ievo wb2 lab4 solutions

8/10/2019 IEVO WB2 Lab4 Solutions

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INE Voice Workbook Volume IILab 4 Solutions


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INE Voice Workbook Volume II Lab 4 Solutions ii

Table of Contents

Lab 4 ......................................................................................................................... 1Task 4.1 - Network Infrastructure ............................................................................................................. 1

Task 4.2 - Network Infrastructure ............................................................................................................. 9

Task 4.3 CUCM Server and Phone Basics ........................................................................................... 13



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 .................................................................................................................. 247Task 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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INE Voice Workbook Volume II 1

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 withthe CorpHQ router, and every router should retain the proper time after beingrebooted 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 Timeo 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 server 177.1.254.1

!

Sync Branch2 Router with NTP master clock:

!


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 aComputer 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 forthat 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 optionthat 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!


description == Connection CorpHQ-Phone2 ==

switchport trunk encapsulation dot1q




spanning-tree portfast!

Configure switchports on Branch1 Router:

!

interface FastEthernet0/1/3

description == Connection Branch1-Phone1 ==




spanning-tree portfast

end

!


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Configure switchports on Branch2 Router:

!







!







!

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:

!


network 177.2.11.0 255.255.255.0



!


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Also the Same on Branch2 Router:

!


network 177.3.11.0 255.255.255.0



!

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, ensuringthat the CUCM Publisher is the primary server registered to and that theSubscriber can always take over as a CPE should failover occur, and ensure thefollowing 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 forapproximately 20 mins for all phones to complete their firmware upgrades)

o CorpHQ Phone1 should use DN 1001 and use SCCP firmware and beallowed to place International calls

o CorpHQ Phone2 should use DN 1002 and use SIP firmware and be

allowed to place National callso 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 beallowed to place International calls

o Branch2 Phone2 should use DN 3002 and use SCCP firmware and beallowed to place National calls

o When any phones place calls within any given site ensure that they shoulduse the G.711 codec; when any phones place calls between any two sitesensure 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 Partition

CSS_CorpHQ-INTL PT_US-INTL

PT_US-NATLPT_Internal_DNs

CSS_CorpHQ-NATL PT_US-NATLPT_Internal_DNs

CSS_Branch1-INTL PT_US-INTLPT_US-NATL

PT_Internal_DNs

CSS_Branch2-INTL PT_NL-INTL

PT_NL-NATLPT_Internal_DNs

CSS_Branch2-NATL 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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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

!

sccp ccm group 1

bind interface Loopback0

associate ccm 1 priority 1


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

!


description == FR To BR2

ip rsvp bandwidth 184!


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On the Branch1 router:

!

voice-card 0


!!




sccp

!

sccp ccm group 1




associate profile 2 register Brnch1-G729-MTP


!

!


codec g729r8

rsvp



!

!

interface Serial0/0/1:0

ip rsvp bandwidth!


description == FR To HQ


!

On the Branch2 router:

!

voice-card 0


!

!


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


associate ccm 1 priority 1associate ccm 2 priority 2


!


codec g729r8

rsvp



!

!

!

!interface Serial0/0/1:0

ip rsvp bandwidth

!



!

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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Ensure that all IP Phones show their PSTN DID number in the top right of their

displayo 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 phoneringing) both display the full name (only FName LName) of the person that iseither 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, dhumeandjfordon 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 theconference

If any member of the conference puts the conference on hold, MoH should not beallowed 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

resources3pts


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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


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 g711alawcodec g729ar8

codec g729abr8

codec g729r8

codec g729br8

codec ilbc

maximum sessions 1


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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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 neverallow 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 andmulticast MoH for the Branch1 PSTN gateway using the G.711 codec

o Ensure that no additional RSVP reservations are made when streamingmusic 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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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 respectivelyfor those lines

Anytime Jack goes off-hook on his intercom line, it should automatically connectto Desmonds intercom line, and vice-versa as well

Every other phone should be able to go off-hook and call to these two phone'sand 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 iscustomizable:


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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 alwaysable to park calls anywhere in the DNrange of 5300-5499, and they all must be in the 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 1second 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 notconfigure 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:


no ip address

encapsulation hdlc



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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 codecnow:

!voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 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-patternwhen 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 ipv4:177.1.10.10dtmf-relay sip-kpml rtp-nte

!

!


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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Find the T1 slot/sub-slot:

Branch1#sh inventory


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





!



!

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




%CONTROLLER-x-UPDOWN: Controller T1 0/0/0, changed state to up


Now configure the created serial interface to allow ie facility:


no ip address

encapsulation hdlc



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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 codecnow:






!

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

!




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work for H.323 as instructed. We need to configure this on the dial-peers; we shall create these


dial-plans.

!


preference 1

voice-class codec 1

voice-class h323 1


dtmf-relay h245-signal

!


preference 0

voice-class codec 1

voice-class h323 1

session target ipv4:177.1.10.20dtmf-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 h323section as voice service

voip:

!

voice service voip

h323

h225 display-ie ccm-compatiblecall 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 PSTNwould remain up with RTP media

o Ensure that if for some reason the CUCM Sub server isn't responding fastenough, that call setup is retried quickly to the CUCM Pub (within 2seconds of CUCM Sub setup failure)

o Ensure that DTMF works properly over the link back to CUCM using amethod that carries not only the tone's frequency, but also the tone'sduration

3pts


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Find the E1 slot/sub-slot:

Branch2#sh inventory


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




isdn switch-type primary-net5

!



!

Now configure the controller T1 0/0/0:CorpHQ(config)#controller e1 0/0/0

CorpHQ(config-controller)#linecode hdb3

CorpHQ(config-controller)#framing crc4CorpHQ(config-controller)#clock source line




%CONTROLLER-x-UPDOWN: Controller E1 0/0/0, changed state to up


Now configure the created serial interface to allow display ie:


no ip address

encapsulation hdlc


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isdn switch-type primary-net5


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 codecnow:






!

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

!



work for H.323 as instructed. We need to configure this on the dial-peers; we shall create these


dial-plans.


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!


preference 1

voice-class codec 1

voice-class h323 1

session target ipv4:177.1.10.10dtmf-relay h245-signal

!


preference 0

voice-class codec 1

voice-class h323 1


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 h323section as voice service

voip:

!

voice service voip

h323

h225 display-ie ccm-compatible

call preserve limit-media-detection

!


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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 phonewhen 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 CorpHQrouter 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

Settingsshould 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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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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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 Phones3. 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 onthe information in the three tables below

Ensure these PSTN Patterns globalize every Called Party Number to aproperFull E.164 (including preceding +) number format before the call everreaches a Route List in CUCM

o NOTE:Emergency numbers need to be globalized to match the globalroute 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 withthe 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 1Call 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

National/LD 1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW 11 ANI Digits National

(2) Branch1 GW 11 ANI Digits Subscriber

International 011 + Variable length (1) CorpHQ GW CC+ 10 ANI Digits International

(2) Branch1 GW CC+ 10 ANI Digits International

5pts4.2 Dialing From Branch1 Phones out to the PSTN


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

National/LD 1[2-9]XX[2-9]XXXXXX (1) Branch1 GW 11 ANI Digits National

(2) CorpHQ GW 11 ANI Digits Subscriber

International 011 + Variable length (1) Branch1 GW CC+ 10 ANI Digits International

(2) CorpHQ GW CC+ 10 ANI Digits International

5pts


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4.3 Dialing From Branch2 Phones out to the PSTN


Emergency 112 (1) Branch2 GW 10 ANI Digits Subscriber

Local [1-8]XXXXXX (1) Branch2 GW 7 ANI Digits Subscriber


Local 020XXXXXXX (1) Branch2 GW 10 ANI Digits Subscriber


National/LD 0[1-8]XXXXXXXX (1) Branch2 GW 10 ANI Digits National



(2) CorpHQ GW CC+ 9 ANI Digits

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 Groupconcept.

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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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-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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Local [2-9]XX[2-9]XXXXXX (1) CorpHQ GW 10 ANI Digits Subscriber

(2) Branch1 GW 11 ANI Digits National



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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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redundant Branch1 Gateway (which is National type of calls in actual):


We have already assigned it.


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Assign the CdPTP into the Branch1 Gateway:

On the CorpHQ Router:

!

!


rule 1 /^[2-9].........$/ /9&/

rule 2 /^1[2-9].........$/ /9&/

rule 3 /^011.*/ /9&/

!

voice translation-profile Prefix_9_DNIS

translate called 9

!


rule 1 /^2065011...$/ /&/ type any subscriber

rule 2 /^15126022...$/ /&/ type any national

rule 3 /^31207033...$/ /&/ type any international!


rule 1 // // type any subscriber

!


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voice translation-profile Add_ANI_Type_LOCAL


translate called 21

!

dial-peer voice 101 voiptranslation-profile incoming Prefix_9_DNIS

voice-class codec 1

incoming called-number .

!


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:

!


rule 1 /^[2-9].........$/ /9&/

rule 2 /^1[2-9].........$/ /9&/rule 3 /^011.*/ /9&/

!


translate called 9

!



rule 2 /^2065011...$/ /1&/ type any national

rule 3 /^31207033...$/ /&/ type any international

!


rule 1 // // type any national

!


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voice translation-profile Add_ANI_Type_NATL


translate called 22

!

dial-peer voice 12 potstranslation-profile outgoing Add_ANI_Type_NATL


port 0/0/0:23

forward-digits 11

!


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National/LD 1[2-9]XX[2-9]XXXXXX (1) CorpHQ GW 11 ANI Digits National

(2) Branch1 GW 11 ANI Digits Subscriber



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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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redundant Branch1 Gateway (which would be Local type of calls in actual):






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!


rule 1 /^12065011...$/ /&/ type any nationalrule 2 /^5126022...$/ /1&/ type any national


!


rule 1 // // type any national

!



translate called 22


translation-profile outgoing Add_ANI_Type_NATL


port 0/0/0:23

forward-digits 11

!

On the Branch1 Router:





!


rule 1 // // type any subscriber!



translate called 21


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!




port 0/0/0:23forward-digits 10

!


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International 011 + Variable length (1) CorpHQ GW CC+ 10 ANI Digits International

(2) Branch1 GW CC+ 10 ANI Digits International



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Translation Pattern for International Calls:


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CSS and Partition for Called Party Transformation Pattern for CorpHQ Gateway

(already created):


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Create Called Party Transformation Pattern for outgoing International calls for

CorpHQ and Branch1 gateway:






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!


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 called 23


translation-profile outgoing Add_ANI_Type_INTL

destination-pattern 9011T

port 0/0/0:23

prefix 011

!


!voice translation-rule 23


!





!



translate called 23

!



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port 0/0/0:23

prefix 011

!


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4.17 Dialing from Branch1 Phones out to the PSTN


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-to-

PSTN-RP

Translation Pattern (already created):

We have already created when we were configuring the CorpHQ 911 of the pattern 911


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_PSTN-

SLRG


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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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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-peertowards PSTN/PRI.

!



!



!

voice translation-profile Add_ANI_Type_EMStranslate calling 10

translate called 20

!




port 0/0/0:23

forward-digits 3

!


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Local [2-9]XX[2-9]XXXXXX (1) Branch1 GW 10 ANI Digits Subscriber

(2) CorpHQ GW 11 ANI Digits National


We have already the CSS with the name CSS_TP-to-PSTN-RPand 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


The Route Pattern is also created of the pattern: \+!


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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Branch1 Gateway:


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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-GWat Branch1 Gateway.

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

Already assigned the CSS CSS_CdPTP-CorpHQ-GWat CorpHQ Gateway.


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!





!



!



translate called 21




port 0/0/0:23

forward-digits 10

!



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National/LD 1[2-9]XX[2-9]XXXXXX (1) Branch1 GW 11 ANI Digits National

(2) CorpHQ GW 11 ANI Digits Subscriber


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


Of the pattern:\+!


RL name: RL_PSTN-SLRG


(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


Branch1 Gateway (already created):

Pattern: \+.1!


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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.


CSS CSS_CdPTP-CorpHQ-GW is already assigned on the CorpHQ gateway.

In the Branch1 Gateway (already created):

!



rule 2 /^2065011...$/ /1&/ type any national


!


rule 1 // // type any national!



translate called 22

!

translation-profile outgoing Add_ANI_Type_NATL


port 0/0/0:23

forward-digits 11

!


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In the CorpHQ Router (already created):

!

voice translation-rule 11rule 1 /^2065011...$/ /&/ type any subscriber



!



!



translate called 21!




port 0/0/0:23

forward-digits 10

!


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Partition name: PT_TP-to-PSTN-RP, CSS name: CSS_TP-to-PSTN-RP

Translation Pattern for International Calls (already created):

Pattern: 9011.!


Pattern: \+!


Route List name: RL_PSTN-SLRG

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


CSS and Partition for Called Party Transformation Pattern for CorpHQ Gateway

(already created):



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Create Called Party Transformation Pattern for outgoing International calls for

CorpHQ and Branch1 gateway (already created):

Pattern: \+.!


We have already assigned it, CSS name: CSS_CdPTP-Branch1-GW


We have already assigned it, CSS name: CSS_CdPTP-CorpHQ-GW

On the Branch1 Router (already created):

!



!




rule 3 // // type any international!



translate called 23

!




port 0/0/0:23

prefix 011!


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Emergency 112 (1) Branch2 GW 10 ANI Digits Subscriber


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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Pattern: \+!





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Create Called Party Transformation Pattern for outgoing Emergency calls through

Branch2 gateway:


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!



!


rule 1 // // type any subscriber!

voice translation-profile Add_ANI_Type_EMS


translate called 20

!




port 0/0/0:15

forward-digits 3!


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Local [1-8]XXXXXX (1) Branch2 GW 7 ANI Digits Subscriber


Local 020XXXXXXX (1) Branch2 GW 10 ANI Digits Subscriber



CSS name: CSS_TP-to-PSTN-RPand Partition name: PT_TP-to-PSTN-RP

Translation Pattern for Local Calls:


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Another pattern:


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Pattern: \+!




(already created):



(already created):



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Branch2 gateway:


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In the Branch2 Router:

!


rule 1 /^703....$/ /0&/rule 2 /^0[1-8]........$/ /0&/

rule 3 /^00.*/ /0&/

!


translate called 9

!




rule 3 /^15126022...$/ /&/ type any international!



!



translate called 21

!


translation-profile incoming Prefix_0_DNISincoming called-number .

!



destination-pattern 0[1-8]......$

port 0/0/0:15

forward-digits 7

!


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On the CorpHQ Router (already created):

!


translation-profile outgoing Add_ANI_Type_INTLdestination-pattern 9011T

port 0/0/0:23

prefix 011

!


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National/LD 0[1-8]XXXXXXXX (1) Branch2 GW 10 ANI Digits National



CSS name: CSS_TP-to-PSTN-RPand Partition name: PT_TP-to-PSTN-RP

Translation Pattern for National Calls (already created):

We already created the pattern: 00.[1-8]XXXXXXXXfor use in Local call, we can use the same

pattern for National type of calls.


Pattern: \+!


RL name: RL_PSTN-SLRG


(already created):

CSS name: CSS_CdPTP-Branch2-GWwith Partition: PT_CdPTP-Branch2-GW


(already created):

CSS name: CSS_CdPTP-CorpHQ-GW with Partition: PT_CdPTP-CorpHQ-GW


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Branch2 gateway:


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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Assigned CSS: CSS_CdPTP-CorpHQ-GW


!





!

ievo wb2 lab4 solutions

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