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INE Voice Workbook Volume II

Lab 4 Solutions

Table of Contents
Lab 4 ......................................................................................................................... 1
Task 4.1 - Network Infrastructure ............................................................................................................. 1
Task 4.2 - Network Infrastructure ............................................................................................................. 9
Task 4.3 CUCM Server and Phone Basics ........................................................................................... 13
Task 4.4 CUCM Server and Phone Basics ........................................................................................... 21
Task 4.5 CUCM Server and Phone Basics ........................................................................................... 34
Task 4.6 and 4.7 CUCM Media Resources ........................................................................................... 42
Task 4.8 CUCM Media Resources ........................................................................................................ 50
Task 4.9 CUCM Features ...................................................................................................................... 71
Task 4.10 CUCM Features .................................................................................................................... 82
Task 4.11- Gateways and Trunks ............................................................................................................ 86
Task 4.12 Gateways and Trunks ........................................................................................................... 93
Task 4.13 Gateways and Trunks ......................................................................................................... 100
Task 4.14 Inbound Calling from the PSTN......................................................................................... 107
Task 4.15 Inbound Calling from the PSTN......................................................................................... 110
Task 4.16 4.18 Inbound Calling from the PSTN................................................................................ 121
Task 4.19 Inbound Calling from the PSTN........................................................................................... 205
Task 4.20 Inbound Calling from the PSTN........................................................................................... 207
Task 4.21 Mobility ................................................................................................................................ 226
Task 4.22 Mobility ................................................................................................................................ 239
Task 4.23 Mobility ................................................................................................................................ 241
Task 4.24 High Availability .................................................................................................................. 247
Task 4.25 High Availability .................................................................................................................. 254
Task 4.26 Messaging ............................................................................................................................. 256
Task 4.27 Messaging ............................................................................................................................. 280
Task 4.28 Messaging ............................................................................................................................. 286
Task 4.29 Messaging ............................................................................................................................. 301
Task 4.30 Contact Center ...................................................................................................................... 307

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Lab 4
Task 4.1 - Network Infrastructure

Configure the CorpHQ to be the NTP master clock for the network all phones,
voicemail messages and voice gateways should ultimately be kept in sync with
the CorpHQ router, and every router should retain the proper time after being
rebooted but before NTP fully syncs up including the CorpHQ router
o All Devices at the CorpHQ site should use Pacific Time Zone (GMT -8)
and should follow Daylight Savings Time
o All Devices at the Branch1 site should use Central Time Zone (GMT -6)
and should follow Daylight Savings Time
o All Devices at the Branch2 site should use Central European Time Zone
(GMT +1) and should follow Daylight Savings Time
2 pts

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First configure the NTP master at CorpHQ router and then sync NTP with Branch1 and Branch2
routers.
Configure NTP master clock at CorpHQ Router:
!
ntp source Loopback0
ntp master 10
!

Sync Branch1 Router with NTP master clock:


!
ntp source Loopback0
ntp server 177.1.254.1
!

Sync Branch2 Router with NTP master clock:


!
ntp source Loopback0
ntp server 177.1.254.1
!

Now lets configure the timezone on the routers.


Configure timezone at CorpHQ Router:
!
clock timezone PST -8
clock summer-time PDT recurring
!

Configure timezone at Branch1 Router:


!
clock timezone CST -6
clock summer-time CDT recurring
!

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Configure timezone at Branch2 Router:


!
clock timezone CET +1
clock summer-time CEDT recurring
!

So now the network devices like routers maintain the time and timezone as we have configured
already. But the phones will follow the time from CUCM Publisher so we will also need to
configure NTP and site specific timezone parameters from Date/Time group in CUCM.

Navigate to OS Administration at CUCM Publisher:

Navigate to System> NTP Servers to set NTP server:

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Add new NTP server and save:

Now lets configure the DateTime/Group in CUCM. This is the first time we have navigated to
Cisco Unified CM Administration so we need to configure the Cisco Unified CM Group and
Phone NTP Reference first before we configure Date/Time Group.

Navigate to Cisco Unified CM Group:

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Configure the CM group and save:

Navigate to Phone NTP Reference from System>Phone NTP Reference:

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Create a Phone NTP Reference and save:

Now navigate to Date/Time Group from 'System>Date/Time Group':

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Create the Date/Time group for all the three sites and save:

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Task 4.2 - Network Infrastructure

Provision all sites phones with the following infrastructure configuration


o Ensure the phone-connected ports using 802.1Q trunking to provide a
Computer Data VLAN of 12, and a Phone Voice VLAN of 11
o Ensure that the port is specifically configured as a trunking-only port
o Ensure that phones are talking on the network as fast as possible
Provision each sites router to distribute IP addressing and options via DHCP for
that sites respective IP phones
o Subnet for CorpHQ Phones is 177.1.11.0/24
o Subnet for Branch1 Phones is 177.2.11.0/24
o Subnet for Branch2 Phones is 177.3.11.0/24
Ensure that there are no DHCP conflicts and that any that occur are logged
Ensure that all phones receive their proper TFTP server and choose an option
that will allow for FQDNs in the future
2pts

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First we need to configure the switchports where the phones are connected. The routers only
support 802.1Q vlan trunking protocol so the encapsulation dot1q command might not be visible
on the routers.
Configure switchports on CorpHQ-Switch:
!
interface FastEthernet0/10
description == Connection CorpHQ-Phone1 ==
switchport trunk encapsulation dot1q
switchport trunk native vlan 12
switchport mode trunk
switchport voice vlan 11
spanning-tree portfast
!
interface FastEthernet0/11
description == Connection CorpHQ-Phone2 ==
switchport trunk encapsulation dot1q
switchport trunk native vlan 12
switchport mode trunk
switchport voice vlan 11
spanning-tree portfast
!

Configure switchports on Branch1 Router:


!
interface FastEthernet0/1/3
description == Connection Branch1-Phone1 ==
switchport trunk native vlan 12
switchport mode trunk
switchport voice vlan 11
spanning-tree portfast
end
!

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Configure switchports on Branch2 Router:


!
interface FastEthernet0/10
description == Connection Branch2-Phone1 ==
switchport trunk native vlan 12
switchport mode trunk
switchport voice vlan 11
spanning-tree portfast
!
interface FastEthernet0/11
description == Connection Branch2-Phone2 ==
switchport trunk native vlan 12
switchport mode trunk
switchport voice vlan 11
spanning-tree portfast
!

Now configure the DHCP server on CorpHQ, Branch1 and Branch2 routers as instructed.
Configure DHCP server on CorpHQ Router:
!
ip dhcp pool VOICE_R1
network 177.1.11.0 255.255.255.0
default-router 177.1.11.1
option 66 ascii 177.1.10.10
!

Configure DHCP server on Branch1 Router:


!
ip dhcp pool VOICE_R2
network 177.2.11.0 255.255.255.0
default-router 177.2.11.1
option 66 ascii 177.1.10.10
!

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Also the Same on Branch2 Router:


!
ip dhcp pool VOICE_R3
network 177.3.11.0 255.255.255.0
default-router 177.3.11.1
option 66 ascii 177.1.10.10
!

Phones should get IP from the DHCP pool and should display Registration Rejected or
Unprovisioned. If you see this message then you have configured the switchports and DHCP
pool properly. The Registration Rejected and Unprovisioned message should be there as we
havent configured the phones in CUCM yet.

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Task 4.3 CUCM Server and Phone Basics

Register all IP phones (except the PSTN phone) to the CUCM server, ensuring
that the CUCM Publisher is the primary server registered to and that the
Subscriber can always take over as a CPE should failover occur, and ensure the
following stipulations and configurations are met:
o NOTE: This lab may require you to upgrade a phone with new firmware,
depending on where the last candidate in the rack left it (allow for
approximately 20 mins for all phones to complete their firmware upgrades)
o CorpHQ Phone1 should use DN 1001 and use SCCP firmware and be
allowed to place International calls
o CorpHQ Phone2 should use DN 1002 and use SIP firmware and be
allowed to place National calls
o Branch1 Phone1 should use DN 2001 and use SCCP firmware and be
allowed to place International calls
o Branch2 Phone1 should use DN 3001 and use SCCP firmware and be
allowed to place International calls
o Branch2 Phone2 should use DN 3002 and use SCCP firmware and be
allowed to place National calls
o When any phones place calls within any given site ensure that they should
use the G.711 codec; when any phones place calls between any two sites
ensure that they should use the G.729 codec
3pts

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We have already created the Unified CM Group where the Publisher is the primary call server
and Subscriber as the secondary. Now we shall create Region and Device Pool for each site and
also Partition and CSS to follow the calling permissions as instructed.
Navigate to Region from System > Region:

Create three Regions (as indicated) for each site:

As shown on the 'Audio Codec' field, set the desired audio codec using underside fields as
marked in the box. Do not create other regions now (i.e. R_Branch1_GW and R_MoH-Servers),
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we shall create those later when required with justifications. Setting the 'Video Call Bandwidth'
will also be discussed later on the following chapter. So let's create other two regions
R_Branch1_Phones and R_Branch2 along with codec settings like below:

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Navigate to Device Pool from System > Device Pool

Create Device Pool for each of the CorpHQ, Branch1 and Branch2 sites:

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Assign the Unified CM Group, Date/Time Group and Region in the Device Pool settings.

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Navigate to Partition configuration page from Call Routing > Class of Control >
Partition

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Create the following partitions :


PT_Internal_DNs
CSS_CorpHQ-INTL
CSS_CorpHQ-NATL
CSS_Branch1-INTL
CSS_Branch2-INTL
CSS_Branch2-NATL

Navigate to following Calling Search Space from Call Routing > Class of Control >
Calling Search Space:

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Create the following CSS with mentioned partitions included:


Calling Search Space
CSS_CorpHQ-INTL
CSS_CorpHQ-NATL
CSS_Branch1-INTL
CSS_Branch2-INTL
CSS_Branch2-NATL

Partition
PT_US-INTL
PT_US-NATL
PT_Internal_DNs
PT_US-NATL
PT_Internal_DNs
PT_US-INTL
PT_US-NATL
PT_Internal_DNs
PT_NL-INTL
PT_NL-NATL
PT_Internal_DNs
PT_NL-NATL
PT_Internal_DNs

Now you are good! You have created the device pool, you have created the partitions and
Calling Search Space also, and its time to register your phones!

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Task 4.4 CUCM Server and Phone Basics

Allow for maximum total of 5 G.729 calls coming from or going to the Branch1
site
Allow for maximum total of 7 G.729 calls coming from or going to the Branch2
site
Use RSVP to accomplish this task
3pts

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Our first task should be configuring the MTP on IOS (as dspfarm profile) at all three routers.
Then assign the RSVP bandwidth on the serial interfaces.
On the CorpHQ router:
!
voice-card 0
dsp services dspfarm
!
!
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 7.0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register CorpHQ-G729-MTP
!
dspfarm profile 2 mtp
codec g729r8
rsvp
maximum sessions software 25
associate application SCCP
!
!
!
!
interface Serial0/0/1:0
ip rsvp bandwidth
!
!
interface Serial0/0/1:0.1 point-to-point
description == FR To BR1
ip rsvp bandwidth 136
!
interface Serial0/0/1:0.2 point-to-point
description == FR To BR2
ip rsvp bandwidth 184
!

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On the Branch1 router:


!
voice-card 0
dsp services dspfarm
!
!
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 7.0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register Brnch1-G729-MTP
associate profile 1 register Brnch1-G711-MTP
!
!
dspfarm profile 2 mtp
codec g729r8
rsvp
maximum sessions software 25
associate application SCCP
!
!
interface Serial0/0/1:0
ip rsvp bandwidth
!
interface Serial0/0/1:0.1 point-to-point
description == FR To HQ
ip rsvp bandwidth 136
!

On the Branch2 router:


!
voice-card 0
dsp services dspfarm
!
!
sccp local Loopback0
sccp ccm 177.1.10.20 identifier 2 version 5.0.1

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sccp ccm 177.1.10.10 identifier 1 version 5.0.1


sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register Brnch2-G729-MTP
!
dspfarm profile 2 mtp
codec g729r8
rsvp
maximum sessions software 25
associate application SCCP
!
!
!
!
interface Serial0/0/1:0
ip rsvp bandwidth
!
interface Serial0/0/1:0.1 point-to-point
ip rsvp bandwidth 184
!

Register the MTP to UCM and then create MRG/MRGL for each of the CorpHQ and Branch1
sites.
Navigate to Media Resources > Media Termination Point

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Create new MTP with name 'CorpHQ-G729-MTP'

Add the MTP for Branch1 with name 'Brnch1-G729-MTP'

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And for the Branch1 add the MTP with name 'Brnch1-G729-MTP'

Now navigate to Media Resources > Media Resource Group

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Create three Media Resource Group for each CorpHQ, Branch1 and Branch2 sites:

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So Media Resource Groups (MRG) are created, now time to create Media Resource Group List
(MRGL) for each site.
Navigate to Media Resources > Media Resources Group List

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Create three MRGL for each of the sites:

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Create three Locations for all three sites and assign RSVP settings as mandatory; it will force
the calls to use the RSVP through IOS MTP.
Here is the Location for CorpHQ:

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The Location for Branch1:

And the Location for Branch2:

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Now assign all these Location settings in the Device Pools:

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Task 4.5 CUCM Server and Phone Basics

Ensure that all IP Phones show their PSTN DID number in the top right of their
display
o CorpHQ Phones should show this number: 206501100X
o Branch1 Phones should show this number: 512602200X
o Branch2 Phones should show this number: 020703300X
Associate users to their corresponding IP Phones using the below information:
o CorpHQ Phone1 belongs to Jack Shepherd (userid: jshepherd)
o CorpHQ Phone2 belongs to Hugo Reyes (userid: hreyes)
o Branch1 Phone1 belongs to Benjamin Linus (userid: blinus)
o Branch2 Phone1 belongs to Desmond Hume (userid: dhume)
o Branch2 Phone2 belongs to James Ford (userid: jford)
Ensure all IP Phones' primary line displays their User's First and Last Name
along with their extension (DN) in the following format:
o FName LName xYYYY (where YYYY is the 4 digit extension)
o (e.g. Jack Shepherd x1001)
Ensure two IP phones setting up a call (one phone dialing and the other phone
ringing) both display the full name (only FName LName) of the person that is
either calling or being called respectively
3pts

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So its time to register your phones! We have already created the basic stuffs like Device Pools,
Locations, Regions and even MRG/MRGL. What we are left only, creation of user ID, lets
create those first.
Create End User for each of the users from User Management > End User

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Click Save and create the users hreyes, blinus, dhume and jford on the same way.
Lets configure the phones with the all the display attributes as instructed.

Navigate to Devices > Phone

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Click 'Add New', then choose the 'Phone Type' and next the device protocol.

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For CorpHQ-Phone2 select the SIP as device protocol.


Here we have shown the detail configuration of CorpHQ Phone1

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Now configure the extention for CorpHQ Phone 1

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One of the phone configurations is shown; now configure the phones one by one.

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Task 4.6 and 4.7 CUCM Media Resources

Setup a Meet-Me Conference at the DN of 4010


You may only allow Jack or Desmond to setup this Meet-Me conference,
however any phone (up to a maximum of 8) should be able to dial into the
conference
If any member of the conference puts the conference on hold, MoH should not be
allowed to be played to the conference
3pts
Setup MTP, transcoding and conferencing where applicable and as necessary
Do not ever allow the usage of CUCM Software MTP or Conference Bridge
resources
3pts

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We need to configure the Conference Bridge first at both the CorpHQ and Branch2 routers.
At CorpHQ router:
!
dspfarm profile 4 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 1
associate application SCCP
no shut
!
!
sccp ccm group 1
associate profile 4 register CorpHQ-HW-Conf
!

Also on the Branch2 router:


!
dspfarm profile 4 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 1
associate application SCCP
no shut
!
!
sccp ccm group 1
associate profile 4 register Brnch2-HW-Conf
!

Restrict the number of conference participant from each of the CorpHQ and Branch2 routers:

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At CorpHQ router:
Branch2(config)#dspfarm profile 4 conference
Branch2(config-dspfarm-profile)#maximum conference-participants 8

Same on Branch2 router:


Branch2(config)#dspfarm profile 4 conference
Branch2(config-dspfarm-profile)#maximum conference-participants 8

Now register these conference bridges to CUCM.


Navigate to Media Resources > Conference Bridge

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Create Conference Bridge for CorpHQ site:

Create the Conference Bridge for Branch2 site:

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Associate the new created CorpHQ-HW-Conf into CorpHQ MRG:

Associate the new created Brnch2-HW-Conf into Branch2 MRG:

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Navigate to Device > Device Settings > Softkey Template:

Set a name for a new Softkey Template without MeetMe softkey:

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Navigate to 'Configure Softkey Layout'

Select the call state to "Off Hook" and then create the customized softkey and save
it:

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Apply this new softkey to all phones except for 'Jack Sheperd (1001)' and 'Desmond
Hume (3001)' and restart phones, below is the example for Hugo Reyes phone:

Follow the same procedure for all the other phones as instructed.
To disable MoH during conference on hold we need to recheck the following flag from the
Service Parameter, this value should be set to True to suppress the MOH.

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Task 4.8 CUCM Media Resources

Use the Publisher as everyones primary MoH server, and never allow the
Subscriber to act as a backup
Provision multicast MoH for the CorpHQ site phones and PSTN gateway using
the G.711 codec
Provision unicast MoH for the Branch1 site phonesusing the G.729 codec and
multicast MoH for the Branch1 PSTN gateway using the G.711 codec
o Ensure that no additional RSVP reservations are made when streaming
music to Branch1
Provision multicast MoH for the Branch2 site phones and PSTN gateway
o Stream all MoH traffic using the G.711 codecwithout any WAN bandwidth
3pts

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We need to create a separate Region, Location and Device Pool the for MoH servers, because the
MoH codec would be varying from site to site and the separate Device Pool for the MoH servers
would give us much flexibility to choose codec. Also we need separate Location as we dont
want additional RSVP reservation as instructed the new Location settings will ensure that.

Navigate to Systems > Region:

Create a new Region for the MoH servers and set the codec as instructed:

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Create a new Region for the Branch1 gateway, we shall need it while adding the
gateway for Branch1 site:

The gateway and phones are split on the Branch1 site because the codec requirement is different
for phones and gateway.
Navigate to Service Parameter for IP Voice Media Streaming Application for
177.1.10.10 (PUB):

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G.711 codec is selected by default, add the G.729 on the list by holding the Ctrl
button, select 'OK' for the warning appears:

Save the Service Parameter.


Navigate to Location from Systems > Location:

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Create a new location for the MoH servers:

Also make sure no RSVP reservation is set.

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Navigate to Systems > Device Pool

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Now create a new Device Pool with this new Region and Location set for MoH
servers:

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Also create another Device Pool for Branch1 gateway:

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To check it navigate to Media Resources > Music on Hold Audio Source:

Click on the Media Streaming Number ( 1 ), check if the 'Allow Multicasting' field is
enabled:

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Navigate to Media Resources > Music on Hold Servers:

Check if the servers are registered properly:

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Configure the Multicast parameter into each of the servers, below image is for PUB
MoH Server, don't forget to set the correct Device Pool:

And for the SUB MoH Server:

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Create a new MRG with PUB MOH Server for Multicasting only:

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Also create another MRG with PUB MoH Server but for Unicasting only:

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Assign the MRG into the CorpHQ MRGL to provide Multicasting as instructed:

Now the CorpHQ phones will get MoH at G.711 codec.

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Assign the MRG into the Branch1 MRGL (only for phones) to provide Unicasting as
instructed:

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Assign the MRG into the Branch1 MRGL (for gateway only) to provide
Multicasting as instructed:

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Assign the MRG into the Branch2 MRGL to provide Multicasting as instructed:

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Double check if the proper MRGL is selected on proper Device Pool, here is the
CorpHQ Device Pool:

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The Branch1 Phone Device Pool:

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The Branch1 Gateway Device Pool:

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The Branch2 Device Pool:

But we have instructed not to use WAN bandwidth for MoH, so we need to provide Branch2 site
MoH from the Branch2 router itself. On the Branch2 router:
!
telephony-service
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 177.3.11.1
!

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Task 4.9 CUCM Features

Enable whisper intercom functionality between Jack Shepherd's (x1001) and


Desmond Humes phones (x3001) using the DNs of 1001 and 3001 respectively
for those lines
Anytime Jack goes off-hook on his intercom line, it should automatically connect
to Desmonds intercom line, and vice-versa as well
Every other phone should be able to go off-hook and call to these two phone's
and trigger whisper intercom on them
3pts

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Configure the Intercom functionalities in the CUCM.


Navigate to Call Routing > Intercom > Intercom Route Partition

Create Intercom Partition first.

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Navigate to Call Routing > Intercom > Intercom Calling Search Space:

Intercom CSS will be created automatically.

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Navigate to Call Routing > Intercom > Intercom Directory Number

Now create Intercom DNs for each of the phone, for CorpHQ phones DN range
would be 1001 to 1002, for Branch1 phone range would be 2001 to 2001 and for
Branch2 phones range would be 3001 to 3002. Below is the example of CorpHQ
phones.

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So the DNs should be looking like this:

Now we need to assign these intercom lines to phones, so let's create a Phone Button
Template with intercom line added on 6th button. Navigate to Device > Device
Settings > Phone Button Template

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Click on 'Standard 7961 SCCP' template:

Click 'Copy' inside the template to create a duplicate template which is


customizable:

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Give a suitable name of the new template other than any default one.

Click on 'Save' Button; it will direct you to template configuration page.


Modify the sixth button as Intercom button:

Save the template.

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Now apply this template on each of the phones from phone device configuration
page, below is shown for 'Jack Shepherd' phone:

After saving the page the Intercom button would be appeared at left panel.

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Select on the left panel Intercom line, choose the respective intercom line (i.e. 1001
for Jack Shepherd, 3002 for James Ford etc). Here is the example for Jack
Shepherd's:

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Configure the same for all phones. Select the Speed Dial for only phones for Jack Shepherd (to
3001) and Desmond Hume (to 1001), just as instructed, when Jack Shepherd will press his
intercom line it would automatically connect to Desmond Humes.
Here is the screenshot for Desmond Hume:

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But do NOT assign Speed Dial for rest of the phones (i.e. for Hugo Reyes x1002,
Benjamin Linus x2001 and James Ford x3002), as intructed. Here is the example for
Benjamin Linus:

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Task 4.10 CUCM Features

Ensure that any IP Phone user can park any call that is active on their phone
Ensure that phones anywhere are always able to park calls anywhere in the DN
range of 5300-5499, and they all must be in the <None> Partition
No prefix should have to be dialed to retrieve calls from Park
3pts

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Navigate to Call Routing > Call Park

The term 'always' refers the situation even when PUB is down, so we have to define
the same number range on PUB and SUB by different wildcards, as we can't use
partitions. Here is for the range 5300 to 5399:

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The same number defined on SUB but with different wildcard:

Range 5400 to 5499 on the PUB:

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Same number range defined on SUB by different wildcard:

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Task 4.11- Gateways and Trunks


Provision the CorpHQ router (R1) as a gateway to the PSTN for CUCM using the
following specifications:
L1::T1::Linecoding::B8ZS
L1::T1::Framing::ESF
L1::T1::Time Source::LOCAL
L2/3::ISDN::PRI Switch-Type::National
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::SIP sourced from Loop0
o Ensure that if for some reason the Sub server doesnt respond within 1
second of a setup failure, that call setup is retried quickly to the Pub server
o Ensure that DTMF works properly over the link back to CUCM, but do not
configure any type that requires the use of a MTP
3pts

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Find the T1 slot/sub-slot:


CorpHQ#sh inven
NAME: "2811 chassis", DESCR: "2811 chassis"
PID: CISCO2811 , VID: V04 , SN: FTX1127A469
NAME: "Two port T1 voice interface daughtercard on Slot 0 SubSlot 0",
DESCR: "Two port T1 voice interface daughtercard"
PID: VWIC-2MFT-T1= , VID: 1.0, SN: 35817546

Here the controller is set on the router slot/sub-slot 0/0, the second port i.e. 0/0/1 is using for FR
so the unused first PRI port is 0/0/0.
Configure the ISDN switch type:
!
isdn switch-type primary-ni
!

Enable clock participation for the T1 controller, WIC 0:


!
network-clock-participate wic 0
!

Now configure the controller T1 0/0/0:


CorpHQ(config)#controller t1 0/0/0
CorpHQ(config-controller)#linecode b8zs
CorpHQ(config-controller)#framing esf
CorpHQ(config-controller)#clock source internal
CorpHQ(config-controller)#pri-group timeslots 1-3

It will automatically create the serial interface 0/0/0:23, the virtual voice interface. The following
console messages might appear:
%CONTROLLER-x-UPDOWN: Controller T1 0/0/0, changed state to up
%LINK-x-UPDOWN: Interface Serial0/0/0:23, changed state to up

Now configure the created serial interface to allow ie facility:


!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni

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isdn incoming-voice voice


isdn supp-service name calling
isdn negotiate-bchan resend-setup
isdn outgoing ie facility
no isdn outgoing display-ie
no cdp enable
!

Now configure the SIP parameters in the gateway:


!
sip-ua
timers trying 500
retry invite 1
!

SIP source interface should be configured now:


!
voice service voip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
!

Also configure the voice class codec now:


!
voice class codec
codec preference
codec preference
codec preference
codec preference
!

1
1
2
3
4

g711ulaw
g711alaw
g729r8
ilbc

As instructed the SUB should be primary server for all gateway to CUCM communications and
if SUB is down it should directed to PUB. Also we need to configure the DTMF type as it should
work for SIP as instructed. We need to configure this on the dial-peers; we shall create these
dial-peers now but would be modified later with destination-pattern when we shall configure the
dial-plans.

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!
dial-peer voice 101 voip
preference 1
voice class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
dtmf-relay sip-kpml rtp-nte
!
!
dial-peer voice 102 voip
preference 0
voice class codec 1
session protocol sipv2
session target ipv4:177.1.10.20
dtmf-relay sip-kpml rtp-nte
!

Its time to add the CorpHQ gateway on the CUCM, but its the SIP gateway so it wont be
added as gateway type it would be added as SIP trunk.
Navigate to Device > Trunk

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Click 'Add New' and then select 'SIP Trunk' as Trunk Type:

Now input the parameters correctly:

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Save the configuration.

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Task 4.12 Gateways and Trunks

Provision the Branch1 router (R2) as a gateway to the PSTN for CUCM using the
following specifications:
L1::T1::Linecoding::B8ZS
L1::T1::Framing::ESF
L1::T1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::NI
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::H.323 sourced from Loop0
o Ensure that if a WAN failure were to occur, all active calls to the PSTN
would remain up with RTP media
o Ensure that if for some reason the CUCM Sub server isn't responding fast
enough, that call setup is retried quickly to the CUCM Pub (within 2
seconds of CUCM Sub setup failure)
o Ensure that DTMF works properly over the link back to CUCM using a
method that carries not only the tone's frequency, but also the tone's
duration
3pts

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Find the T1 slot/sub-slot:


Branch1#sh inventory
NAME: "2811 chassis", DESCR: "2811 chassis"
PID: CISCO2811
, VID: V01 , SN: FTX0932A29T
NAME: "Two port T1 voice interface daughtercard on Slot 0 SubSlot 0",
DESCR: "Two port T1 voice interface daughtercard"
PID: VWIC-2MFT-T1=
, VID: 1.0, SN: 31850499

Here the controller is set on the router slot/sub-slot 0/0, the second port i.e. 0/0/1 is using for FR
so the unused first PRI port is 0/0/0.
Configure the ISDN switch type:
!
isdn switch-type primary-ni
!

Enable clock participation for the T1 controller, WIC 0:


!
network-clock-participate wic 0
!

Now configure the controller T1 0/0/0:


CorpHQ(config)#controller t1 0/0/0
CorpHQ(config-controller)#linecode b8zs
CorpHQ(config-controller)#framing esf
CorpHQ(config-controller)#clock source line
CorpHQ(config-controller)#pri-group timeslots 1-3

It will automatically create the serial interface 0/0/0:23, the virtual voice interface. The following
console messages might appear:
%CONTROLLER-x-UPDOWN: Controller T1 0/0/0, changed state to up
%LINK-x-UPDOWN: Interface Serial0/0/0:23, changed state to up

Now configure the created serial interface to allow ie facility:


!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni

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isdn incoming-voice voice


isdn supp-service name calling
isdn negotiate-bchan resend-setup
isdn outgoing ie facility
no isdn outgoing display-ie
no cdp enable
!

H.323 source interface should be configured now:


!
interface Loopback0
ip address 177.1.254.2 255.255.255.255
h323-gateway voip bind srcaddr 177.1.254.2
end
!

Configure the voice class codec now:


!
voice class codec
codec preference
codec preference
codec preference
codec preference
!

1
1
2
3
4

g711ulaw
g711alaw
g729r8
ilbc

Also the voice class h323 to configure the tcp establish parameters on failover:
!
voice class h323 1
h225 timeout tcp establish 2
h225 timeout setup 2
!
As instructed the SUB should be primary server for all gateway to CUCM communications and
if SUB is down it should directed to PUB. Also we need to configure the DTMF type as it should

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work for H.323 as instructed. We need to configure this on the dial-peers; we shall create these
dial-peers now but would be modified later with destination-pattern when we shall configure the
dial-plans.
!
dial-peer voice 101 voip
preference 1
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.10
dtmf-relay h245-signal
!
dial-peer voice 102 voip
preference 0
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.20
dtmf-relay h245-signal
!

To preserve the call over RTP media only even when the WAN failure occurred, without h225
negotiation, we have to configure the following command under h323 section as voice service
voip:
!
voice service voip
h323
h225 display-ie ccm-compatible
call preserve limit-media-detection
!

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Lets add this gateway (Branch1 gateway) on the CUCM as H.323 gateway.
Navigate to Device > Gateway

Click on 'Add New' and then choose the gateway type as 'H.323 Gateway'

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Input the correct parameters into the gateway and then save

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Reset the gateway after saving the configuration.

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Task 4.13 Gateways and Trunks

Provision the Branch2 router (R3) as a gateway to the PSTN for CUCM using the
following specifications:
L1::E1::Linecoding::HDB3
L1::E1::Framing::CRC4
L1::E1::Time Source::PSTN
L2/3::ISDN::PRI Switch-Type::EURO
L3::PRI::BChannels::3
L3::PRI::CNAM::Supported
L3::VoIP::H.323 sourced from Voice VLAN
o Ensure that if a WAN failure were to occur, all active calls to the PSTN
would remain up with RTP media
o Ensure that if for some reason the CUCM Sub server isn't responding fast
enough, that call setup is retried quickly to the CUCM Pub (within 2
seconds of CUCM Sub setup failure)
o Ensure that DTMF works properly over the link back to CUCM using a
method that carries not only the tone's frequency, but also the tone's
duration
3pts

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Find the E1 slot/sub-slot:


Branch2#sh inventory
NAME: "2811 chassis", DESCR: "2811 chassis"
PID: CISCO2811
, VID: V05 , SN: FTX1129F0GH
NAME: "Two port E1 voice interface daughtercard on Slot 0 SubSlot 0",
DESCR: "Two port E1 voice interface daughtercard"
PID: VWIC-2MFT-E1=
, VID: 1.0, SN: 31868050

Here the controller is set on the router slot/sub-slot 0/0, the second port i.e. 0/0/1 is using for FR
so the unused first PRI port is 0/0/0.
Configure the ISDN switch type:
!
isdn switch-type primary-net5
!

Enable clock participation for the T1 controller, WIC 0:


!
network-clock-participate wic 0
!

Now configure the controller T1 0/0/0:


CorpHQ(config)#controller e1 0/0/0
CorpHQ(config-controller)#linecode hdb3
CorpHQ(config-controller)#framing crc4
CorpHQ(config-controller)#clock source line
CorpHQ(config-controller)#pri-group timeslots 1-3

It will automatically create the serial interface 0/0/0:15, the virtual voice interface. The following
console messages might appear:
%CONTROLLER-x-UPDOWN: Controller E1 0/0/0, changed state to up
%LINK-x-UPDOWN: Interface Serial0/0/0:15, changed state to up

Now configure the created serial interface to allow display ie:


!
interface Serial0/0/0:15
no ip address
encapsulation hdlc

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isdn switch-type primary-net5


isdn incoming-voice voice
isdn outgoing display-ie
isdn outgoing ie redirecting-number
no cdp enable
!

H.323 source interface should be configured now:


!
interface Vlan11
description == Voice VLAN
ip address 177.3.11.1 255.255.255.0
h323-gateway voip bind srcaddr 177.3.11.1
end
!

Configure the voice class codec now:


!
voice class codec
codec preference
codec preference
codec preference
codec preference
!

1
1
2
3
4

g711ulaw
g711alaw
g729r8
ilbc

Also the voice class h323 to configure the tcp establish parameters on failover:
!
voice class h323 1
h225 timeout tcp establish 2
h225 timeout setup 2
!
As instructed the SUB should be primary server for all gateway to CUCM communications and
if SUB is down it should directed to PUB. Also we need to configure the DTMF type as it should
work for H.323 as instructed. We need to configure this on the dial-peers; we shall create these
dial-peers now but would be modified later with destination-pattern when we shall configure the
dial-plans.

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!
dial-peer voice 101 voip
preference 1
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.10
dtmf-relay h245-signal
!
dial-peer voice 102 voip
preference 0
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.20
dtmf-relay h245-signal
!

To preserve the call over RTP media only even when the WAN failure occurred, without h225
negotiation, we have to configure the following command under h323 section as voice service
voip:
!
voice service voip
h323
h225 display-ie ccm-compatible
call preserve limit-media-detection
!

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Now add this gateway on CUCM:


Navigate to Device > Gateway

Click 'Add New' and select 'H.323 Gateway' as gateway type:

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Now input the correct parameters to configure the gateway:

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Save the gateway and reset it.

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Task 4.14 Inbound Calling from the PSTN

Globalize all calls coming inbound from the PSTN to gateways at all sites using
the proper Full E.164 numbering format (including preceding +) for each site
The preceding 0 coming into the Branch2 Amsterdam site from the PSTN
should not be included in the globalized format of the number - drop this "0"
before doing anything else to the number
The new Globalized Calling number should display at every hardwareIP phone
when the user at any phone looks at the Call History
3pts

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Globalize CorpHQ site inbound calls


Globalization can be configured on the CUCM at H.323 gateways at the Incoming Calls section,
but what about the SIP trunks? We have the SIP trunk for CorpHQ site and the SIP trunk doesnt
include with Incoming Calling Party Settings, but we can globalize the call at the CorpHQ
router by voice translation rule.
!
voice translation-rule 1
rule 1 // /+1/ type subscriber subscriber
rule 2 // /+1/ type national national
rule 3 // /+/ type international international
!
voice translation-profile Globalize_Inbound
translate calling 1
!
dial-peer voice 1 pots
translation-profile incoming Globalize_Inbound
incoming called-number .
direct-inward-dial
!

This rule will globalize the calling number to full E.164 numbering format and deliver to
CUCM, so we dont need to configure on the CUCM for this requirement.

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Globalize Branch1 site inbound calls


As CUCM is receiving calls from Branch1 site from H.323 gateway, we have chance to
configure the globalization option at the incoming gateway at the Incoming Calling Party
Settings section.
Add the following prefix at the 'Incoming Calling Party Settings section of Branch1
gateway:

Globalize Branch2 site inbound calls


For Branch2 the globalization task is pretty tricky. If you run debug isdn q931 on the Branch2
gateway you will notice that for Subscriber type of calls the Calling Number pattern is
7037XXX, so the initial 020 is already omitted by the PSTN. But for the National type of calls
the Calling Number pattern is 0107047XXX, means it included the 0 as prefix so we need to
remove this 0from the prefix. So here is how the Branch2 gateway Incoming Calling party
Settings should be configured:

Notice the prefix for the National type calls, its +31:1, means it will omit the 0 from the
pattern 0107047XXX, so the result is +31107047XXX.

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Task 4.15 Inbound Calling from the PSTN

Localize all calls inbound from the PSTN as they arrive at every IP phone
Local (Subscriber) Calls:
o During inbound alerting, IP Phones in the US sites should display calling
party numbers that are local to each site as 10 digits
o During inbound alerting, IP Phones in the NL site should display calling
party numbers that are local to that site as 10 digits
Long Distance (National) Calls:
o During inbound alerting, IP Phones in the US sites should display calling
party numbers as 10 digits if that call is from the same country, but a
different geographic/area code
o During inbound alerting, IP Phones in the NL site should display calling
party numbers as 10 digits if that call is from the same country, but a
different geographic/area code. This means that the calling party number
needs to have the national access code of "0" added back to the front of
the geographic code
International Calls:
o During inbound alerting, IP Phones in the US sites should display calling
party numbers with all digits, including the country code,but no + if that
call is from a different country
o During inbound alerting, IP Phones in the NL site should display calling
party numbers with all digits, including the country code and + if that call
is from a different country
3pts

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Calling Search Space and Partition for Calling Party Transformation Pattern:
For localization we will use Calling Party Transformation Pattern at the incoming gateway. To
work with Calling Party Transformation Pattern we shall need one separate Calling Search
Space and associated Partition. Lets create three partitions for each of the sites:
Partitions created for Localization

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Calling Search Spaces created for Localization, here is for CorpHQ site:

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For Branch1 site:

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For Branch2 site:

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Create the Calling Party Transformation Pattern:

Navigate to Call Routing > Transformation Pattern > Calling Party Transformation
Pattern:

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Now create the Localization pattern for Subscriber and National type calls for US
(CorpHQ and Branch1) phones.

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Another pattern for International type calls for US (CorpHQ and Branch1) phones:

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Same for NL (Branch2) phones for Subscriber and National type calls:

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Apply the Calling Party Transformation Pattern:


We can apply the patterns at the
1. Device Pools
2. Individual Phones
3. Incoming Gateways and Trunks
We cannot apply the patterns into the incoming Gateways or Trunks because we are already
applying the Globalization patterns there. So its better to apply the pattern on the Device Pool as
our Device Pool is designed with individual site considerations.
'Calling Party Transformation Pattern' is applied on CorpHQ and Branch1 Device
Pools:

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The same for NL site (Branch2):

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Task 4.16 4.18 Inbound Calling from the PSTN


Tasks 4.16 4.18
NOTE: This begins the explanation for the next three tasks one for each site. Each task has a
point value independently from one another, however for each task, everything relevant to a
given site (listed below in this explanation), must be configured properly to be awarded the
points for that site.

Create only one set of PSTN Patterns for each country (US and NL) based on
the information in the three tables below
Ensure these PSTN Patterns globalize every Called Party Number to a
properFull E.164 (including preceding +) number format before the call ever
reaches a Route List in CUCM
o NOTE:Emergency numbers need to be globalized to match the global
route patterns - but will not be in "proper" E.164 format - this is OK
Ensure that the Calling and/or Called Party numbers are sent to the PSTN with
the proper Digits and/or Types as listed in the tables below (see next page)
Provision redundancy for egress GW choice as specified in the tables below

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4.1

Dialing From CorpHQ Phones out to the PSTN


Table 1

Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

911

(1) CorpHQ GW

10 ANI Digits

Unknown

Local

[2-9]XX[2-9]XXXXXX

(1) CorpHQ GW

10 ANI Digits

Subscriber

(2) Branch1 GW

11 ANI Digits

National

1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW

11 ANI Digits

National

(2) Branch1 GW

11 ANI Digits

Subscriber

(1) CorpHQ GW

CC+ 10 ANI Digits

International

(2) Branch1 GW

CC+ 10 ANI Digits

International

National/LD

International

011 + Variable length

5pts

4.2

Dialing From Branch1 Phones out to the PSTN


Table 2

Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

911

(1) Branch1 GW

10 ANI Digits

Unknown

Local

[2-9]XX[2-9]XXXXXX

(1) Branch1 GW

10 ANI Digits

Subscriber

(2) CorpHQ GW

11 ANI Digits

National

1[2-9]XX[2-9]XXXXXX (1) Branch1 GW

11 ANI Digits

National

(2) CorpHQ GW

11 ANI Digits

Subscriber

(1) Branch1 GW

CC+ 10 ANI Digits

International

(2) CorpHQ GW

CC+ 10 ANI Digits

International

National/LD

International

011 + Variable length

5pts

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4.3

Dialing From Branch2 Phones out to the PSTN


Table 3

Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

112

(1) Branch2 GW

10 ANI Digits

Subscriber

Local

[1-8]XXXXXX

(1) Branch2 GW

7 ANI Digits

Subscriber

(2) CorpHQ GW

CC+ 9 ANI Digits

International

(1) Branch2 GW

10 ANI Digits

Subscriber

(2) CorpHQ GW

CC+ 9 ANI Digits

International

(1) Branch2 GW

10 ANI Digits

National

(2) CorpHQ GW

CC+ 9 ANI Digits

International

(1) Branch2 GW

CC+ 9 ANI Digits

International

(2) CorpHQ GW

CC+ 9 ANI Digits

Local

020XXXXXXX

National/LD

International

0[1-8]XXXXXXXX

00 + Variable length

5pts

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At the beginning of these tasks we need to prepare the Route Groups and Route Lists
accordingly.
Route Groups:
CorpHQ Site:

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Branch1 Site:

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Branch2 Site:

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Now assign these Route Groups to respective site Device Pools to make them local to site as of
Standard Local Route Group concept.
CorpHQ Device Pool:

Branch1 Phones Device Pool:

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Branch2 Device Pool:

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4.16 Dialing from CorpHQ Phones out to the PSTN


Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

911

(1) CorpHQ GW

10 ANI Digits

Unknown

CSS and Partition for Translation Pattern:

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Translation Pattern

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Route Pattern:

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The Route List Configuration:

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CSS and Partition for Called Party Transformation Pattern:

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Create Called Party Transformation Pattern for outgoing 911 calls:

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Assign the CdPTP into the CorpHQ Trunk:

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On the CorpHQ Router we need to create separate translation rules and profile for the emergency
call to set the calling/called party type unknown/unknown, also need to create a dial-peer
towards PSTN/PRI.
!
voice translation-rule 10
rule 1 // // type any unknown
!
voice translation-rule 20
rule 1 // // type any unknown
!
voice translation-profile Add_ANI_Type_EMS
translate calling 10
translate called 20
!
dial-peer voice 10 pots
translation-profile outgoing Add_ANI_Type_EMS
destination-pattern 911$
port 0/0/0:23
forward-digits 3
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Local

[2-9]XX[2-9]XXXXXX

(1) CorpHQ GW

10 ANI Digits

Subscriber

(2) Branch1 GW

11 ANI Digits

National

CSS and Partition for Translation Pattern:

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Translation Pattern for Local Calls:

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Route Pattern:

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It's the same route pattern used for 911 calls, we are reusing this RP.
Route List Configuration:

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CSS and Partition for Called Party Transformation Pattern for CorpHQ Gateway:

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CSS and Partition for Called Party Transformation Pattern for Branch1 Gateway:

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Create Called Party Transformation Pattern for outgoing Local calls through
CorpHQ Gateway:

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Create Called Party Transformation Pattern for outgoing Local calls through
redundant Branch1 Gateway (which is National type of calls in actual):

Assign the CdPTP into the CorpHQ Trunk:


We have already assigned it.

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Assign the CdPTP into the Branch1 Gateway:

On the CorpHQ Router:


!
!
voice translation-rule 9
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011.*/ /9&/
!
voice translation-profile Prefix_9_DNIS
translate called 9
!
voice translation-rule 11
rule 1 /^2065011...$/ /&/ type any subscriber
rule 2 /^15126022...$/ /&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 21
rule 1 // // type any subscriber
!

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voice translation-profile Add_ANI_Type_LOCAL


translate calling 11
translate called 21
!
dial-peer voice 101 voip
translation-profile incoming Prefix_9_DNIS
voice-class codec 1
incoming called-number .
!
dial-peer voice 11 pots
translation-profile outgoing Add_ANI_Type_LOCAL
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 10
!

On the Branch1 Gateway:


!
voice translation-rule 9
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011.*/ /9&/
!
voice translation-profile Prefix_9_DNIS
translate called 9
!
voice translation-rule 12
rule 1 /^15126022...$/ /&/ type any national
rule 2 /^2065011...$/ /1&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 22
rule 1 // // type any national
!

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voice translation-profile Add_ANI_Type_NATL


translate calling 12
translate called 22
!
dial-peer voice 12 pots
translation-profile outgoing Add_ANI_Type_NATL
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
!

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Call Type

PSTN Pattern

National/LD

Egress GW

Calling # Format

Called Party Type

1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW

11 ANI Digits

National

(2) Branch1 GW

11 ANI Digits

Subscriber

CSS and Partition for Translation Pattern:

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Translation Pattern for National Calls:

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Route Pattern (already created):

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Route List Configuration (already created):

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CSS and Partition for Called Party Transformation Pattern for CorpHQ gateway
(already created):

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CSS and Partition for Called Party Transformation Pattern for Branch1 gateway
(already created):

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Create Called Party Transformation Pattern for outgoing National calls through
CorpHQ Gateway:

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Create Called Party Transformation Pattern for outgoing National calls through
redundant Branch1 Gateway (which would be Local type of calls in actual):

Assign the CdPTP into the CorpHQ Trunk:


We have already assigned it.
Assign the CdPTP into the Branch1 Gateway:
We have already assigned it.

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On the CorpHQ Router:


!
voice translation-rule 12
rule 1 /^12065011...$/ /&/ type any national
rule 2 /^5126022...$/ /1&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 22
rule 1 // // type any national
!
voice translation-profile Add_ANI_Type_NATL
translate calling 12
translate called 22
!
dial-peer voice 12 pots
translation-profile outgoing Add_ANI_Type_NATL
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
!

On the Branch1 Router:

voice translation-rule 11
rule 1 /^5126022...$/ /&/ type any subscriber
rule 2 /^12065011...$/ /&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 21
rule 1 // // type any subscriber
!
voice translation-profile Add_ANI_Type_LOCAL
translate calling 11
translate called 21
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!
dial-peer voice 11 pots
translation-profile outgoing Add_ANI_Type_LOCAL
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 10
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

International

011 + Variable length

(1) CorpHQ GW

CC+ 10 ANI Digits

International

(2) Branch1 GW

CC+ 10 ANI Digits

International

CSS and Partition for Translation Pattern:

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Translation Pattern for International Calls:

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Route Pattern (already created):

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Route List Configuration (already created):

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CSS and Partition for Called Party Transformation Pattern for CorpHQ Gateway
(already created):

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CSS and Partition for Called Party Transformation Pattern for Branch1 Gateway:

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Create Called Party Transformation Pattern for outgoing International calls for
CorpHQ and Branch1 gateway:

Assign the CdPTP into the CorpHQ Trunk:


We have already assigned it.
Assign the CdPTP into the Branch1 Gateway:
We have already assigned it.

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On the CorpHQ Router:


!
voice translation-rule 13
rule 1 /^7033...$/ /3120&/ type any international
rule 2 /^0\(207033...\)$/ /31\1/ type any international
rule 3 // // type any international
!
voice translation-rule 23
rule 1 // // type any international
!
voice translation-profile Add_ANI_Type_INTL
translate calling 13
translate called 23
!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!
On the Branch1 Router:
!
voice translation-rule 23
rule 1 // // type any international
!
voice translation-rule 13
rule 1 /^7033...$/ /3120&/ type any international
rule 2 /^0\(207033...\)$/ /31\1/ type any international
rule 3 // // type any international
!
voice translation-profile Add_ANI_Type_INTL
translate calling 13
translate called 23
!
dial-peer voice 13 pots

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translation-profile outgoing Add_ANI_Type_INTL


destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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4.17 Dialing from Branch1 Phones out to the PSTN


Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

911

(1) Branch1 GW

10 ANI Digits

Unknown

CSS and Partition for Translation Pattern (already created):


We have already created when we were configuring the CorpHQ 911 of the CSS: CSS_TP-toPSTN-RP
Translation Pattern (already created):
We have already created when we were configuring the CorpHQ 911 of the pattern 911
Route Pattern (already created):
We have already created when we were configuring the CorpHQ 911 of the pattern \+!
The Route List Configuration (already created):
We have already created when we were configuring the CorpHQ 911 of the name RL_PSTNSLRG

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CSS and Partition for Called Party Transformation Pattern:

Create Called Party Transformation Pattern for outgoing 911 calls:


We have already created it of the pattern \+.911

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Assign the CdPTP into the Branch1 Gateway:

On the Branch1 Router we need to create separate translation rules and profile for the emergency
call to set the calling/called party type unknown/unknown, also need to create a dial-peer
towards PSTN/PRI.
!
voice translation-rule 10
rule 1 // // type any unknown
!
voice translation-rule 20
rule 1 // // type any unknown
!
voice translation-profile Add_ANI_Type_EMS
translate calling 10
translate called 20
!
dial-peer voice 10 pots
translation-profile outgoing Add_ANI_Type_EMS
destination-pattern 911$
port 0/0/0:23
forward-digits 3
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Local

[2-9]XX[2-9]XXXXXX

(1) Branch1 GW

10 ANI Digits

Subscriber

(2) CorpHQ GW

11 ANI Digits

National

CSS and Partition for Translation Pattern (already created):


We have already the CSS with the name CSS_TP-to-PSTN-RP and the Parition with the name
PT_TP-to-PSTN-RP
Translation Pattern for Local Calls (already created):
The Translation Pattern is also created of the pattern 9.[2-9]XX[2-9]XXXXXX
Route Pattern (already created):
The Route Pattern is also created of the pattern: \+!
Route List Configuration (already created):
The RL is created already with the name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch1 Gateway
(already created):
This is also created already with the name: CSS_CdPTP-Branch1-GW

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Create Called Party Transformation Pattern for outgoing Local calls through
Branch1 Gateway:

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Create Called Party Transformation Pattern for outgoing Local calls through
redundant CorpHQ Gateway (which is National type of calls in actual):

Assign the CdPTP into the Branch1 Trunk (already assigned):


Already assigned the CSS CSS_CdPTP-Branch1-GW at Branch1 Gateway.
Assign the CdPTP into the COrpHQ Trunk (already assigned):
Already assigned the CSS CSS_CdPTP-CorpHQ-GW at CorpHQ Gateway.

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On the Branch1 Router:


!
voice translation-rule 11
rule 1 /^5126022...$/ /&/ type any subscriber
rule 2 /^12065011...$/ /&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 21
rule 1 // // type any subscriber
!
voice translation-profile Add_ANI_Type_LOCAL
translate calling 11
translate called 21
!
dial-peer voice 11 pots
translation-profile outgoing Add_ANI_Type_LOCAL
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 10
!
On the CorpHQ Router:

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Call Type

PSTN Pattern

National/LD

Egress GW

Calling # Format

Called Party Type

1[2-9]XX[2-9]XXXXXX (1) Branch1 GW

11 ANI Digits

National

(2) CorpHQ GW

11 ANI Digits

Subscriber

CSS and Partition for Translation Pattern (already created):


Partition name: PT_TP-to-PSTN-RP, CSS name: CSS_TP-to-PSTN-RP
Translation Pattern for national Calls (already created):
Of the pattern: 9.1[2-9]XX[2-9]XXXXXX
Route Pattern (already created):
Of the pattern: \+!
Route List Configuration (already created):
RL name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch1 Gateway
(already created):
CSS name: CSS_CdPTP-Branch1-GW
CSS and Partition for Called Party Transformation Pattern for CorphHQ Gateway
(already created):
CSS name: CSS_CdPTP-CorpHQ-GW
Create Called Party Transformation Pattern for outgoing National calls through
Branch1 Gateway (already created):
Pattern: \+.1!

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Create Called Party Transformation Pattern for outgoing National calls through
redundant CorpHQ Gateway, which is Local type of calls in actual (already
created):
Pattern: \+1.206!
Assign the CdPTP into the Branch1 Trunk (already assigned):
CSS CSS_CdPTP-Branch1-GW is already assigned on the Branch1 gateway.
Assign the CdPTP into the COrpHQ Trunk (already assigned):
CSS CSS_CdPTP-CorpHQ-GW is already assigned on the CorpHQ gateway.

In the Branch1 Gateway (already created):


!
voice translation-rule 12
rule 1 /^15126022...$/ /&/ type any national
rule 2 /^2065011...$/ /1&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 22
rule 1 // // type any national
!
voice translation-profile Add_ANI_Type_NATL
translate calling 12
translate called 22
!
translation-profile outgoing Add_ANI_Type_NATL
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
!

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In the CorpHQ Router (already created):

!
voice translation-rule 11
rule 1 /^2065011...$/ /&/ type any subscriber
rule 2 /^15126022...$/ /&/ type any national
rule 3 /^31207033...$/ /&/ type any international
!
voice translation-rule 21
rule 1 // // type any subscriber
!
voice translation-profile Add_ANI_Type_LOCAL
translate calling 11
translate called 21
!
dial-peer voice 11 pots
translation-profile outgoing Add_ANI_Type_LOCAL
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 10
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

International

011 + Variable length

(1) Branch1 GW

CC+ 10 ANI Digits

International

(2) CorpHQ GW

CC+ 10 ANI Digits

International

CSS and Partition for Translation Pattern (already created):


Partition name: PT_TP-to-PSTN-RP, CSS name: CSS_TP-to-PSTN-RP
Translation Pattern for International Calls (already created):
Pattern: 9011.!
Route Pattern (already created):
Pattern: \+!
Route List Configuration (already created):
Route List name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch1 Gateway
(already created):
CSS name: CSS_CdPTP-Branch1-GW
CSS and Partition for Called Party Transformation Pattern for CorpHQ Gateway
(already created):
CSS name: CSS_CdPTP-CorpHQ-GW

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Create Called Party Transformation Pattern for outgoing International calls for
CorpHQ and Branch1 gateway (already created):
Pattern: \+.!
Assign the CdPTP into the Branch1 Gateway:
We have already assigned it, CSS name: CSS_CdPTP-Branch1-GW
Assign the CdPTP into the CorpHQ Trunk:
We have already assigned it, CSS name: CSS_CdPTP-CorpHQ-GW

On the Branch1 Router (already created):


!
voice translation-rule 23
rule 1 // // type any international
!
voice translation-rule 13
rule 1 /^7033...$/ /3120&/ type any international
rule 2 /^0\(207033...\)$/ /31\1/ type any international
rule 3 // // type any international
!
voice translation-profile Add_ANI_Type_INTL
translate calling 13
translate called 23
!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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On the CorpHQ Router (already created):


!
voice translation-rule 13
rule 1 /^7033...$/ /3120&/ type any international
rule 2 /^0\(207033...\)$/ /31\1/ type any international
rule 3 // // type any international
!
voice translation-rule 23
rule 1 // // type any international
!
voice translation-profile Add_ANI_Type_INTL
translate calling 13
translate called 23
!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Emergency

112

(1) Branch2 GW

10 ANI Digits

Subscriber

CSS and Partition for Translation Pattern (already created):


CSS: CSS_TP-to-PSTN-RP, Partition: PT_TP-to-PSTN-RP
Translation Pattern for Emergency Calls:

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Another pattern with access code '0'

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Route Pattern (already created):


Pattern: \+!
Route List Configuration (already created):
Route List name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch2 Gateway:

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Create Called Party Transformation Pattern for outgoing Emergency calls through
Branch2 gateway:

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Assign the CdPTP into the Branch2 Gateway:

On the Branch2 Router:


!
voice translation-rule 10
rule 1 // // type any subscriber
!
voice translation-rule 20
rule 1 // // type any subscriber
!
voice translation-profile Add_ANI_Type_EMS
translate calling 10
translate called 20
!
dial-peer voice 10 pots
translation-profile outgoing Add_ANI_Type_EMS
destination-pattern 112$
port 0/0/0:15
forward-digits 3
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

Local

[1-8]XXXXXX

(1) Branch2 GW

7 ANI Digits

Subscriber

(2) CorpHQ GW

CC+ 9 ANI Digits

International

(1) Branch2 GW

10 ANI Digits

Subscriber

(2) CorpHQ GW

CC+ 9 ANI Digits

International

Local

020XXXXXXX

CSS and Partition for Translation Pattern (already created):


CSS name: CSS_TP-to-PSTN-RP and Partition name: PT_TP-to-PSTN-RP
Translation Pattern for Local Calls:

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Another pattern:

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Route Pattern (already created):


Pattern: \+!
Route List Configuration (already created):
Route List name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch2 Gateway
(already created):
CSS name: CSS_CdPTP-Branch2-GW
CSS and Partition for Called Party Transformation Pattern for CorphHQ Gateway
(already created):
CSS name: CSS_CdPTP-CorpHQ-GW
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Create Called Party Transformation Pattern for outgoing Local calls through
Branch2 gateway:

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Create Called Party Transformation Pattern for outgoing Local calls through
CorpHQ gateway (which is actually International type calls from CorpHQ):

Assign the CdPTP into the Branch2 Gateway (already assigned):


CSS to assign: CSS_CdPTP-Branch2-GW
Assign the CdPTP into the CorpHQ Gateway (already assigned):
CSS to assign: CSS_CdPTP-CorpHQ-GW

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In the Branch2 Router:


!
voice translation-rule 9
rule 1 /^703....$/ /0&/
rule 2 /^0[1-8]........$/ /0&/
rule 3 /^00.*/ /0&/
!
voice translation-profile Prefix_0_DNIS
translate called 9
!
voice translation-rule 11
rule 1 /^0207033...$/ /&/ type any subscriber
rule 2 /^12065011...$/ /&/ type any international
rule 3 /^15126022...$/ /&/ type any international
!
voice translation-rule 21
rule 1 // // type any subscriber
!
voice translation-profile Add_ANI_Type_LOCAL
translate calling 11
translate called 21
!
dial-peer voice 103 voip
translation-profile incoming Prefix_0_DNIS
incoming called-number .
!
dial-peer voice 12 pots
translation-profile outgoing Add_ANI_Type_LOCAL
destination-pattern 0[1-8]......$
port 0/0/0:15
forward-digits 7
!

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On the CorpHQ Router (already created):


!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

National/LD

0[1-8]XXXXXXXX

(1) Branch2 GW

10 ANI Digits

National

(2) CorpHQ GW

CC+ 9 ANI Digits

International

CSS and Partition for Translation Pattern (already created):


CSS name: CSS_TP-to-PSTN-RP and Partition name: PT_TP-to-PSTN-RP
Translation Pattern for National Calls (already created):
We already created the pattern: 00.[1-8]XXXXXXXX for use in Local call, we can use the same
pattern for National type of calls.
Route Pattern (already created):
Pattern: \+!
Route List Configuration (already created):
RL name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch2 Gateway
(already created):
CSS name: CSS_CdPTP-Branch2-GW with Partition: PT_CdPTP-Branch2-GW
CSS and Partition for Called Party Transformation Pattern for CorphHQ Gateway
(already created):
CSS name: CSS_CdPTP-CorpHQ-GW with Partition: PT_CdPTP-CorpHQ-GW

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Create Called Party Transformation Pattern for outgoing National calls through
Branch2 gateway:

Create Called Party Transformation Pattern for outgoing National calls through
CorpHQ gateway, which is actually International type calls from CorpHQ (already
created):
The pattern is: \+.!
Assign the CdPTP into the Branch2 Gateway (already assigned):
Assigned CSS: CSS_CdPTP-Branch2-GW

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Assign the CdPTP into the COrpHQ Trunk (already assigned):


Assigned CSS: CSS_CdPTP-CorpHQ-GW

On the Branch2 Router:


!
voice translation-rule 12
rule 1 /^0207033...$/ /&/ type any national
rule 2 /^12065011...$/ /&/ type any international
rule 3 /^15126022...$/ /&/ type any international
!
voice translation-rule 22
rule 1 // // type any national
!
voice translation-profile Add_ANI_Type_NATL
translate calling 12
translate called 22
!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_NATL
destination-pattern 00[126]........$
port 0/0/0:15
forward-digits 10
!

On the CorpHQ Router (already created):


!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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Call Type

PSTN Pattern

Egress GW

Calling # Format

Called Party Type

International

00 + Variable length

(1) Branch2 GW

CC+ 9 ANI Digits

International

(2) CorpHQ GW

CC+ 9 ANI Digits

CSS and Partition for Translation Pattern (already created):


CSS name: CSS_TP-to-PSTN-RP and Partition name: PT_TP-to-PSTN-RP
Translation Pattern for International Calls:

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Another pattern:

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Route Pattern (already created):


Pattern: \+!
Route List Configuration (already created):
RL name: RL_PSTN-SLRG
CSS and Partition for Called Party Transformation Pattern for Branch2 Gateway
(already created):
CSS name: CSS_CdPTP-Branch2-GW with Partition: PT_CdPTP-Branch2-GW

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CSS and Partition for Called Party Transformation Pattern for CorphHQ Gateway
(already created):
CSS name: CSS_CdPTP-CorpHQ-GW with Partition: PT_CdPTP-CorpHQ-GW
Create Called Party Transformation Pattern for outgoing National calls through
Branch2 gateway:

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Create Called Party Transformation Pattern for outgoing National calls through
CorpHQ gateway which is actually International type calls from CorpHQ (already
created):
Pattern: \+.!
Assign the CdPTP into the Branch2 Gateway (already assigned):
CSS assigned: CSS_CdPTP-Branch2-GW
Assign the CdPTP into the COrpHQ Trunk (already assigned):
CSS assigned: CSS_CdPTP-CorpHQ-GW

In the Branch2 Router:


!
voice translation-rule 13
rule 1 /^31207033...$/ /&/ type any international
rule 2 /^12065011...$/ /&/ type any international
rule 3 /^15126022...$/ /&/ type any international
!
voice translation-rule 23
rule 1 // // type any international
!
voice translation-profile Add_ANI_Type_INTL
translate calling 13
translate called 23
!
dial-peer voice 14 pots
corlist outgoing Door-INTL
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 000T
port 0/0/0:15
prefix 00
!

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On the CorpHQ Router (already created):


!
dial-peer voice 13 pots
translation-profile outgoing Add_ANI_Type_INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
!

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Task 4.19 Inbound Calling from the PSTN

As per Table 3, notice that users may dial local calls in two ways: both with 7 digit
and 10 digit numbers (beginning with 020). Ensure that either way a user dials
this, that is the way the call goes out to the PSTN (i.e. user dials 7 digits, 7 digits
are sent to PSTN, if 10 are dialed, then send 10 to PSTN), however all numbers
must still be able to match a globalized + route pattern, before being localized for
PSTN at the egress
4pts

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We have already completed the task while configuring the Local calls. The manipulations of
digits are:
For the pattern: [1-8]XXXXXX
Applying Translation Pattern to change the pattern from 0.[1-8]XXXXXX to +#3120[18]XXXXXX, so its matching the globalized Route Pattern: \+!
For the Pattern: 00.[1-8]XXXXXXXX
With the Translation Pattern to change the pattern from 00.[1-8]XXXXXXXX to +31[18]XXXXXXXX, so its also matching the globalized Route Pattern : \+!

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Task 4.20 Inbound Calling from the PSTN

Ensure that any user at any hardware IP Phone who views their Missed Calls
can simply press the "Dial" softkey to return any call
Urgent Priority must be used in your patterns, and SIP phones must work
properly
You must maintain the proper Calling and/or Called Digits and/or Types
described by the previous tasks
You do not have to worry about CoR from task 4.3
3pts

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CorpHQ Phones
Any Subscriber/National of International type calls are globalized at the egress IOS. So the
CorpHQ phones dialing from Missed Calls globalized numbers should dial the same type of
called numbers we need to ensure this in this section.

For CorpHQ Subscriber (Local) Calls:


Globalized calling number: +1[2-9]XX[2-9]XXXXXX
Seen at Missed Call: +12065015XXX
Translation Pattern:

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Route Pattern (already created):

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Route List (already created):


RL name: RL_PSTN-SLRG
Called Party Transformation Pattern (already created):

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For CorpHQ National Calls:


Globalized calling number: +1[2-9]XX[2-9]XXXXXX
Seen at Missed Call: +15126026XXX
Translation Pattern:

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Route Pattern (already created):


Pattern: \+!
Called Party Transformation Pattern (already created):
Pattern: \+.1!

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For CorpHQ International Calls:


Globalized calling number: +!
Seen at Missed Call: +!
Translation Pattern:

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Route Pattern (already created):


Pattern: \+!
Called Party Transformation Pattern (already created):
Pattern: \+.!

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Create a SIP Dial Rule for SIP Phone:

Create a SIP Dial Rule:

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Apply the new dial rule on CorpHQ Phone2:

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Branch1 Phones
Any Subscriber/National of International type calls are globalized at the egress gateway settings
on UCM. So the Branch1 phones dialing from Missed Calls globalized numbers should dial the
same type of called numbers we need to ensure this in this section.

Branch1 Subscriber (Local) calls:


Globalized calling number: +1[2-9]XX[2-9]XXXXXX
Seen at Missed Call: +15126026XXX

Translation Pattern:

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Route Pattern (already created):

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Route List (already created):


RL name: RL_PSTN-SLRG
Called Party Transformation Pattern (already created):

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Branch1 National calls:


Globalized calling number: +1[2-9]XX[2-9]XXXXXX
Seen at Missed Call: +12065015XXX
Translation Pattern:

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Route Pattern (already created):


Pattern: \+!
Called Party Transformation Pattern (already created):
Pattern: \+.1!

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For Branch1 International Calls:


Globalized calling number: +!
Seen at Missed Call: +!
Translation Pattern:

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Route Pattern (already created):


Pattern: \+!
Called Party Transformation Pattern (already created):
Pattern: \+.!

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Task 4.21 Mobility

Configure CUCM such that when Hugo Reyes CorpHQ Phone 2 at DN 1002 is
called, that his home phone is rung immediately, and his mobile phone is rung
after 1 full ring at his desk phone
From the information below, calculate the time it takes those mobile and home
phones to ring to VM, and prevent this mechanism from ringing into any of their
respective voicemail boxes
o Hurley's Mobile #: +1 206 501 5555 -Forwards to VM after 4 rings
This phone should only be rung Mon-Fri 12:00 24:00 and all day
Sat local to his site
Mobile Connect calls displaying on this phone should show up in
the globalized format if coming from outside the enterprise, and as
the standard 4 digit extension if coming from inside the company
o Hurley's Home #: +1 206 501 5151 -Forwards to VM after 3 rings
This phone should be rung Mon-Fri 00:00 12:00 and all day Sun
local to his site
Mobile Connect calls displaying on this phone should show up in
the proper format so that he can simply press Dial from his home
phone, if the call came from inside or outside of the enterprise
o Note: For testing purposes, these calls will both ring to the PSTN Phone
Line 5, however they have different CFwdNoAn timeouts and simulated
VM on INE/GradedLabs Voice Racks, and will show the differing CallerID
based on the above requirements being met for Home and Mobile phones
respectively
Allow Hurley to be able to transfer calls from his mobile phone back to his desk
phone, and also from his desk phone back to his mobile phone with a single
button
Also allow Hurley to be able to login to his CCMUser page and setup, change or
even add any Mobile Connect remote phones
4pts

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Enable mobility option from user page for the user 'hreyes':

Navigate to Device > Device Settings > Remote Destination Profile:

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Create new RDP for the user 'hreyes':

Associate the DN '1002' with the Remote Destination Profile:

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Navigate to Device > Remote Destination:

Create a new Remote Destination for the 'Home' phone:

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The Ring Schedule is right below:

Don't forget to associate the line '1002' with this RD:

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Create another Remote Destination for the 'Mobile':

And the time schedule is here:

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Associate the line '1002':

Return the 'Remote Destination Profile' and check if both of the Remote Destination
is associated with it:

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Navigate to Call Routing > Dial Rules > Application Dial Rules

Create a dial rule as of the below:

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Go to Phone settings for Hugo Reyes (1002) and check if the phone is associated
with correct user id (hreyes):

Also check if the line '1002' is associated with both the phone and RDP

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Now configure the softkey so that the 'Mobility' softkey appears during 'on-hook'
and 'connected' state, navigate to 'softkey template':

Click the right softkey to change:

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Click on the 'Configure Softkey' next page:

Bring 'Mobility' softkey on the right panel for both 'On Hook' and 'Connected'
state:

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Don't forget to restart the phones where this softkey is assigned, you can restart them from right
this page with 'Reset' button at the left top.

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Assign the user 'hreyes' into the 'Standard CCM End User' group to permit him to
login at 'CCMUser' page:

To globalize the calling number on Home/Mobile phone lets tweak the calling number on
CorpHQ Router:
!
voice translation-rule 11
rule 1 /^2065011...$/ /&/ type any subscriber
rule 2 /^15126022...$/ /&/ type any national
rule 3 /^31.*/ /+\0/ type any international
rule 4 /^206501.*/ /+1\0/ type any subscriber
rule 5 /^1512602.*/ /+\0/ type any national
voice translation-rule 21
rule 1 // // type any subscriber
voice translation-profile Add_ANI_Type_LOCAL
translate calling 11
translate called 21
!

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Task 4.22 Mobility

Ensure that when Mobile Connect is invoked, that the call being placed out to the
PSTN tries to ring out the gateway that is local to the Hurleys remote PSTN
phone
o If Mobile Connect is ringing Hurley's mobile or home phone, the call
should try to ring out the CorpHQ GW primarily, if it is available
3pts

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Phone for Hugo Reyes (1002) is located on CorpHQ site, so as instructed the calls for his Home
and Mobile should ring out through CorpHQ router and its already configured so. On the Route
Pattern \+! the route list is set to RL_PSTN-SRLG where the first preferred gateway is SLRG.
So we dont need any additional configuration for this task.

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Task 4.23 Mobility

Allow Hurley the ability to call into the company from his Mobile Phone (PSTN
Line 5) and make International calls back out to through the PSTN.
3pts

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This task requires to configure the Mobile Voice Access (MVA), MVA allows the users to call
International/National numbers from their Remote Destination (i.e. Mobile for Hugo Reyes).
Enable Mobile Voice Access from Service Parameter, navigate to Service
Parameter:

Make necessary changes on Service Parameter:

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Navigate to Media Resources > Mobile Voice Access:

Add the MVA DN and Partition, also the Locale:

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Enable Mobile Voice Access from the user page for 'hreyes', also set a PIN for the
user:

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In the Branch1 Router:


Create an application with MVA access URL from UCM:
!
application
service MVA
http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
!

After applying this command the router will download the vxml script from the
UCM and will show similar console logs:
Branch1(config-app-param)#
//-1//HIFS:/hifs_http_cb: hifs http read succeeded. size=1551,
url=http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
//-1//HIFS:/hifs_http_cb: hifs ifs file read succeeded. size=1551,
url=http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
//-1//HIFS:/hifs_free_idata: hifs_free_idata: 0x4B12527C
//-1//HIFS:/hifs_hold_idata: hifs_hold_idata: 0x4B12527C
Branch1(config-app-param)#

Create an incoming dial-peer with MVA service attached:


!
dial-peer voice 501 pots
description == MVA Inbound from PSTN
service mva
incoming called-number 2800
!

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Outgoing dial-peer which will forward the call to UCM to reach the MVA DN 2800:
!
dial-peer voice 101 voip
preference 1
destination-pattern ^2...$
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.10
dtmf-relay h245-signal
!
dial-peer voice 102 voip
preference 2
destination-pattern ^2...$
voice-class codec 1
voice-class h323 1
session target ipv4:177.1.10.20
dtmf-relay h245-signal
!

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Task 4.24 High Availability

Provision Branch2 R3 as a SCCP server for CME as SRST phone fallback with
the following stipulations:
o Provision the SCCP server IP address as the same as the Loopback0
interface
o Normal calls to the PSTN should function the same way
You need not worry about inbound/outbound
globalization/localization
o Ensure that the phone looks and behaves just as it would when registered
to CUCM (some features may not work, configure all that you can, and
ask specific questions for clarification on things you suspect may not work
when in fallback mode)
o Do not provision any upgrades to firmware for the fallen back phones
o You may use the srst auto-provision all command, but you must first
prebuild all of the necessary ephone-dns, and ensure that proper call
routing to the CUCM works properly when the phones are not in fallback
mode
o Ensure that when the CUCM is accessible again, that the phone register
again with it within 15 seconds or less
4pts

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Navigate to System > SRST:

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Create SRST reference for Branch2:

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Assign SRST Reference on Device Pool for Branch2:

Save and Reset to affect this on phones.

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Configuring SRST on Branch2 Router:


!
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 1
sdspfarm tag 4 Brnch2-HW-Conf
no privacy
privacy-on-hold
conference hardware
srst mode auto-provision all
max-ephones 10
max-dn 100
ip source-address 177.1.254.3 port 2000
system message Your current options
time-zone 23
time-format 24
keepalive 15
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 177.3.11.1
177.1.254.3
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 0
!
!
!
ephone-template 1
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd
Login HLog
softkeys connected Hold Endcall Trnsfer Confrn Park HLog
Join
!
!
ephone-dn 1 octo-line

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number 3001
label Desmond Hume x3001
description 0207033001
name Desmond Hume
preference 1
!
!
ephone-dn 2 octo-line
number 3002
label James Ford x3002
description 0207033002
name James Ford
preference 1
!
As the DN 3001 and 3002 are created in the router, the PSTN to CUCM call routing might not
work properly unless we reconfigure the dial-peers with more specific destination-pattern.
Otherwise these ephone-dns would be choose as primary destination and all calls towards
CUCM will fail. So lets modify the route patterns accordingly:
!
dial-peer voice 101 voip
preference 2
destination-pattern ^3001$
session target ipv4:177.1.10.10
!
dial-peer voice 102 voip
preference 3
destination-pattern ^3001$
session target ipv4:177.1.10.20
!
dial-peer voice 103 voip
preference 2
destination-pattern ^3002$
session target ipv4:177.1.10.10
!
dial-peer voice 104 voip
preference 3

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destination-pattern ^3002$
dtmf-relay h245-signal
session target ipv4:177.1.10.20
!

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Task 4.25 High Availability

No matter what network situation may occur (e.g. WAN down or low CAC
bandwidth), all calls from any phone to any other 4 digit extension must continue
to work seamlessly
All IP phones at all sites should see 4 digit ANI when receiving a call from any
other IP phone at any site at any time
You are not permitted to change any phones top-right display from previous task
requirements
3pts

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On the Branch2 router:


!
dial-peer voice 4000
destination-pattern
port 0/0/0:15
prefix 0012065011
!
dial-peer voice 4001
destination-pattern
port 0/0/0:15
prefix 0015126022
!

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1...$

pots
2...$

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Task 4.26 Messaging

Integrate CUCM with Unity Connection using only SIP


Import both CorpHQ users via AXL and create mailboxes for them
Make all passwords 55555
MWI must work properly
Set all phones Busy and NoAn to forward to VM after 4 rings
From both CUCM andCUC only allow RFC2833 to be used for DTMF Relay
3pts

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Navigate to System > Security Profile > SIP Trunk Security Profile to modify the
SIP Trunk Security Profile:

Click on the 'Non-Secure SIP Trunk Profile:

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Check the following 'Accept' parameters:

Save the profile.

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Navigate to Device > Trunk:

Click 'Add New' to create a new SIP Trunk with CUC.


Configure the new trunk as shown:

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Save and Reset the trunk.


Navigate to Route Pattern:

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Now create a new Route Pattern towards CUC SIP trunk:

Save the Route Pattern.


Now we need to create a new Voice Mail Pilot, navigate to Voice Mail > Voice Mail
Pilot:

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Create the Voice Mail Pilot:

Navigate to Voice Mail > Voice Mail Profile to create a new profile:

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Create a new profile or edit the 'Default' profile:

If the Default profile is modified, all phones would be associated with the 'Default' profile
automatically. You can check one CorpHQ phone.

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For the DN 1001:

Configure the 'Busy' and 'No Answer to Forward' settings as instructed for both
1001 and 1002:

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Navigate to End User page:

For each CorpHQ users choose the 'Primary Extension', one example below:

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Navigate to Telephony Integrations > Phone Systems:

Click on 'Add New' and configure the Phone System as shown:

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Make is default Phone System:

Navigate to Edit > Cisco Unified Communication Manager AXL Servers:

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Click on 'Add New' and create and configure the new AXL server IP Address/Port
and Credentials:

Save it and click on 'Test' at the right panel to test the integration.
Save and click on Add Port Group to create a new Port Group:

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Configure the Port Group as shown:

Save the Port Group.

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Click on 'Register with SIP Server' after saving the Port Group:

Reset the Port Group after saving.

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Navigate to Edit > Servers:

Click 'Add' and set both PUB and SUB in the list, also add TFTP Servers
accordingly:

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Click on the 'Add Ports' at top right to add new port:

Configure and save it:

Save ports.

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Navigate to System Settings > Authentication Rules:

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Click on 'Recommended Voice Mail Authentication Rule' and uncheck the option
'Check for Trivial Passwords':

Click on 'Save'.

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Navigate to User Templates from Templates > User Templates:

Click on 'voicemailusertemplate' and change the Phone System to 'CUC-SIP':

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Navitage to Edit > Password Settings under the voicemailusertemplate:

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Uncheck the option 'User Must Change on Next Logon':

Navigate to 'Edit > Change Password':

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Set the password '55555' as instructed:

Save it.
To import the CorpHQ users navigate to Tools > Import Users:

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Import the users; it should import all CorpHQ users:

Select both of the users and click on 'Import Selected'.

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Task 4.27 Messaging

Provision Live Record with the DN 5281 on Unity Connection and test it from
Branch1 Phone1
3pts

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Navigate to Call Routing > Directory Number:

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Create a new directory number for live recording:

Save the DN.


Modify the Service Parameter for the field 'Drop Ad Hoc Conference' to 'When
Conference Controller Leaves':

Save Service Parameter.

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On the CUC navigate to Call Routing > Forwarded Routing Rules:

Click 'Add New' and put a name on the text box, we made it 'Live Record':

Save it.

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Make the status 'Active' and under Conversation set it 'Start Live Record':

Click Save.
Click on the 'Add New' under 'Routing Rule Condition' at the below:

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The condition should be looking like below:

Save it.

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Task 4.28 Messaging

Provision the Unity Express module on Branch2 (R3) with the IP address of
177.1.250.254, however you may not change the IP address or Subnet Mask on
Loopback1
Create mailboxes that work properly at all times for the users of phones at
Branch2
You may only import any users needed to create mailboxes from the CUCM
Make all passwords 55555
MWI must work properly at all times
Set all phones Busy and NoAn to forward to VM after 4 rings
3pts

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Create two CTI Ports on CUCM with the name Branch2VMPort1 and
Branch2VMPort2:

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Assign extension 3851 on the ports:

Create another CTI Port with the name Branch2VMPort2 and assign the extention 3852

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Navigate to Devices > CTI Route Point:

Now create a CTI Route Point:

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Assign the extension 3850 on this CTI RP:

Save the CTI Route Point.

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Navigate to Voice Mail > Voice Mail Pilot:

Create a new Voice Mail Pilot with the same extension of CTI Route Point:

Save the Pilot.

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Navigate to Voice Mail > Voice Mail Profile:

Create a new Voice Mail Profile for the CUE:

Save the Profile.

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Navigate to Voice Mail > Message Waiting:

Create two DN each for MWI ON and MWI OFF:

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Navigate to User Management > Application User:

Create one Application User 'cue' with permission to control CTI Port and CTI
Route Point we have created:

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Save the user.

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Set the newly created voice mail profile to Branch2 phones:

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Configure 'Call Forward No Answer' and 'Call Forward Busy' forwarded to


voicemail after 4 rings:

Repeat the same for 3002 (phone for James Ford).

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On the Branch2 Router:


!
interface Loopback1
ip address 177.1.250.1 255.255.255.0
ip ospf network point-to-point
!
interface Service-Engine0/0
ip unnumbered Loopback1
service-module ip address 177.1.250.254 255.255.255.0
service-module ip default-gateway 177.1.250.1
!
ip route 177.1.250.254 255.255.255.255 Service-Engine0/0
!
dial-peer voice 3850 voip
destination-pattern 385[012]
session protocol sipv2
session target ipv4:177.1.250.254
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
sip-ua
mwi-server ipv4:177.1.250.254 expires 3600 port 5060
transport udp unsolicited
!
telephony-service
voicemail 3850
mwi relay
!
ephone-dn 1 octo-line
call-forward busy 3850
call-forward noan 3850 timeout 10
mwi sip
!
ephone-dn 2 octo-line
call-forward busy 3850
call-forward noan 3850 timeout 10

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mwi sip
!
Access the CUE Module:
Branch2#service-module service-Engine 0/0 session
Trying 177.1.250.1, 2194 ... Open
CUE#
CUE# conf t
Enter configuration commands, one per line. End with
CNTL/Z.
Create JTAPI Subsystem in the CUE:
CUE(config)# ccn subsystem jtapi
CUE(config-jtapi)# ccm-manager address 177.1.10.10
177.1.10.20
Save the change to startup configuration and reload the
module for the new changes to take effect.
CUE(config-jtapi)# ctiport 1858 1859 3851 3852
CUE(config-jtapi)# ccm-manager username cue password cisco
Save the change to startup configuration and reload the
module for the new changes to take effect.
CUE(config-jtapi)# exit
Create VoiceMail Trigger:
CUE(config)# ccn trigger jtapi phonenumber 3850
Adding new trigger
CUE(config-trigger)# application voicemail
CUE(config-trigger)# maxsessions 2
CUE(config-trigger)# enabled
CUE(config-trigger)# exit
Modify the voicemail application to accept two simultaneous sessions:
CUE(config)# ccn application voicemail
Modifying existing application

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CUE(config-application)# maxsessions 2
CUE(config-application)# enabled
CUE(config-application)# exit
Modify the MWI Application to configure DN:
CUE(config)# ccn application ciscomwiapplication
Modifying existing application
CUE(config-application)# parameter strMWI_OFF_DN 1859
CUE(config-application)# parameter strMWI_ON_DN 1858
CUE(config-application)# exit
Create VoiceMail boxes for both of the Branch2 users:
CUE(config)# username dhume create
CUE(config)# username dhume phonenumber 3001
CUE(config)# voicemail mailbox owner dhume
CUE(config-mailbox)# no tutorial
CUE(config-mailbox)# greeting standard
CUE(config-mailbox)# enable
CUE(config-mailbox)# exit
CUE(config)# username jford create
CUE(config)# username jford phonenumber 3002
CUE(config)# voicemail mailbox owner jford
CUE(config-mailbox)# no tutorial
CUE(config-mailbox)# greeting standard
CUE(config-mailbox)# enable
CUE(config-mailbox)# exit
Set the PIN for the users as instructed:
CUE(config)# exit
CUE# username dhume pin 55555
CUE# username jford pin 55555
CUE#
Do a write memory and reload the CUE module after configuration done.

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Task 4.29 Messaging

Provision VPIM and test by sending messages back and forth between Unity
Connection and Unity Express using the following pre-provisioned DNS
information:

Server

IP Address

FQDN

DNS Server
Unity Connection
Unity Express

177.1.100.110
177.1.10.30
177.1.250.254

n/a
cuc.hq.ine.com
cue.br2.ine.com

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Configuration tasks at CUE:


Set the Name Server on CUE:
CUE#
CUE# conf t
Enter configuration commands, one per line. End with
CNTL/Z.
CUE(config)# ip name-server 177.1.100.110
NAME-SERVER Found at: 177.1.100.110
CUE(config)# exit
CUE# ping cuc.hq.ine.com
PING cuc.hq.ine.com (177.1.10.30) 56(84) bytes of
data.
64 bytes from 177.1.10.30: icmp_seq=1 ttl=62 time=6.38
ms
64 bytes from 177.1.10.30: icmp_seq=2 ttl=62 time=5.65
ms
64 bytes from uc1.corphq.ine.com (177.1.10.30):
icmp_seq=3 ttl=62 time=5.52 ms
64 bytes from cuc.hq.ine.com (177.1.10.30): icmp_seq=4
ttl=62 time=5.72 ms
64 bytes from uc1.corphq.ine.com (177.1.10.30):
icmp_seq=5 ttl=62 time=5.61 ms
Create a Location ID for both CUC and CUE, lets take the CUC Location ID is 1 and CUE
Location ID is 3.

CUE(config)# network location id 1


CUE(config-location)# name CUC
CUE(config-location)# email domain hq.ine.com
CUE(config-location)# voicemail extension-length max 4
CUE(config-location)# voicemail extension-length min 4
CUE(config-location)# voicemail vpim-encoding G711ulaw
CUE(config-location)# enable
CUE(config-location)# end

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CUE(config)# network location id 3


CUE(config-location)# name CUE
CUE(config-location)# email domain br2.ine.com
CUE(config-location)# voicemail extension-length max 4
CUE(config-location)# voicemail extension-length min 4
CUE(config-location)# voicemail vpim-encoding G711ulaw
CUE(config-location)# enable
CUE(config-location)# end
Configure the Location ID 200 as local to CUE:
CUE(config)# network local location id 3

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Configuration task at CUC:


Set the Name Server at CUC CLI:
admin:set network dns primary 177.1.100.110
***
W A R N I N G
***
This will cause the system to temporarily lose network
connectivity
Continue (y/n)?y
admin:
admin:utils network ping cue.br2.ine.com
PING cue.br2.ine.com (177.1.250.254) 56(84) bytes of data.
64 bytes from cue.br2.ine.com (177.1.250.254): icmp_seq=0
ttl=62 time=5.97 ms
64 bytes from cue.br2.ine.com (177.1.250.254): icmp_seq=1
ttl=62 time=5.79 ms
64 bytes from cue.br2.ine.com (177.1.250.254): icmp_seq=2
ttl=62 time=5.84 ms
64 bytes from cue.br2.ine.com (177.1.250.254): icmp_seq=3
ttl=62 time=5.83 ms
--- cue.br2.ine.com ping statistics --4 packets transmitted, 4 received, 0% packet loss, time
3004ms
rtt min/avg/max/mdev = 5.794/5.861/5.971/0.085 ms, pipe 2

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Check out the local VPIM Location which already exists:

Navigate to Networking > VPIM Locations:

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Click on 'Add New' and create new VPIM location:

After saving some extended configuration items would be on the same config page,
configure those as below:

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Task 4.30 Contact Center

Use the script included with your student files for this lab to build an application in
UCCX that can be accessed by ringing DN 1000
Use 2 CTI Ports that begin with DN 1891
You may use Resource Groups for your Agents, you do not have to use skills
Provision the following phones as agents in the proper CSQ, and assign to them
the IP Phone Agent XML service:
o CorpHQ Phone2 (x1002)
o Branch1 Phone1 (x2001)
o Branch2 Phone2 (x3002)
Troubleshoot the script as it has errors, and get it to route the calls to the agents
properly
5pts

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Modify the CM Configuration, Subscriber is set there as default, we need to modify


it to Publisher as Publisher is the primary call server:

Navigate to Application > Script Management:

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Click on 'Upload New Scripts' to upload the provided script from the Startup
Configuration files provided to students:

Choose the script on the pop-up window and then select 'Upload':

Then 'Refresh' the script and click on 'Yes' on the next page.

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Now login to UCCX Server using RDP:

Open the Cisco Unified CCX Editor and login with the credential of the user
'uccxadmin':

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Open the script 'ICD.aef' from 'Script Repository > Default' and validate from
'Tools > Validate' to check for any scripting error:

The default script provided contains an error 'queueLoop not exist'. If we check the script there is
one Label with the name 'queueLoops', so we need to modify it accordingly.

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Right click on 'Goto' step then 'Properties' and choose the correct label
'queueLoops':

Right click on 'Select Resource' step and modify the value of 'Connect' from 'No' to
'Yes':

Apply and save the script on desktop using the same name 'ICD.aef'. We need to reupload the
script using Application Administration.

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Navigate to Application > Application Management:

Click on 'Add a New Application':

Choose 'Cisco CRS Application' then click 'Next':

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Configure the 'Application' page and save:

Click on 'Add a New Cisco Unified CM Telephony Call Control Group':

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Configure the Call Control Group:

Save the group.

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Click on 'Cisco Unified CM Telephony Triggers' and then on 'Add a New Cisco
Unified CM Telephony Trigger' to create a new Trigger:

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Configure the new Trigger:

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Save the Trigger.


Now jump to CUCM to finish some agent related tasks, add the IPCC extension for
CorpHQ Phone2 (x1002), Branch1 Phone1 (x2001) and Branch2 Phone2(x3002).
For CorpHQ Phone2 is shown below:

Configure the same for rest of the phones as mentioned and save.

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Also assign all these three users to the group 'Standard CCM End User' and
'Standard CTI Enabled':

Navigate to Device > Device Settings > Phone Services:

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Click on 'Add New' and create a service named 'UCCX IP Phone Agent Login':

The Service URL is: http://177.1.10.40:6293/ipphone/jsp/sciphonexml/IPAgentLogin.jsp

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Assign this service to phone x1002, x2001 and x3002. One of them is shown below,
repeat the same for rest of them. Navigate to Phone with extension x1002. Select the
'Subscribe/Unsubscribe Services' from the top right selection box and then click
'Go':

Select the service 'UCCX IP Phone Agent Login' from the service selection and click
'Next':

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We dont need to input anything on the following page, just click on 'Subscribe' and close
the page:

Repeat the same for the rest two phones (x2001 and x3002).
Associate the DN 1002, 2001 and 3002 with the RM JTAPI user UCCXAgents:

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CorpHQ site also needs to have a Transcoder which will be used when Branch1 and Branch2
phones are dialing 1000 and accessing IVR over G.729 coded.
On the CorpHQ Router:
!
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 7.0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 3 register CorpHQ-HW-XCode
!
dspfarm profile 3 transcode universal
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 1
associate application SCCP
!

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At CUCM:

Now jump to UCCX again and navigate to Subsystems > RmCm:

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Click on 'Resources', all three users we have configured on CUCM should be shown
up:

Click on 'Resource Group' and then 'Add a new Resource Group':

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Give a suitable name and save the Resource Group, we choose the word 'RG-LAB':

Click on 'Contact Service Queues' and then 'Create a new Contact Service Queue':

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Configure the new 'Contact Service Queue' where the CSQ name is 'TVOICE' (as
we have noticed on the script as it defined), click 'Next':

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On the next page choose the right Resource Group we have just created and then
click on 'Add':

Click on 'Resources' again and choose the Resource Group = 'RG-LAB' for all
resources:

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Repeat the same for rest of the users.

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