
10.1.1.1
255.255.255.0
VLAN 100
ISM Module0/0 IP addressing:

ISM IP Address
ISM Subnet Mask
ISM Default Router
10.1.10.1
255.255.255.252
10.1.10.2
SPIAD user name and password

Username
Password
spiad
spiad
Cisco Unity Express user name and password

Username
Password
spiad
spiad
Cisco Communications Manager Express GUI user name and password

Username
Password
admin
spiad
WiFi Connection
SSID
Password
SPIAD-PODx
Cisc0123
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TASK 2: LAB OVERVIEW

Lab Objective:
To gain experience configuring the Cisco SPIAD solution, configuring call processing, PBX
System, Voicemail, Auto Attendant, hunt groups, and additional features of the Cisco
Communications Manager Express
Audience and Prerequisites

This document is intended to assist solution architects, sales engineers, field engineers, and
consultants in learning many of the features of Cisco Communications Manager Express 10
running on a bundled system (CISCO SPIAD). This document assumes the reader has an
architectural and administrative understanding of the CCME and has reviewed the latest
administration guide.
Basic knowledge of how to install and administer CCME and CUE is recommended however
not necessary.
All configurations are based on Command Line Interface (CLI) so is highly recommendable
that you were familiar with the IOS commands.
Please refer to the following links to get further information about operation of Cisco
Communications Manager Express and Cisco Unity Express:
Cisco CME main page:
http://www.cisco.com/go/ccme
Cisco CME Install Guides:
http://www.cisco.com/c/en/us/support/unified-communications/unified-communicationsmanager-express/products-installation-guides-list.html
Cisco Unity Express:
http://www.cisco.com/go/cue
Cisco Unity Express Maintain and Operate Guides:
http://www.cisco.com/c/en/us/support/unified-communications/unity-express/productsmaintenance-guides-list.html
Cisco IOS compatibility matrix
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/requirements/guide/33matri
x.html
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TASK 3: NETWORK SETUP

Step 1: In order to get access to the SPIAD system a previous configuration for voice and
data VLAN was loaded, as well as DHCP pool for phones and users.
Step 2: Connect your laptop via WI-FI or with a cable to one of the switch ports and validate that you
get IP address via DHCP.
C:\> ipconfig
Step 3: Test connectivity to the default gateway assigned by SPIAD.

C:\> ping 192.168.10.1
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ISM Module Configuration

On Cisco SPIAD Cisco Unity Express is loaded from factor in ISM 0/0, Voice Mail and Auto
attendant Ports licenses are loaded also, so in order to get access to CUE IP addressing;
configuration is needed, this section describes the configuration procedure.
Step 4: Open SSH putty client and create a new session, use the IP address 192.168.10.1
click open and accept the security alert.
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Step 5: When login prompt appears, type spiad for user and spiad for password.
Step 6: A previous configuration was loaded in order to save time:

Step 7: Validate that you can reach ISM IP address and its default gateway from the router
pinging both IP addresses 10.1.10.1 and 10.1.10.2
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Step 8: Do the same tests, now form your PC.
Cisco Unity Express will be configured later in this lab at this point you have done with ISM
configuration.
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Internet Connection
Step 9: PSTN access will can be provided via SIP trunk, analog or digital PSTN trunks,
besides to use SPIAD as Internet gateway; this section will provide a basic configuration to
gain internet access for voice, data VLAN and ISM deploying enabling NAT for such
interfaces.
config terminal
! Define NAT inside interfaces
interface GigabitEthernet0/1.1
ip nat inside
ip nat inside
interface ISM0/0
ip nat inside
! Define NAT outside interfaces
interface GigabitEthernet0/0
ip nat outside
! Create an access list to allow Internet access to the following source subnets
access-list 1 permit 192.168.10.0 0.0.0.255
! Packets received on inside interfaces and permitted on access list 1 will be
translated to the IP address assigned to interface Gigabit Ethernet 0/0
ip nat inside source list 1 interface GigabitEthernet0/0 overload
end
Step 10: Dont forget to save your configuration copying running configuration to startup
configuration.
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Step 11: Validate that you can get internet access, open your browser and go to:
http://www.cisco.com.
Internet connection is done.
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TASK 4: CONFIGURING CCME AS IPPBX SYSTEM

In this section CCME will be configured as key system, we will see a couple of examples
when CCME is enabled to register Cisco Proprietary Skinny Client Control Protocol (SCCP)
phones and Session Initiation Protocol (SIP) phones.
The simplest model is the PBX model, in which most of the IP phones in your system have a
single unique extension number. Incoming PSTN calls are routed to a receptionist at an
attendant console or to an automated attendant. Phone users may be in separate offices or
be geographically separated and therefore often use the telephone to contact each other.
For this model, is recommended that you configure directory numbers as dual-lines so that
each button that appears on an IP phone can handle two concurrent calls. Dual-line
directory numbers enable your configuration to support call waiting, call transfer with
consultation, and three-party conferencing.
This section contains a list of the types of files that must be downloaded and installed in the
router flash memory to use with Cisco Unified CME. The files listed in this section are
included in zipped or tar archives that are downloaded from the Cisco Unified CME software
download website at:
https://software.cisco.com/download/type.html?mdfid=277641082
To get software a valid CCO ID is needed with and partner level privileges for such section.
Configure the Cisco Unified Communications Manager Express Parameters

Step 12: The following configuration besides to start the telephony service will allow SCCP
phones registration.
configure terminal
! Enter the command telephonyservice in order to enter telephone configuration mode.
telephonyservice
! Enter the command maxephones maxnumphones in order to set the maximum
number of IP phones to be supported by this platform.
max-ephones 30
! Enter the command maxdn maxdirectorynumbers in order to set the maximum
number of extensions that can exist in this platform.
max-dn 200
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! For security reasons enter the command no autoregephone in order to prevent the
connection of any phone to the system.
no auto-reg-ephone
! Enter the command load phonetype firmwarefile in order to identify the firmware
file that the IP phone uses to register in the system.
load 7975 SCCP75.9-2-1S

! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set
date format as dd-mm-yy
time-zone 9
time-format 24
date-format dd-mm-yy
! Assign the voice mail extension 399 according to the Lab Topology
voicemail 399
! Define the call forward and call park behavior
call-park system application

call-forward pattern .T
! Define the call transfer behavior allowing transfer to any number and call transfer
method
transfer-system full-consult
transfer-pattern 9.T
transfer-pattern .T
! Create another tone when you dial 9 to place an outside call.
secondary-dialtone 9
! Enter the command ip sourceaddress ipaddress in order to identify the IP address
and port number that the Cisco CallManager Express router uses for IP phone registration.
The default port is 2000.
ip source-address 10.1.1.1
! Set the interdigit timeout
timeouts interdigit 5
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! Enter the command create cnffiles in order to build the XML configuration files.
create cnf-files
end
Step 13: This is a generic phone template that will be used by SCCP phones load it.
configure terminal
ephone-template
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url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress

softkeys remote-in-use Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer TrnsfVM Confrn ConfList RmLstC Acct Park Select Join
Step 14: Dont forget to save your configuration copying running configuration to startup
configuration.
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Configure the Cisco Unified Communications Manager Express to allow SIP

phones registration.
Step 15: To enter voice register global configuration mode in order to set global
parameters for all supported Cisco SIP IP phones in a Cisco Unified CME or Cisco Unified
Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environment,
use the voice register global command in global configuration mode. To automatically
remove the existing DNs, pools, and global dialplan patterns, use the no form of this
command.
config terminal
voice register global
! Enables mode for provisioning SIP phones in Cisco Unified CME
mode cme
! Enter the command sourceaddress ipaddress in order to identify the IP address and
port number that the Cisco CallManager Express router uses for IP phone registration.
source-address 10.1.1.1 port 5060

! SIP Outbound Proxy wont be needed for SIP phones
no outbound-proxy
! Enter the command maxpool maxnumphones in order to set the maximum
! number of IP phones to be supported by this platform.
max-pool 30
! Enter the command maxdn maxdirectorynumbers in order to set the maximum
! number of extensions that can exist in this platform.
max-dn 200
! Enter the command load phonetype firmwarefile in order to identify the firmware
! file that the IP phone uses to register in the system .
load 7841 sip78xx.10-2-1-12

! Registration requests from SIP phones in a CME system must be authenticated
authenticate register
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! Assign the voice mail extension 399 to the message button
voicemail 399
! Set a repeating audible alert notification when a call is on hold on all supported SIP
phones directly connected in Cisco Unified CME
hold-alert
! Specify the directory to which the configuring files for SIP phones in CME are written
tftp-path flash:
! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set
! date format as dd-mm-yy
timezone 9
time-format 24
date-format D/M/Y
! Enter the command create profile in order to build the XML configuration files.
create profile
exit
Step 16: Configure voice service voip

! Start voice service voip
voice service voip

! allow only trusted hosts or network
ip address trusted list

ipv4 10.1.1.0 255.255.255.0
ipv4 192.168.10.0 255.255.255.0
ipv4 10.1.10.0 255.255.255.252
! Allow interworking or leg to leg communication
allow-connections
allow-connections
allow-connections
allow-connections
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h323 to h323
h323 to sip
sip to h323
sip to sip
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! FAX relay configuration
fax protocol none

no fax-relay sg3-to-g3
sip
! enable SIP registrar functionality
registrar server expires max 3600 min 120

no update-callerid
end
copy running-config startup-config
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CCME is now ready to receive SCCP or SIP phones registration, but phone loads needs to be
uploaded into the flash directory and define the TFTP alias for such loads, the ones needed
for this lab are for Cisco Phones 7842 (SIP) and 7975 (SCCP).
Note: In order to save time files were loaded to flash previously just validate that were
loaded on flash
(step 20).
Step 17: Start TFTP server and set

the root path as follows:
Phone loads compressed files for
Cisco Phones are:
7841 (SIP)
cmterm-78xx.10-2-1-12_REL.tar
7975 (SCCP)
cmterm-7975-sccp.9-2-1.tar
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Step 18: Open a SSH session to the SPIAD system if you are not logged in. And follow the
next procedure to upload and uncompress phones firmware files.
Create a directory to save the files

mkdir flash:phones
now create a directory for each model
mkdir flash:phones/7800
mkdir flash:phones/7975
validate that your directories were created
dir flash:phones
Step 19: Proceed to download and extract files from TFTP server
archive tar /xtract tftp://192.168.10.11/cmterm-78xx.10-2-1-12_REL.tar flash:/phones/7800
archive tar /xtract tftp://192.168.10.11/ cmterm-7975-sccp.9-2-1.tar flash:/phones/7975
Step 20: Once all the files were extracted check that files are on the directories that were
created previously.
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Step 21: Now create the TFTP alias for each file that was uploaded in order to make the file
reachable when phone request it.
configure terminal
ip tftp source-interface Loopback0
tftp-server
tftp-server
tftp-server
tftp-server
flash:/phones/7800/kern78xx.10-2-1-12.sbn alias kern78xx.10-2-1-12.sbn

flash:/phones/7800/rootfs78xx.10-2-1-12.sbn alias rootfs78xx.10-2-1-12.sbn
flash:/phones/7800/sboot78xx.10-2-1-12.sbn alias sboot78xx.10-2-1-12.sbn
flash:/phones/7800/sip78xx.10-2-1-12.loads alias sip78xx.10-2-1-12.loads
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
flash:/phones/7975/apps75.9-2-1TH1-13.sbn alias apps75.9-2-1TH1-13.sbn

flash:/phones/7975/cnu75.9-2-1TH1-13.sbn alias cnu75.9-2-1TH1-13.sbn
flash:/phones/7975/cvm75sccp.9-2-1TH1-13.sbn alias cvm75sccp.9-2-1TH1-13.sbn
flash:/phones/7975/dsp75.9-2-1TH1-13.sbn alias dsp75.9-2-1TH1-13.sbn
flash:/phones/7975/jar75sccp.9-2-1TH1-13.sbn alias jar75sccp.9-2-1TH1-13.sbn
flash:/phones/7975/SCCP75.9-2-1S.loads alias SCCP75.9-2-1S.loads
flash:/phones/7975/term75.default.loads alias term75.default.loads
end
Phone firmware is now loaded and ready to upgrade phones if it is necessary.

In case you have a phone on a pre-8.3.2 firmware and can't do a direct upgrade to 9.2.1.
You should load an interim firmware for instance 8.4.4 on your flash and upgrade it to that
load first. Then upgrade to 9.1.2.
Cisco Unified Communications Manager Express Phone Provisioning

In order to start to make and receive calls, you need to register the specific IP phones that
you want on the CCME system. In this process you set up individual ephonedns or
register-dns and then associate each with a button or buttons on one or more ephones.
Each ephonedn or register-dn is a virtual line, or extension, on which call connections can
be made. Each physical phone must be configured as an ephone or register pool in the Cisco
CME router in order to receive support in the LAN environment. With the use of the
ephonedn command and dualline keyword you create an ephonedn in dualline
mode. The reason is to have one voice port and two channels in order to handle two
independent calls. This mode enables call transfer, call waiting, and conference options.
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Registering Cisco SCCP phones to CCME

Step 22: This procedure allows registering ephones and ephonesdns to Cisco SPIAD:
configure terminal
! Create the ephone directory number with dual-line mode for Jim Smith, in case that
phone is busy or no answer call will be forwarded to voice mail extension.
ephone-dn 1 dual-line
number 201 no-reg both
label 201
description Jim Smith
name Jim Smith
call-forward busy 399
call-forward noan 399 timeout 10
! Create the ephone directory number with dual-line mode for Sara Noa, in case that
phone is busy or no answer call will be forwarded to voice mail extension.
label 202
description Sara Noa
name Sara Noa
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! Create the ephone for Jim Smith and assign the ephone-dn 1 to button 1
*replace mac-address for the one of your phone 7975 SCCP IP phone.
ephone 1
mac-address 0026.99EF.1DEB
type 7975
ephone-template 16
username "jsmith" password qwer201
button 1:1
! Create the ephone for Sara Noa and assign the ephone-dn 2 to button 1
! *replace mac-address for the one of your Cisco IP Communicator Client (CIPC)
ephone 2
mac-address AAAA.BBBB.0001
type CIPC
ephone-template 16
username "snoa" password qwer202
button 1:2
! Go to telephony-service an create CNF files
telephony-service
create cnf-files
end
Step 23: Connect your Cisco IP Phone 7975 to the switch, if your phone was upgraded you
can see on your console typing:
sh ephone phone-load
Output shows something like this (if phone is not connected nothing is showed)
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Cisco IP Communicator Client (CIPC) Configuration

Sara Noa has no a physical phone assigned, instead IT deparment has installed Cisco IP
communicator client which is a SCCP softphone that offers all features that you could find in
a Cisco phone 7975, an ephone and an ephone-dn have been configured previously.
IP Communicator Client is available on Cisco Download Software:
https://software.cisco.com/download/release.html?mdfid=278468661&catid=280789323&s
oftwareid=282074237&release=8.6(4)&relind=AVAILABLE&rellifecycle=&reltype=latest
Step 24: Once the software is installed, run the application and perform the audio test if
needed.
Step 25: On any place of the software interface right click and select menu preferences
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Step 26: On the new window opened go to network tab, click on use this device name
option, if you remember on the ephone 2 configuration, mac-address AAAA.BBBB.0001 was
assigned to this device for Sara Noa. The device name is composed by prefix SEP followed
by the mac address in this case is SEPAAAABBBB0001.
TFTP server is needed also to load CIPC parameters, TFTP server will be reached on IP
address 10.1.1.1.
Step 27: Click OK and wait until CIPC is reloaded, when finished you will see that it is
registered and extension 202 is assigned on button
one to Sara Noa.
Now Sara Noa should be able to call Jim Smith on
extension 201, and he can call her back to
extension 202.
Try a call from Jim to Sara and validate that basic
call between SCCP phones are working.
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Analog ports Configuration

Step 28: Follow next steps to get a FXS port (port 0/1/0) configured for a FAX Machine with
extension 210.
configure terminal
! Enable stcapp application
!
sccp local Loopback0

sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
!
stcapp ccm-group 1
stcapp
!
stcapp supplementary-services
port 0/1/0
fallback-dn 210
exit
exit
sccp ccm group 1

associate ccm 1 priority 1
!
voice-port 0/1/0
cptone MX
caller-id enable
!
dial-peer voice 1 pots
service stcapp
port 0/1/0
! Create the ephone directory number associated for FAX Machine, in case that phone is
busy or no answer call will be forwarded to Auto Attendant extension (200).
label 210
description FAX Machine
name Fax Machine
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! Find the mac address assigned to FXS port issue this command in configuration mode to
continue with configuration once you find mac address information:
do sh stcapp device summary
! From the output take the latest 12 characters to conform the mac address id associated to
the analog port 0/0/0 and continue with the configuration, create the ephone associated to
fxs port using phone type as anl which means that is an analog port
ephone 10
mac-address 6F24.CA5E.8000
username "faxmachine" password qwer210
type anl
max-calls-per-button 2
button 1:10
! issue again do sh stcapp device summary and check output now call state is
IDLE which means that analog port 0/0/0 is ready to place and receive calls,
exit from configuration mode and save your configuration.
end
Step 29: Connect an analog phone and test incoming and outgoing calls from and to
another extensions.
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FAX Considerations
Cisco SPIAD includes FXS ports that can be used for FAX, Point of sale or analog phones,
and some Service Providers may not offer such features over SIP trunk, in the next session
we will assume that SP supports protocol T.38 to send and receive fax.
Step 30: Follow this steps to enabling T.38 support over the SIP trunk
voice service voip
allow-connections sip to sip
fax protocol t38
end
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Registering Cisco SIP phones to CCME

Step 31: This procedure allows registering Cisco IP phones with SIP protocol into Cisco
SPIAD:
configure terminal
! Create Directory Number 203 associated to Emma Smith in case that no answer call will
! be forwarded to voice mail extension 399
voice register dn 1
number 203
call-forward b2bua noan 399 timeout 20
name Emma Smith
label Emma Smith
! Prevent that this DN attempts to register to external SIP proxy
no-reg
! Create the voice register pool for the cisco phone 78xx associated to Emma Smith,
! replace id mac with the mac address of your phone *check phone settings
voice register pool 1

id mac 2C3E.CF86.23C0
type 7821 (choose the one you have)
! Assign voice register dn 1 to button 1
number 1 dn 1
! Set DTMF relay method
dtmf-relay rtp-nte
! set the username and password for this device
username esmith password qwer203

! force codec to G711ulaw
codec g711ulaw
! go to voice register global and create configuration files

create profile
end
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Step 32: Connect your phone 78xx and place some calls to other extensions.
Enabling Video Calls (optional if you have video phones)

Step 33: In order to enable video calls between two IP phones an extra configuration is
needed for test purposes a couple of Cisco phones 8945 will be configured; follow the next
steps to get a video call.
! Create DNs 204 and 204 associated to George Thomas and Daniela Sanchez in case
that no answer call will be forwarded to voice mail extension 399
voice register dn 2
number 204
name George Tomas
label George Tomas
no-reg
voice register dn 3
number 205
name Daniela Sanchez
label Daniela Sanchez
no-reg
! Create the voice register pool for the cisco phone 8945 associated to George Thomas
and Daniela Sanchez, replace id mac with the mac address of your phone

id mac 0007.7d42.d4be
type 8945
number 1 dn 2
dtmf-relay rtp-nte
username gthomas password qwer204
codec g711ulaw
! The following command will enable video on the phone, otherwise only audio calls can be
placed
video
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id mac 0007.7d42.c1ec
type 8945
number 1 dn 3
dtmf-relay rtp-nte
username dsanchez password qwer205
codec g711ulaw
video

create profile
end
Step 34: Connect your 7845 IP phones and place a call between them, you should see
video in both phones.
Support for Cisco Jabber

Cisco Unified CME 8.6 and later versions support Cisco Jabber. The softphone SIP client is
an iPhone application and works as a SIP softphone. The SIP softphone client is capable of
supporting VoIP over WLAN. Cisco Unified CME 8.6 supports supplementary services such as
Hold, Resume, Transfer, Call Park, and Call Pickup for the softphone SIP client.
You can configure the softphone SIP client using the phone type CiscoJabber-iOS option.
Prerequisites
Cisco Unified CME 8.6 or a later version.
Restrictions
Conferencing feature through the Add Call action key is not supported.
Call hand off to the mobile network is not supported.
Shared line is not supported.
Step 35: These steps describes how to configure CME to get Cisco Jabber Client for iPhone
registered with Communications Manager Express. This device will be configured for
Daniela Sanchez (206)
conf terminal
voice register dn 4
number 206
name Daniela Sanchez
label Daniela Sanchez
no-reg
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Here's how to find your iPhone's MAC address:

1. Go to home screen, tap Settings.
2. Tap General.
3. Tap About. The window shown below appears.
The iPhone's MAC address - is the Wi-Fi Address in iOS settings
voice register pool
id mac E0C9.7AAB.F2A5
type CiscoMobile-iOS
session-transport tcp
number 1 dn 4
dtmf-relay rtp-nte
username dsanchez-jabber password qwer206
codec g711ulaw

create profile
end
Step 36: Configure Cisco Jabber Client for iPhone. First of all download Cisco Jabber Voice
from App Store.
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Step 37: Open Cisco Jabber Voice App and tap on Enter Account Settings to start set up
assistance to get this client registered.
After wizard starts tap on Begin, then Continue
In the next screen enter followin values:

Device ID: SEP<mobile MAC ADDRESS>
TFTP SERVER: 10.1.1.1
Enable Use Athentication Option
Username: dsanchez-jabber
Password: qwer206
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You will see that Jabber Voice is ready, tap on continue, now you are ready to place and
receive calls, you may be prompted to allow access to the iPhones microphone the first
time you make o receive a call.
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TASK 5: DEPLOYING VOICEMAIL AND AUTOATTENDAT

Cisco Unified Communications Manager Express provides call processing for Cisco Unified IP
Phones. Cisco Unity Express offers voicemail and automated-attendant capabilities for IP
phone users connected to Cisco Unified Communications Manager Express. The voicemail
and automated-attendant capabilities are fully integrated into the Cisco access router using
a network module or services-ready engine (SRE). With this solution, the Cisco portfolio of
access routers delivers features similar to those of a key system or hybrid private branch
exchange (PBX) plus the rich data and routing capabilities expected on the award winning
Cisco integrated services routers(ISRs), Customers can now deliver unified communications
to their small sites and branch offices with a solution that is very simple to deploy,
administer, and maintain. Cisco Unified Communications Manager Express with Cisco Unity
Express offers customers a cost-effective, highly reliable, feature-rich solution for an office
deployment.
As was described on Internal Service Module network setup; Cisco Unity Express is loaded
from factor in ISM 0/0, Voice Mail and Auto attendant Ports licenses are loaded also on
Cisco SPIAD.
Cisco Unity Express GUI can read users and extension from Communications Manager
Express in order to do it graphical user interface for CCME should be loaded and configured,
follow this procedure to download GUI files from TFTP server and apply the CLI commands
to enable http access and create user and password for this task.
CCME GUI setup
Step 44: Open a ssh connection to the SPIAD system, download and extract GUI files into
flash (Note: files have been loaded previously in order to save time go to step 45):
Issue:
archive tar /xtract tftp://192.168.10.11/gui.tar flash:
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version 1.3
Step 45: Apply the following commands to create the GUI administrative user and
password to grant http access for such user.
GUI username:
GUI Password:
admin
spiad
conf t
aaa new-model
!
!
aaa authentication login default local
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:/gui
!
file privilege 0
!
telephony-service
web admin system name admin password spiad
dn-webedit
time-webedit
Step 46: Open your browser and test CCME GUI access, go to
http://192.168.10.1/ccme.html
Type username: admin and password spiad
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version 1.3
After a successful authentication you should be able to see the administrative web interface
for Cisco Unified Communications Manager Express, from here you can create SCCP phones,
SCCP extensions and configure some telephony-service parameters.
Note: SIP phones and extensions cannot be managed on this interface.
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version 1.3
Voice Mail Message Waiting Indicator (MWI)

Step 47: As we said before, Cisco Unity Express GUI reads configuration from CCME in
order to have all parameters ready for CUE is necessary to create MWI configuration, follow
the next steps to create indicator for ON and OFF and dial-peer to get a proper
communication to CUE system.
! Create ephone-dn for ON and OFF indicators
ephone-dn 199
number A801... no-reg primary
mwi off
!
!
ephone-dn 200
number A800... no-reg primary
mwi on
! Create voice voip dial-peer to Passthrough Inbound Calls for MWI from CUE
dial-peer voice 1005 voip

description ** Passthrough Inbound Calls for MWI from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number A80T
dtmf-relay rtp-nte
codec g711ulaw
no vad
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version 1.3
Cisco Unity Express Setup

Step 48: With the previous steps done no we are ready to complete CUE setup.
Open your browser and go to http://10.1.10.1/ by default username and password are
cisco
When login, chose Communications Manager Express Integration.
This will trigger a system reload. If integration with Cisco Unified Communications Manager
Express is already selected go to next step.
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version 1.3
Wait for a few minutes until system goes up.
Step 49: Click on Run Initialization Wizard.
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version 1.3
Step 50: Enter the username and password that was created previously to get access to
CCME GUI (admin / spiad)
If you dont see the next button, click TAB several times until you see it and click on it.
CUE GUI has read users and extensions for CCME and can be imported; check the mailboxes
for the user to create a voice mail box, and choose jsmith as administrator, click next.
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version 1.3
Step 51: Left the default values for the mailboxes and click next.
40
version 1.3
Step 52: Enter following values to complete setup:

Voice Mail Number:
399
Voice Mail Operator Extension:
200
Auto Attendant Access Number:
200 (if no VM box call will be redirected to AA)
Auto Attendant Operator Extension:
201 (this is usually reception extension)
Administration via Telephone Number:
397
SIP MWI Notification Mechanism:
by default outcalling
MWI ON Number (Outcalling mechanism): read from ccme A800
MWI OFF Number (Outcalling mechanism): read from ccme A801
Click next.
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version 1.3
Step 53: Review the values that you just entered and commit your setup, checking the box
Finally, save to startup configuration (will take a few minutes more) and click finish
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version 1.3
Step 54: Once system finish initialization you will see that voicemail boxes were created
and configuration was success, take note about the PIN number is needed to get access to
the voice mail.
Click logout.
Creating Mailbox for a SIP account

Step 55: SIP users cannot be imported by CUE, so if you want to assign a voicemail box,
this has to be done manually, follow the next steps to add and create voice mail box for
Emma Smith (esmith) extension 203.
Login to CUE once again, use credentials username:
spiad password: spiad
Go to configure->users and click add.
Fill the new open window with the following mandatory
values
User ID:
esmith
First Name:
Emma
Last Name:
Smith
Nick Name:
(automatically filled with firstname lastname)
Display Name: (automatically filled with firstname lastname)
Associated Phone: none
Primary Extension: chose other and set it to 203
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version 1.3
Step 56: Left the rest of the parameters as default and scroll down to the button, mark the
check box to create mailbox
Click on the icon
located on the upper left corner to finishing user addition.
Step 57: As mailbox creation was selected in a

previous step a new window is showed:
Left default values and click on the add icon
Now the new user esmith was created and a mailbox
was associated to it.
You can validate that user was added going to
configure->users
Step 58: By default we have created user and extension that forward calls to voicemail if
there is no answer.
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version 1.3
Before to test voicemail and auto attendant, go to the next section to configure VOIP
routing for Cisco Unity Express.
Step 59: In order to get access to voicemail and basic auto attendant from SIP or SCCP
clients some dial peers are needed, add such dial-peers as follows:
conf t
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 399
b2bua
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
description ** Passthrough Inbound Calls for Internal Extensions from CUE **
b2bua
incoming called-number ^...$
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
b2bua
dtmf-relay rtp-nte
codec g711ulaw
no vad
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!
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
b2bua
dtmf-relay rtp-nte
codec g711ulaw
no vad
Step 60: Save your configuration, issue:
Testing Voice Mail and Auto Attendant

Step 61: Now you can test voicemail service dialing from extension 203 (Emma Smith) to
201 (Jim Smith), left a short message and check that message indicator is turned on once
you hang up.
Step 62: Retrieve your messages pressing the message button you will be prompt for a
password use the PIN that was created.
Step 63: Do the same test now dialing from Jim Smith (201) to Emma Smith (203).
To Validate Auto Attendant Operation dial from extension 202 (Sara Noa) to extension 200
you should be able to hear Auto Attendant prompt. Once you are connected chose 1 to dial
by telephone number and the dial 201 then # your call should be transferred to Jim Smith.
Repeat the test but this time dial to extension 203 then #
Whit these tests we have finished the Voice Mail and Basic Auto Attendant deployment, your
system is fully working as an IPPBX. You can place and receive call from SIP trunk directly
to a specific extension and to a basic auto attendant.
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version 1.3
TASK 6: ADVANCED FEATURES CONFIGURATION

In this section we will cover some advance features that can be enabled on Cisco SPIAD, we
will cover following implementation:
Hunt Groups
Call Pickup
Paging Groups
Call Park
Call restriction by user (pin before dial)
Extension Mobility
Meetme Conference
Single Number Reach
Voicemail to e-mail notification
Hunt Group
Call hunt allows you to use multiple directory numbers to provide coverage for a single
called number. You do this by assigning the same number to several primary or secondary
ephone-dns or by using wildcards in the number associated with the directory numbers.
In Cisco Unified CME, incoming calls search through the virtual dial peers that are
automatically created when you define directory numbers. These virtual dial peers are not
directly configurable; you must configure the directory number to control call hunt for
virtual dial peers.
Step 64: For this example we will assign a pilot number (303) to receive the call
associated to this group, then extension 201 and 203 will be members of this group in
case of any of those users answer the phone call will be forwarded to AutoAttendant (200).
Extension will be ring sequentially starting 201 if in 10 seconds is not answered call jumps
to the next extension.
Open a ssh session to SPIAD device an follow the next steps:
config term
voice hunt-group 4 sequential
final 200
list 201,203
timeout 10
pilot 303
end
Step 65: Test the described behavior calling to pilot number 303 from your Cisco
IPCommunicator Client.
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version 1.3
Call Pickup
Call Pickup allows a phone user to answer a call that is ringing on another phone. Since
Cisco Unified CME 7.1 introduces Call Pickup features for SIP phones. SCCP and SIP phones
support three types of Call Pickup:
Directed Call PickupCall pickup, explicit ringing extension. Any local phone user can pick
up a ringing call on another phone by pressing a soft key and then dialing the extension. A
phone user does not need to belong to a pickup group to use this method. The soft key that
the user presses, either GPickUp or PickUp, depends on your configuration.
Group Pickup, Different GroupCall pickup, explicit group ringing extension. A phone user
can answer a ringing phone in any pickup group by pressing the GPickUp soft key and then
dialing the pickup group number. If there is only one pickup group defined in the Cisco
Unified CME system, the phone user can pick up the call simply by pressing the GPickUp soft
key. A phone user does not need to belong to a pickup group to use this method.
Local Group PickupCall pickup, local group ringing extension. A phone user can pick up a
ringing call on another phone by pressing a soft key and then the asterisk (*) if both phones
are in the same pickup group. The soft key that the user presses, either GPickUp or PickUp,
depends on your configuration.
Step 66: For test purposes all three users (201, 202 and 203) will belong to the same
pickup group (33) configure the ephone-dn and voice register dn to be part of these pickup
group:
Configure tem
! For SCCP users
!
ephone-dn 1
pickup-group 33
!
exit
!
ephone-dn 2
pickup-group 33
!
exit
! For SIP user
!
voice register dn 1
pickup-group 33
!
end
!
write mem
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version 1.3
Step 67: Test Call Pickup, dialing from jabber client (ext 201) to 203 then from CIPC take
the ringing call pressing softkey
then
on your softphone.
Paging Group
A paging number can be defined to relay audio pages to a group of designated phones.
When a caller dials the paging number (ephone-dn), each idle IP phone that has been
configured with the paging number automatically answers using its speakerphone mode.
Displays on the phones that answer the page show the caller ID that has been set using the
name command under the paging ephone-dn. When the caller finishes speaking the
message and hangs up, the phones are returned to their idle states.
Audio paging provides a one-way voice path to the phones that have been designated to
receive paging. It does not have a press-to-answer option like the intercom feature. A
paging group is created using a dummy ephone-dn, known as the paging ephone-dn, that
can be associated with any number of local IP phones. The paging ephone-dn can be dialed
from anywhere, including on-net.
Restrictions for Paging:
Paging is not supported on IP phones without speakerphones.
Paging is not supported on Cisco Unified 3905 SIP IP phones.
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version 1.3
The paging mechanism supports audio distribution using IP multicast, replicated unicast,
and a mixture of both (so that multicast is used where possible, and unicast is used for
specific phones that cannot be reached using multicast).
Step 68: For this exercise we will create Paging Group Paging test and number 313, all
three users will belong to this paging group.
ephone-dn 99
number 313 no-reg primary
name Paging test
paging ip 239.0.1.20 port 2000
ephone 1
paging-dn 99
ephone 2
paging-dn 99 no-reg primary
paging-dn 99
end
write mem
Step 69: Test paging group calling for ext 206 to pilot 313, notice that call will be
connected and speaker on the remote phones will be opened, now you can ping the guy
who forgot the keys at desk front.
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version 1.3
Call Park
The Call Park feature allows a phone user to place a call on hold at a special extension so it
can be retrieved from any other phone in the system. A user parks the call at the extension,
known as the call-park slot, by pressing the Park soft key. Cisco Unified CME chooses the
next available call-park slot and displays that number on the phone. A user on another
phone can then retrieve the call by dialing the extension number of the call-park slot.
A reminder ring can be sent to the extension that parked the call by using the timeout
keyword with the park-slot command. The timeout keyword and argument set the interval
length during which the call-park reminder ring is timed out or inactive. If the timeout
keyword is not used, no reminder ring is sent to the extension that parked the call. The
number of timeout intervals and reminder rings are configured with the limit keyword and
argument. For example, a limit of 3 timeout intervals sends 2 reminder rings (interval 1,
ring 1, interval 2, ring 2, interval 3). The timeout and limit keywords and arguments also
set the maximum time that calls stay parked. For example, a timeout interval of 10 seconds
and a limit of 5 timeout intervals ( park-slot timeout 10 limit 5 ) will park calls for
approximately 50 seconds.
Step 70: Follow these steps to create a park slot for sales department where are
participating Jim Smith (201), Sara Noa (202) and Emma Smith (203)
Config term
telephony-service
call-park system application
ephone-dn 100
number 310
park-slot directed
park-slot reservation-group 1 timeout 60 limit 2 transfer 200
description park-slot for Sales
! assign park reservation group 1 to SCCP phones for: Jim Smith (ephone 1)
and Sara Noa (ephone 2)
ephone 1
park reservation-group 1
ephone 2
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version 1.3
! assign park reservation group 1 to SIP phones for: Emma Smith (voice
register pool 1)
! You may need to recreate XML phone configuration files
telephony-service
no create cnf-files
create cnf-files
ephone 1
reset
ephone 2
reset
no create profile
create profile
reset
end
write term
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version 1.3
Step 71: To test this feature call from Daniela Sanchez (206) to Sara Noa (202) and
answer the phone, click on the softkey more until you see the park option, press the
button to park, as you can see your call is now parked on slot 310, now you can run to
Emma Smith extension (203) and dial 310 to retrieve you call.
Call restriction by user (pin before dial)

This section describes the use of logical partition class of restriction (LPCOR) with Cisco
Unified Communications Manager Express (CME). The most common reason for the use of
LPCOR is the prevention of on-net calls from transfer or conference with a public switch
telephone network (PSTN) calls and vice versa.
For further details about LPCOR check following tech note:
http://www.cisco.com/c/en/us/support/docs/unified-communications/unifiedcommunications-manager-express/117880-config-cme-00.html
Step 72: In this section calls to mobile and long distance international dialing will be
blocked for Jim Smith (201) and Emma Smith (203), follow these steps to accomplish this
task.
Load audio files that will prompt for user and pin from your tftp server (previously loaded):
copy tftp://192.168.10.11/enter_pin.au flash:

copy tftp://192.168.10.11/enter_account.au flash:
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version 1.3
Step 73: Define parameters for the authorization package:
configure terminal
application
package auth
param max-retries 0
param passwd-prompt flash:enter_pin.au
param user-prompt flash:enter_account.au
param passwd 12345
param term-digit #
param abort-digit *
param max-digits 32
exit
exit
Step 74: Configure the AAA to force FAC for code and PIN.
aaa
aaa
aaa
aaa
aaa
new-model
authentication login default local
authentication login h323 local
authorization exec h323 local
authorization network h323 local
aaa session-id common

gw-accounting aaa
exit
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version 1.3
Step 75: Define the username and password.
username 8200 password 0 8200
Step 76: Define LPCOR policy.
voice lpcor enable

voice lpcor custom
group 10 user-LDI
group 20 trunk-LDI
exit
voice lpcor policy user-LDI
service fac
accept user-LDI fac
exit
voice lpcor policy trunk-LDI
service fac
accept user-LDI fac
exit
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version 1.3
Step 77: Associate an LPCOR policy with a device/resource. * some outgoing dial-peer were
loaded for test purposes.
! for SCCP phones
ephone 2
lpcor type local
lpcor incoming user-LDI
exit
! for SIP phones
lpcor type local
lpcor incoming user-LDI
exit
! for dial-peer
lpcor outgoing trunk-LDI
end
Step 78: Test LPCOR dialing to 90014085256800 from Sara Noa (IPComm), you should hear a
prompt asking for account number (username 8200) and pin number (password 8200).
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version 1.3
Extension Mobility
Provides the benefit of phone mobility for end users by enabling the user to log into any local Cisco
Unified IP Phone that is enabled for Extension Mobility
A user login service allows phone users to temporarily access a physical phone other than their own
phone and utilize their personal settings, such as directory number, speed-dial lists, and services, as if
the phone is their own desk phone. The phone user can make and receive calls on that phone using
the same personal directory number as is on their own desk phone.
Each Cisco Unified IP phone that is enabled for Extension Mobility is configured with a logout profile.
This profile determines the default appearance of a phone that is enabled for Extension Mobility when
there is no phone user logged into that phone. Minimally, the logout profile allows calls to emergency
services such as 911. A single logout profile can be applied to multiple phones.
Step 79: We will test Extension Mobility with Cisco IP Communicator Client (CIPC) assigning
extension 400 when it is idle, then we create a user that can login into the CIPC, this user is Kim Lee
with extension 222.
Configure extension mobility for SCCP phones
configure terminal
ip http server
telephony-service
url authentication http://10.1.1.1/CCMCIP/authenticate.asp emadmin empasswd
url authentication http://10.1.1.1/voiceview/authentication/authenticate.do emadmin empasswd
service phone webAccess 0

authentication credential emadmin empasswd
em keep-history
em logout 20:00 23:00
exit
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version 1.3
Step 80: Configure a Logout Profile for an IP Phone
! *If ephone-dn is no create, add it

ephone-dn 40
voice logout-profile 1
user commun400 password c400
number 400
pin 4004
privacy-button
exit
Step 81: Enable an IP Phone for Extension Mobility
ephone 11
mac-address AAAA.BBBB.0011
type CIPC
logout-profile 1
exit
Step 82: Configure a user profile for a phone user who logs into a Cisco Unified IP phone that is
enabled for Extension Mobility
! *If ephone-dn is no create, add it

ephone-dn 22
voice user-profile 11
pin 1421
user klee password klee222
number 222
speed-dial 1 201
max-idle-time 30
end
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version 1.3
Step 83: To test Extension Mobility, go to your PC and open CIPC as you can see it is
registered with user Sara Noa, right click over the client and go to preferences.
Change device name as SEPAAAABBBB0011 this will assign ephone 10 to this device after
the client restarts you will see that now it is registered with extension 400.
Step 84: Now click on the services button

select extension mobility and type the
username and password assigned for Kim Lee (user klee password klee222).
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version 1.3
After a success validation wait until you see that Kim Lee is now logged.
Extension mobility is now working as expected.

*if for some reason after follow steps you get a message on the ipphone showing this function is not enabled go ahead and save your configuration and
reload SPIAD.
Once extension
AAABBBB0001
60
mobility
works
set
the
IP
Communicator
MAC
Address
back
version 1.3
to
Conferencing
Meet-me
A meet-me conference is a hardware scheduled conference and takes place when the
conference creator goes offhook, presses the MeetMe soft key or feature button, and dials
the meet-me conference directory number (DN). Participants can then dial the meet-me DN
to join and connect to the conference bridge.
The phones display shows the meet-me DN as the remote party ID.
Meet-me conferences are straightforward. The conference creator explicitly chooses to make
a voice or video call by dialing a voice or video DN.
The only limiting factor for this solution is the number of T1 or E1 loopback ports and
digital-signal-processor (DSP) resources available.
DSP calculator is a good tool in order to provide an idea about how many DSP resources are
needed if you want to add more participants, do transcoding or add a set of VICs to your
SPIAD system, for further information about this tool go to:
http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html
Step 85: Follow these steps to create a meetme resource to have a maximum of 16
Participants Bridge.
! Create DSP farm

voice-card 0
dspfarm
dsp services dspfarm
! Create Join Tone
voice class custom-cptone jointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
! Create leave Tone
voice class custom-cptone leavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
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version 1.3
! enable SCCP application

sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
! Create dspfarm profile
dspfarm profile 1 conference
description conference profile-codec711
codec g711ulaw
codec g711alaw
maximum conference-participants 16
maximum sessions 1
conference-join custom-cptone jointone
conference-leave custom-cptone leavetone
associate application SCCP
no shutdown
! Setup a ccm group
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
! Configure telephony service
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
! Create DNs for conference 2 octo ephone-dn are needed to cover the
needed of 16 participants on meetme conference.
ephone-dn 170 octo-line
conference meetme
preference 1
no huntstop
conference meetme
preference 2
no huntstop
end
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version 1.3
Step 86: Test meetme feature hanging up your phone then on the softkeys menu click on
more until you see the key for meetme click on it and dial 770 which was the number
associated to ephone-dn 170 and 171 to start the conference, once is is started the rest of
the participants can join the bridge just dialing 770 if they are internal, if they are external
via auto attendant or a direct DID.
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version 1.3
Ad-hoc conference
For Cisco Unified CME 9.0 and later versions, Cisco Unified SIP IP phones act as ad-hoc
conference creators while Cisco Unified SIP or Cisco Unified SCCP IP phones act as the
participants.
Ad-hoc conference calls are unscheduled conferences and occur when the conference
creator adds a third party into the call. However, only consultative conferences, where the
creator commits after the consultative party is connected, are supported in these conference
calls. If the conference is configured to stay, the conference will fall back to a point-to-point
call and the conference bridge resource is released when participants leave the conference,
leaving only two parties.
Step 87: Follow these steps to enable Ad-hoc conferencing for SIP and SCCP phones
configure terminal
! for SCCP
telephony-service
conference hardware
exit
! for SIP
conference hardware
exit
! create a non-dial ephone-dn to handle SCCP ad-hoc conference
number C002 no-reg primary
conference ad-hoc
no huntstop
number C002 no-reg primary
conference ad-hoc
no huntstop
Step 88: Test this feature creating a 3 way conferencing from any phone call from phone A
to phone B, then put on hold phone B and dial phone C, now put all together; this can be
done using a soft-key or a physical key on the phone.
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version 1.3
Single Number Reach

The Single Number Reach (SNR) feature allows users to answer incoming calls to their
extension on either their desktop IP phone or at a remote destination, such as a mobile
phone. Users can pick up active calls on the desktop phone or the remote phone without
losing the connection. This enables callers to dial a single number to reach the phone user.
Calls that are not answered can be forwarded to voice mail.
Remote destinations may include the following devices:
Mobile (cellular) phones.

Smart phones.
IP phones not belonging to the same Cisco Unified CME router as the desktop phone.
Home phone numbers in the PSTN. Supported PSTN interfaces include PRI, BRI, SIP,
and FXO.
For incoming calls to the SNR extension, Cisco Unified CME rings the desktop IP phone first.
If the IP phone does not answer within the configured amount of time, it rings the
configured remote number while continuing to ring the IP phone. Unanswered calls are sent
to a configured voice-mail number.
There are some restrictions for SNR feature:
Each IP phone supports only one SNR directory number.
SNR feature is not supported for the following:
SCCP-controlled analog FXS phones
MLPP calls
Secure calls
Video calls
Hunt group directory numbers (voice or ephone)
MWI directory numbers
Trunk directory numbers
Step 89: Follow the next steps to enable single number reach, this shows extension 202 is
enabled for SNR on IP phone 2. After a call rings at this number for 5 seconds, the call also
rings at the remote number 4085550133. The call continues ringing on both phones for 15
seconds. If the call is not answered after a total of 20 seconds, the call no longer rings and
it is forwarded to the voice-mail number 2001.
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version 1.3
config term
pickup-group 33
label 202
mobility
snr 4085550133 delay 5 timeout 15 cfwd-noan 399
end
Single number reach can be enabled or disabled by user on his phone on idle or connected
state, as a ephone-template was defined, depending on the order that softkeys are
configured you will find the function mobility, clicking on more until you see such softkey.
This will enable or disable SNR to the configured number.
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version 1.3
Step 90: Modify the single number reach for a custom one, clicking on services icon, then
select my phone apps.
Then select Single Number Reach option and enter a custom number, then click submit an d
you are done, your new SNR is ready.
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version 1.3
Voice Mail to E-mail notification

In previous steps we set up Voicemail we can see on our phones if a message was left due
the action of MWI (message waiting indicator) this allow us to consult messages locally, but
a user can receive the notification on his e-mail box in case that his no available to check
directly on the phone and have access on smart phones, tablets, laptops, etc.
Follow these steps to enable email notification upon message arrives, using a Gmail
account.
Step 91: Open Cisco Unity Express web interface and longing with username spiad and
password spiad, on the left menu go to System then click on SMTP settings.
Step 92: Select none from the dropdown Import SMTP Settings menu.
External SMTP server: 200.79.160.7
Security Mode: None
Port: 45
Username: test@tetsz.com
Password: Cisc0SPIAD
Click Apply to save SMTP settings.
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version 1.3
Step 93: Go to Voice Mail menu and expand Message Notification, click on Notification
Administration, check enable system-wide notification for all messages, check attach
message to outgoing email notification and click Apply icon.
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version 1.3
Step 94: Now configure user settings to receive email notifications, go to configure, click on
users, in the right side you will see all users, click on snoa, go to the button of the profile
page and check Enable notification for this user/group. Go to the top of the page and
click on apply icon.
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version 1.3
Step 95: Open Notification tab and click on Email inbox.
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version 1.3
Step 96: Set following values on the new window:

-
72
Check Enable notification to this device,

To e-mail address: youremail@domain.com (this is the email address that will
receive notifications)
Notification Preference:
All Messages
Check Attach message to outgoing email notification
Set Notification Schedule from Monday to Friday between 8am and 6pm, click on add
button, then click apply icon and you will see a pop-up window saying saved
successfully click OK.
version 1.3
Now Sara Noa is ready to receive e-mail notifications. Leave her a message and check email
box.
CONGRATULATIONS!
You have finished this Lab.
73
version 1.3
APPENDIX
A. Cisco Unity Express Factory Restart
CISCO_SPIAD# service-module ism0/0 session
Trying 10.1.10.2, 2067 ... Open

se-10-1-10-1#
se-10-1-10-1# offline
!!!WARNING!!!: If you are going offline to do a backup, it is recommended
that you save the current running configuration using the 'write' command,
prior to going to the offline state.
Putting the system offline will disable management interfaces.
Are you sure you want to go offline?[confirm]
se-10-1-10-1(offline)# restore factory default
!!!WARNING!!!: This operation will cause all configuration and data
on the system to be erased. This operation is not reversible.
Do you wish to continue?[confirm]
Restoring the system. Please wait ...
..done
System will be restored to factory default when it reloads.
Press any key to reload:
System reloading ....
MONITOR SHUTDOWN...
INIT: Sending processes the TERM signal
Sending an RBCP message to IOS notifying module reboot...
Rebooting ...
SNIP
INIT: Entering runlevel: 2
********** rc.post_install ****************
IMPORTANT::
IMPORTANT::
Welcome to Cisco Systems Service Engine
IMPORTANT::
post installation configuration tool.
IMPORTANT::
IMPORTANT:: This is a one time process which will guide
IMPORTANT:: you through initial setup of your Service Engine.
IMPORTANT:: Once run, this process will have configured
IMPORTANT:: the system for your location.
IMPORTANT::
IMPORTANT:: If you do not wish to continue, the system will be halted
IMPORTANT:: so it can be safely removed from the router.
IMPORTANT::
Do you wish to start configuration now (y,n)?
74
version 1.3
Cisco IP Phone Factory Reset

Follow the steps below to successfully Factory reset your Cisco IP phone:
1. Unplug the power cable from the ip phone and then plug it back in.
2. While the phone is powering up, and before the Speaker button flashes on and off, press and
hold the hash # key.
3. Continue to hold # until each line button (right of the LCD screen) flashes on and off in sequence
in orange color.
4. Now release the hash # key and type the following sequence 123456789*0#
After the sequence has been entered the line buttons on the phone flash orange, then green and the
phone goes through the factory reset process. This process can take several minutes and the firmware of
the IP Phone will be erased.
When complete, the IP phone will reboot and the bootloader will try to obtain an IP address via DHCP.
The IP phone also expects the IP address (option 150) or the name (option 66) of the TFTP server to be
delivered by the DHCP server. This is why these DHCP options are critical at this phase.
The phone then tries to obtain the appropriate termXX.default.loads file depending in its model:
This "loads" file indicates all the files the IP phone has to download from the TFTP server to make up the
device firmware. The IP phone should first obtain the loads file and then proceed with the individual
files. Once complete, the IP phone will install the files and finally reboot
75
version 1.3
B. How to Reset Cisco IP Communicator

Follow the steps below to successfully clear settings on Cisco IP Communicator Client:
Press Settings Button (looks like a check in a box)
Press**# (to unlock menu)

Press Erase
76
version 1.3
C. E1 port configuration
This configuration is a sample to enable E1 port with R2 signaling for some carriers in Mexico, it will
enable the first 10 timeslots on the E1 port.
Please refer to DSP calculator to dimension the proper amount of DSPs, for conferencing, transcoding
and voice ports support.
http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html
network-clock-participate wic 3
network-clock-select 1 E1 0/3/0
trunk group ALL_T1E1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_T1E1
controller E1 0/3/0
framing NO-CRC4
ds0-group 0 timeslots 1-10 type r2-digital r2-compelled ani
cas-custom 0
country telmex (this makes reference for Mexico R2 signaling)
category 2
answer-signal group-b 1
caller-digits 4
dnis-digits min 4 max 4
dnis-complete
trunk-group ALL_T1E1 64
description TRONCALES DIGITALES
voice-port 0/3/0:0
cptone MX
! Create dial-peer as needed this shows dial-peer for international calls
dial-peer voice 66 pots
trunkgroup ALL_T1E1
description **Mexico*International Calls**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 3
destination-pattern 900T
forward-digits all
no sip-register
77
version 1.3
D. Configuring PSTN access via SIP trunk

In order to gain PSTN access, in this section we will see the procedure to get a SIP Trunk
registered against a simulated Internet Telephony Service Provider.
This Are the requirements and SIP trunk information.
Inbound SIP Trunk DID (PSTN Call in) Numbers (Use PSTN phone to test)
Auto attendant (200)
Extension 1 (201) Jim Smith
Extension 2 (202) Sara Noa
Extension 3 (203) Emma Smith
VoiceMail (399)
4085xx1200
4085xx1201
4085xx1202
4085xx1203
4085xx1209
Outbound Numbers to Call (From IP Phones on SPIAD to PSTN Phone)

Emergency Number
Local Call
Long Distance
International
9060
952671800
9018182212462
90014085256800
SIP Trunk information; remember XX corresponds to your POD number

SIP Register Server
SIP Outbound Proxy
Username
Password
4.31.34.33
4.31.34.33
4085xx1200
4085xx1200
In this case SIP Trunk requires authentication for each number, follow the next steps to get
registration from ITSP
configure terminal
voice service voip
sip
outbound-proxy ipv4:4.31.34.xx:5060
sip-ua
authentication username 4085011200 password 4085011200
credentials username 4085011200 password 4085011200 realm
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:4.31.34.xx:5060 expires 300
sip-server ipv4:4.31.34.xx:5060
host-registrar
78
4.31.34.xx
4.31.34.xx
4.31.34.xx
4.31.34.xx
version 1.3
end
copy running start
Validate that SIP trunk was successfully registered, issue:

show sip-ua register status
In order to make and receive calls some translation rules and profiles are needed, follow
these steps to create voice translation rules for incoming and outgoing calls.
For further information about voice translation rules check this link:
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voicetransla-rules.html
Allow sip server signaling IP address into the toll fraud prevention list
configure terminal
voice service voip
ip address trusted list
ipv4 4.31.34.xx 255.255.255.255
end
79
version 1.3
Outgoing Calls Setup

configure terminal
! This service provider works only with codec G711u
voice class codec 1

codec preference 1 g711ulaw
! Following translation rules remove 9 prefix and translate source extensions and any
number into the main DID 4085011200
voice translation-rule 410

rule 1 /^9$.*$/ /\1/
rule 15 /^...$/ /4085011200/
rule 15 /^.*/ /4085011200/
rule 1 /^9/ //
! Following profiles translate calling or called number according to translation rules defined,
these profiles will be applied later on voice dial-peers
voice translation-profile CALLER_ID_TRANSLATION_PROFILE

translate calling 1111
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
!
voice translation-profile PSTN_CallForwarding

translate redirect-target 410
translate redirect-called 410
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
Translation rules were created, the next step is to create voip outgoing voice dial-peers,
notice that session pattern includes outgoing prefix 9, session target is pointing to SIP
server that was previously configured on SIP trunk section, class-codec and dtmf relay
method may vary from carrier to carrier.
80
version 1.3
!
description **Local Calls 7 digits**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 9[1-9]
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte digit-drop
description ** International Calls**
preference 1
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
description **National Long Distance**
preference 1
destination-pattern 901..........
voice-class codec 1
no vad
!
description **CCA*Mexico*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
81
version 1.3
!
preference 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
end
write mem
Outgoing calls should be connecting now, test the numbers provided.
Incoming Calls Setup

To receive calls, voice translation rules and translation profiles are needed also; the next
procedure shows how to redirect an incoming call to an specific extension or application
(autoattendant).
configure terminal
! Sent DID 40850xx200 to AutoAttendant extension 200, *AA will be ready later on the lab,
just configure translations
rule 1 /4085011200/ /200/
voice translation-profile DID_AutoAtt
translate called 6
! Sent DIDs 40850xx201 to 203 to extensions 201 to 203
voice
rule
rule
rule
translation-rule 12
1 /4085011201/ /201/
2 /4085011202/ /202/
3 /4085011203/ /203/
voice translation-profile DIDs-SIP_trunk

translate called 12
! In order to get incoming calls properly routed add following voip dial-peers
82
version 1.3

permission term
description ** Incoming call from SIP trunk **
incoming called-number .%
voice-class codec 1
dtmf-relay rtp-nte
no vad
description ** DIDs from SIP_trunk **
translation-profile incoming DIDs-SIP_trunk
incoming called-number 408501120
voice-class codec 1
dtmf-relay rtp-nte
no vad
! dial-peer for incoming DIDs

description ** DIDs from SIP_trunk **
translation-profile incoming DIDs-SIP_trunk
incoming called-number 408501120[1-3]
voice-class codec 1
dtmf-relay rtp-nte
no vad
! AutoAttendant will be enabled later but this is the dial-peer needed

description DIDAutoAtt-AA
translation-profile incoming DID_AutoAtt
incoming called-number 4085011200
voice-class codec 1
dtmf-relay rtp-nte
no vad
83
version 1.3
E. Initial Configuration Setup for LAB

!
version 15.4
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CISCO_SPIAD
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
!
!
!
!
no service config
!
!
!
!
!
!
!
ip dhcp excluded-address 192.168.10.1 192.168.10.10
ip dhcp excluded-address 10.1.1.1 10.1.1.10
!
ip dhcp pool data
import all
network 192.168.10.0 255.255.255.0
default-router 192.168.10.1
dns-server 192.168.10.1
option 150 ip 10.1.1.1
!
ip dhcp pool phone
import all
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
84
version 1.3
!
!
!
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
cts logging verbose
!
!
voice-card 0
!
!
!
!
!
!
!
!
license udi pid SPIAD2911CME16F/K9 sn FTX1804AK0A
hw-module ism 0
!
hw-module pvdm 0/0
!
hw-module sm 1
!
!
!
username spiad privilege 15 secret 5 $1$cX0p$.CI83lMRJxN30J.VXWJGM0
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
85
version 1.3
no ip address
shutdown
!
description WAN_INTERFACE
ip address dhcp
duplex auto
speed auto
!
interface ISM0/0
no ip address
shutdown
!Application: CUE Running on ISM
!
no ip address
duplex auto
speed auto
!
description DATA_INTERFACE
encapsulation dot1Q 1 native
ip address 192.168.10.1 255.255.255.0
!
description VOICE_INTERFACE
encapsulation dot1Q 100
ip address 10.1.1.1 255.255.255.0
! Create a loopback interface
interface Loopback0
ip address 10.1.10.2 255.255.255.252
ip virtual-reassembly in
! according to the lab topology assign the ip address 10.1.10.1/30 and
default gateway 10.1.10.2
interface ISM0/0
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
86
version 1.3
!Application: CUE Running on ISM

service-module ip default-gateway 10.1.10.2
no shutdown
! add an static route in order to reach CUE via ISM0/0
ip route 10.1.10.1 255.255.255.255 ISM0/0
!
interface Vlan1
description DATA_VLAN
no ip address
!
interface Vlan100
description VOICE_VLAN
no ip address
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip dns server
!
!
!
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/2/0
!
voice-port 0/2/1
87
version 1.3
!
voice-port 0/2/2
!
voice-port 0/2/3
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
voice class codec 1
codec preference 1 g711ulaw
!
!
description **Local Calls - 7, 8 or 10 digits**
preference 1
destination-pattern 9[1-9]T
voice-class codec 1
voice-class sip asymmetric payload full
dtmf-relay rtp-nte digit-drop
!
description ** International Calls**
preference 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
88
version 1.3
!
description **National Long Distance**
preference 1
destination-pattern 901..........
voice-class codec 1
no vad
!
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
preference 1
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
!
!
gatekeeper
shutdown
!
!
sip-ua
89
version 1.3
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
host-registrar
!
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
privilege level 15
login local
transport input telnet ssh
!
scheduler allocate 20000 1000
ntp server 24.56.178.140
!
end
90
version 1.3

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CONFIGURING CCME AS IP PBX SYSTEM ..

VOICEMAIL & AUTO ATTENDANT SETUP

ADVANCED FEATURES CONFIGURATION

TASK 1: LAB TOPOLOGY

WAN IP addressing, Interface GigabitEthernet 0/0:

LAN IP addressing Interface GigabitEthernet 0/1:

ISM Module0/0 IP addressing:

SPIAD user name and password

Cisco Unity Express user name and password

Cisco Communications Manager Express GUI user name and password

TASK 2: LAB OVERVIEW

Audience and Prerequisites

TASK 3: NETWORK SETUP

Step 3: Test connectivity to the default gateway assigned by SPIAD.

ISM Module Configuration

Step 6: A previous configuration was loaded in order to save time:

Step 8: Do the same tests, now form your PC.

Internet connection is done.

TASK 4: CONFIGURING CCME AS IPPBX SYSTEM

Configure the Cisco Unified Communications Manager Express Parameters

load 7975 SCCP75.9-2-1S

call-park system application

url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress

Configure the Cisco Unified Communications Manager Express to allow SIP

source-address 10.1.1.1 port 5060

load 7841 sip78xx.10-2-1-12

! Assign the voice mail extension 399 to the message button

Step 16: Configure voice service voip

voice service voip

ip address trusted list

! FAX relay configuration

fax protocol none

! enable SIP registrar functionality

registrar server expires max 3600 min 120

Step 17: Start TFTP server and set

Create a directory to save the files

flash:/phones/7800/kern78xx.10-2-1-12.sbn alias kern78xx.10-2-1-12.sbn

flash:/phones/7975/apps75.9-2-1TH1-13.sbn alias apps75.9-2-1TH1-13.sbn

Phone firmware is now loaded and ready to upgrade phones if it is necessary.

Cisco Unified Communications Manager Express Phone Provisioning

Registering Cisco SCCP phones to CCME

Cisco IP Communicator Client (CIPC) Configuration

Analog ports Configuration

sccp local Loopback0

sccp ccm group 1

Registering Cisco SIP phones to CCME

voice register pool 1

username esmith password qwer203

voice register global

copy running-config startup-config

Enabling Video Calls (optional if you have video phones)

voice register pool 2

voice register pool 3

voice register global

Support for Cisco Jabber

Here's how to find your iPhone's MAC address:

voice register pool

voice register global

In the next screen enter followin values:

TASK 5: DEPLOYING VOICEMAIL AND AUTOATTENDAT

Voice Mail Message Waiting Indicator (MWI)

! Create ephone-dn for ON and OFF indicators

dial-peer voice 1005 voip