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Lab Guide
Table of Contents
TASK 1:
LAB TOPOLOGY
TASK 2:
LAB OVERVIEW
TASK 3:
NETWORK SETUP
TASK 4:
TASK 5:
TASK 6:
ADVANCED FEATURES
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.
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
DHCP
DHCP
DHCP
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
10.1.10.1
255.255.255.252
10.1.10.2
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 2 (203) Emma Smith
VoiceMail (399)
4085xx1200
4085xx1201
4085xx1202
4085xx1203
4085xx1209
9060
952671800
9018182212462
90014085256800
spiad
spiad
spiad
spiad
admin
spiad
config terminal
hostname CISCO_SPIAD
ip dns server
username spiad privilege 15 secret spiad
ntp server 24.56.178.140
interface Vlan1
description DATA_VLAN
no ip address
interface Vlan100
description VOICE_VLAN
no ip address
interface GigabitEthernet 0/1
no ip address
no shutdown
interface GigabitEthernet0/1.1
description DATA_INTERFACE
encapsulation dot1Q 1 native
ip address 192.168.10.1 255.255.255.0
interface GigabitEthernet0/1.100
description VOICE_INTERFACE
encapsulation dot1Q 100
ip address 10.1.1.1 255.255.255.0
interface GigabitEthernet0/0
description WAN_INTERFACE
ip address dhcp
When login prompt appears, type spiad for user and spiad for password.
Load the following configuration that is needed to get Internal Service Module reachable:
config terminal
! 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
!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
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
Cisco Unity Express will be configured later in this lab at this point you have done with ISM
configuration.
Internet Connection
PSTN access will be provided via SIP trunk in this case internet connection is needed, this
section will provide a basic configuration to gain internet access for voice, data VLAN and
ISM deploying NAT.
config terminal
! Define NAT inside interfaces
interface GigabitEthernet0/1.1
ip nat inside
interface GigabitEthernet0/1.100
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
access-list 1 permit 10.1.1.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
! 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
Dont forget to save your configuration copying running configuration to startup
configuration.
Validate that you can get internet access, open your browser and go to:
http://www.cisco.com.
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
! 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.
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
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
! Enter the command create cnffiles in order to build the XML configuration files.
create cnf-files
end
This is a generic phone template that will be used by SCCP phones load it.
configure terminal
ephone-template 16
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
Dont forget to save your configuration copying running configuration to startup
configuration.
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.
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.
authenticate register
! Assign the voice mail extension 399 to the message button according to the Lab
Topology
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 Cisco Unified 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
allow-connections
allow-connections
allow-connections
allow-connections
h323 to h323
h323 to sip
sip to h323
sip to sip
! Disable SIP redirect response for call forwarding, Disable SIP REFER message for call
transfers and enable call-by-call detection of H.450 capabilities
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
! FAX relay configuration
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).
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
Once all the files were extracted check that files are on the directories that were created
previously.
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
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
tftp-server
end
copy running-config startup-config
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.
ephone-dn 2 dual-line
number 202 no-reg both
label 202
description Sara Noa
name Sara Noa
call-forward busy 399
call-forward noan 399 timeout 10
! 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
copy running-config startup-config
Connect your Cisco IP Phone 7975 to the switch, if your phone was upgraded you can see
on your console typing:
sh phone-load
Output shows something like this:
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.
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 7841 associated to Emma Smith,
replace id mac with the mac address of your phone
number 1 dn 1
! Set DTMF relay method
dtmf-relay rtp-nte
! set the username and password for this device
codec g711ulaw
! go to voice register global and create configuration files
4085xx1200
4085xx1201
4085xx1202
4085xx1203
4085xx1209
9060
952671800
9018182212462
90014085256800
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 ipv4:4.31.34.33:5060
sip-ua
authentication username 4085011200 password 4085011200
credentials username 4085011200 password 4085011200 realm
credentials username 4085011201 password 4085011200 realm
credentials username 4085011202 password 4085011200 realm
credentials username 4085011203 password 4085011200 realm
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:4.31.34.33:5060 expires 300
sip-server ipv4:4.31.34.33:5060
host-registrar
4.31.34.33
4.31.34.33
4.31.34.33
4.31.34.33
end
copy running start
In order to make and receive calls some translation rules and profiles are needed, follow
these steps to create translation rules for incoming and outgoing calls.
Allow sip server signaling IP address into the toll fraud prevention list
configure terminal
voice service voip
ip address trusted list
ipv4 4.31.34.33 255.255.255.255
end
copy running-config startup-config
Outgoing Calls Setup
configure terminal
! This service provider works only with codec G711u
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.
!
dial-peer voice 1021 voip
description **Local Calls - 7, 8 or 10 digits**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 9[1-9]T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip asymmetric payload full
dtmf-relay rtp-nte digit-drop
no vad
configure terminal
! Sent DID 40850xx200 to AutoAttendant extension 200, *AA will be ready later on the lab,
just configure translations
voice translation-rule 6
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/
Now that you can place and receive calls your PBX is setup for basic calls in the next
sections we will cover Voice Mail and Autattendant integration, additionally some advanced
features will be covered.
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
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 secret spiad
dn-webedit
time-webedit
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.
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
Enter the username and password that was created previously to get access to CCME GUI.
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.
Left the default values for the mailboxes and click next.
Click next.
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
Once system finish initialization you will see that voicemail boxes were created and
configuration was success.
Click logout.
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
By default we have created user and extension that forward calls to voicemail if there is no
answer.
Before to test voicemail and auto attendant, go to the next section to configure VOIP
routing to Cisco Unity Express.
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
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 399
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2001 voip
description ** Passthrough Inbound Calls for Internal Extensions from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ^...$
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2002 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 200
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2003 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 397
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
APPENDIX
A. Cisco Unity Express Factory Restart