Tech 3

CCIE-VOICE-LABS.COM VOICE-LABS.
NET Lab 3: 22-June-10

CCIE-VOICE-LABS.COM VOICE-LABS.NET Lab 3: 22-June-10




Note:- NO CORE KNOWLEDGE NOW IN THE LAB




3.2 CISCO CALL MANAGER EXPRESS

Customize phone background on CUCM.

Create a shared line 4003 between site C phone 1 and site C phone 2. The phones should
be able to barge in on an active call. Allow site C phone 1 to make the call private when
desired.

CME shared line, ensure 5 concurrent calls can be made into the DN. But STC phone 1 can
only accept 2 inbound calls on this line at a time while STC phone 2 can accept 4 inbound
concurrent calls.

OR

* Shared line 4003 on SiteC PH1 and SiteC PH2
* Max 5 concurrent on shared DN 4003
* SiteC PH1 max incoming calls 4
* SiteC PH2 max incoming calls 2
* SiteC PH1 enable privacy button

Call Park: 4300 4302

CBarge: on phone 1 and 2

Live Record: 4250

Configure a privacy button on 3rd line of phone 1

Monitor line status of 4001 from 3rd line of 4002. When 4001 is off hook or in DND mode
3rd line of 4002 should be red.

Configure a shared line 4003 on SC Phone 1 and Two. This shared line can receive
maximum of 5 incoming calls. Phone 2 can receive maximum 4 incoming calls.


Section 4: Voice Gateways and Signaling

4.1 HQ IOS MGCP T1-PRI gateway

All HQ IP Phones must be able to make local call by dialing 9 followed by 7 digits. PSTN
should send 7 digits calling number 202xxxx along with calling name. Also called-party-
number-type should be subscriber for these local calls.

All HQ IP phones must be able to make Long Distance calls with the area code (972). For
this type of calls, SB router should be used first. If Site B router isnt available then HQ
router should be used. 10 digits calling party number should be sent out to PSTN for these
calls.

Use Local Route Group for the above calls to use HQ local voice gateway.

4.2 SiteB IOS H323 T1-PRI gateway

All SiteB IP Phones must be able to make local call by dialing 9 followed by 7 digits. PSTN
number-type should be subscriber for these local calls. SB router should be selected 1st to
route these calls.

If SB router isnt available, calls should go thru via HQ router. Calling party number for
these calls should be (1972303xxxx).

All SB IP Phones must be able to make International calls by dialing 9011 as leading digits.
SB router should be selected 1st to route these calls. If SB router isnt available, calls
should go thru via HQ router. Calling party number for these calls should be
(+1972303xxxx).

4.3 Site C CME gateway

All Site C IP Phones must be able to make local calls by dialing 9 followed by 8 digits. PSTN
number-type should be subscriber for these local calls.

All Site C IP Phones must be able to make International calls by dialing 900 as leading
digits. PSTN provider expects no leading digits in Called party number field. Calling
number for these calls should be (+8522404xxxx).


Section 5: CUCM Call Routing

PSTN access code for all IP phones 9

Country code for US 1

Country code for Hong Kong 852

National code for HQ and SiteB IP phones 1

International code for HQ and SiteB IP Phones 011

International code for SiteC IP Phones 00

5.1 CUCM Call Routing HQ Gateway

HQ PSTN provider specifications are as follows,

1) HQ PSTN provider expects proper information in called party number and
called party number type fields.

2) Called party number and called party number type information must be
set in ISDN setup messages. (Subscriber for local, National for long distance
and International for International calls).

3) You MUST not use leading digit information to signal national (1) or
international (011) calls.

4) If HQ Phone 1 makes international call to SiteC Phone 1 901185224044001,
service provider expects 85224044001 in called party number field and
International in called party number type field to route this call
properly.

5) Unknown Called party number type field is only accepted for 911
emergency calls.

By considering the above specifications, configure following requirements,

1) All HQ IP phones can make local PSTN calls by dialing 9 followed by 7 digit
PSTN number. Second digit after the access code can be anything between 2
to 9. Rest of the digits can be anything between 0 to 9. For such local calls,
PSTN should send 7-digit calling number 202xxxx along with calling name.
Also, called party number type should be set to subscriber for these calls.

Only HQ gateway should be selected and no redundancy is required.

2) All HQ IP phones can make International calls by dialing 9 followed by 011
then country code and variable length dialing digits. Calling number for such
calls should be US country code leading + i.e. - +1408202xxxx.
International calls should use only HQ gateway and no redundancy is
required.
Also, called party number type should be set to international for these
calls.

3) Configure local route group for both the type of calls mentioned above so
that it uses only HQ gateway for call routing.

(3 points)

5.2 CUCM Call Routing SiteB Gateway

SiteB PSTN provider specifications are as follows,

1) HQ PSTN provider uses leading digits in the called number to signal nonlocal
calls. 1 for national and 011 for international calls.

2) Called party number type information can be ignored except local calls
for which provider expects subscriber as Called party number type field.

3) If SiteB Phone 1 makes international call to SiteC Phone 1
901185224044001, service provider expects 01185224044001 in called
party number field and to route this call properly.

emergency calls.


1) All SiteB IP phones can make local PSTN calls by dialing 9 followed by 7
digit PSTN number. For such local calls, PSTN should send 7-digit calling
number 404xxxx along with calling name. Only SiteB gateway should be
selected and no redundancy is required.

2) If SiteB IP Phone makes national call to numbers in 408 area code, HQ
gateway should be selected to route these calls. 10-digit Calling number
1972303xxxx should be sent out to PSTN along with calling name.

3) For above calls, if HQ gateway is not reachable, it should use SiteB local

gateway. 10-digit Calling number 1972303xxxx should be sent out to PSTN
along with calling name.

(3 points)

5.3 CUCM Call Routing SiteC Gateway

SiteC PSTN provider specifications are as follows,

1) SiteC PSTN provider expects proper information in called party number
and called party number type fields.

2) Called party number and called party number type information must be
set in ISDN setup messages. (Subscriber for local, National for long distance
and International for International calls).

3) If SiteC Phone 1 makes international call to HQ Phone 1 90014082022001,
service provider expects 14082022001 in called party number field and
International in called party number type field to route this call
properly.

emergency calls.


1) All SiteC IP phones can make local PSTN calls by dialing 9 followed by 8-
digit PSTN number. For such local calls, PSTN should send 8-digit calling number
2404xxxx along with calling name. Also, called party number type should be set
to subscriber for these calls. Only SiteC gateway should be selected and no
redundancy is required.

2) All SiteC IP phones can make International calls by dialing 9 followed by 00
then country code and variable length dialing digits. Calling number for such calls
should be Hong kong country code leading + i.e. - +18522404xxxx.
International calls should use only SiteC gateway and no redundancy is required.
Also, called party number type should be set to international for these calls.

- CME Call Routing SiteC Gateway
- Local calls (send 8 digits callerid)
- International calls (send callerid e164)

(4 points)

5.4 CUCM Call Routing + dialing consideration

Select the missed calls menu and highlight the number +1972xxxxxxx and use Dial soft-
key to dial back to PSTN.

This call should use SiteB gateway first. If SB gateway isnt available then it should be
routed via HQ gateway.

Once the call is connected it should show TO 3xxxxxxxon SB Phone 1 display and From
303XXXX on PSTN Phone Display.

When a call goes through HQ, called ID should be 10 digits

Gatekeeper Call Routing

Register the CUCM and CME to match the following outputs that are in bold:

HQ-RTR# sh gatekeeper endpoints

GATEKEEPER ENDPOINT REGISTRATION
================================
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--------------- ----- --------------- ----- --------- ---- -----
142.100.64.11 1720 142.100.64.11 32807 GK VOIP-GW
H323-ID: GK-Trunk_1
Voice Capacity Max.= Avail.= Current.= 0
142.100.64.12 1720 142.100.64.12 32787 GK VOIP-GW
H323-ID: GK-Trunk_2
142.102.66.254 1720 142.102.66.254 65137 GK H323-GW
H323-ID: CUCME
Total number of active registrations = 3

HQ-RTR# sh gatekeeper gw-type-prefix

GATEWAY TYPE PREFIX TABLE
=========================
Prefix: 852*
Zone GK master gateway list:
142.102.66.254:1720 CUCME

Prefix: 1*
Zone GK master gateway list:
142.100.64.12:1720 GK-Trunk_2
142.100.64.11:1720 GK-Trunk_1

You are not allowed to use default tech-prefix, zone subnet, and static alias commands.
SiteC should use its loopback address for all communications with the gatekeeper HQ
phones should be able to call SiteC phones by dialing 4 digits internal extensions.

SiteB phones are not required to dial SiteC phones this time.

Use 852 as tech-prefix to make calls to SiteC phones and 1# to make calls HQ phones from
SC.

- Allow 4 digit dialing from HQ/SB to SC.
- Allow 4 digit dialing from SC to HQ/SB.
- If GK or WAN is down calls should be sent through the local gateway as backup.
(send callerid e164).

Gatekeeper troubleshooting section

We have a customer that places high call volume to UK resulting in high cost.
In order to avoid high toll charges with these calls, the customer would like to send the calls
via the backbone gatekeeper.

Configure so that the calls to UK are sent via the backbone gatekeeper.

Backbone Gatekeeper info:
GK=BBGK Domain: cisco.com IP Address: 157.1.26.30

-connection HQ to an external GK is broken.
-you have no access to the external GK.
-List your troubleshooting work on a notepad file

Write a report on the troubleshooting steps that you performed to accomplish this.

Route calls from HQ through GK over to SiteC and back (Only calls between HQ and SiteC)

When calls from HQ to SiteC fail to go through the GK, route across the PSTN

Calls to +44 should be routed through the Backbone GK


Section 6: Codec Selection

Intra site calls should be G.711 and calls between sites should be G.729.

Show gatekeeper calls, allocated bandwidth for each call should be 16kbps.

Section 7: Media Resource Management

When SiteB IP phones or PSTN users are put on hold, configure local routers to stream
G711 multicast MOH from router flash.

You can use music-onhold.au file in router flash for this multicast requirement.

Call Park for HQ/SB with redundancy configured with null partition (range 2900 - 2902).

CBarge for shared line on SiteC PH1 and SiteC PH2.

(3 points)

Section 8: QoS

It is not restricted to use auto-qos however there should not be any impact of the
configuration generated by auto-qos on functionality of the lab. If there is any
such impact, this section will not be marked.

8.1 Switch QoS

Ensure CoS 5 is mapped to EF 46.
On port go 1/0/1 which is connected to HQ router, guarantee 16k for MGCP signaling traffic.
Excess traffic should be marked to DSCP 8 and then transmitted.

8.2 Link fragmentation and Interleaving

There is a 384K link between HQ and STB and 768K between HQ and STC. Configure
FRF.12 at a 10MS sampling rate.


Section 9: Voice Mail Integration

You should check MWI functionality for Cisco Unity connection as well as Cisco
Unity Express. Make sure to clear MWI once you test the same in the lab. Also,
make sure that voicemail pilot numbers for both Cisco unity Connection as well as
Cisco unity express are reachable from PSTN.

9.1 Cisco Unity Connection Integration and Configuration

Cisco Unity Connection is pre-configured and integrated with CUCM with following
Configuration,

Voicemail Pilot 2220

Voicemail ports 2221-24

MWI On 1998

MWI off 1999

AXL username administrator

AXL password ccievoice

Import HQPh1-HQPh3, SBPh1-SBPh2. You must import users from CUCM. Use existing
users in end users list.

Set user passwords to 246810

Pilot Number for voice is reachable from PSTN

Make sure CUC/CUE voicemail greetings and MWI work.
Test calls from HQ/SB to SC and vice versa.

For HQ Phone1 make sure if PSTN caller left a voicemail the user can hear the
calling number of the PSTN caller and the message disposition time before playback
the message.

OR

When PSTN phone leaves a message for HQ PH1
HQ PH1 message button is pressed, the time and extension should be heard before
the message is played.

(2 points)

9.2 Cisco Unity Express Initial Configuration

Cisco Unity Express is set to factory default settings. You need to run through the initial setup wizard
to configure following settings,

IP Address : 142.102.66.253

Hostname : CUE

Domain name : ccievoice.com

DNS : not required

NTP : 142.102.64.254
GUI web administrator : administrator

GUI web password : ccievoice

9.3 Cisco Unity Express configuration and CUCME integration

Change CUE license file to CUCME and integrate the same with CUCME
Following license files available FTP server .

FTP Login details

FTP Server IP : same candidate pc (access via VNC)

FTP User name : administrator
Pssword : ccievoice

cue-vm-license_12mbx_cme_7.1.2.pkg
cue-vm-langpack.nme.7.0.2.pkg
cue-vm-k9.nme.7.1.2.pkg
cue-vm-installer-k9.nme.7.1.2.prt1
cue-vm-en_US-langpack.nme.7.1.2.prt1

Note :(Already CTI port integrated and registered with CUCM . Once upload new
license delete cti port configuration.


9.4 Advanced CUE

Calls from HQ and SB to CUE voicemail should succeed

Configure Live Record for SiteC users. Live Record Pilot 4250.

CUE Live Record; make sure you are able to record a conference call by pressing
live record softkey.

Section 10: UCCX Applications

Configure two IPCC extensions 1 on HQ Phone 2 ( ext: 2102) and 1 on SiteB Phone 2 (ext:
3102).
Incoming calls from +44 number will be serviced by 2102 phone only and all other calls will
be routed to the normal queue.

If the call hits the queue, the user should hear the defaults prompt, all are agents are
currently busy...
Then they should hear their position in the queue.

OR

If the call hits the queue, the user should hear the default prompt, all agents are currently
busy
Need to hear the user position in the queue just after the queue prompt. There is
queue_prompt.wav saved on the test PC and reads Your position in the queue is.
If there are calls in the queue it should play

First call "your position in the queue is 0" / second call "your position in the queue is 1"

OR NEW Question

Create an script in such a way so that when users call in they hear Thank you for calling
and immediately after that it should play All of our representatives are busy at this time
please stay on the line someone will be with you shortly.

If there are zero call in the queue, the script should play There are currently X calls ahead
of you.

In other words lets say if the first caller calls in, He/She should hear There are currently
ZERO calls ahead of you. If the 2nd call comes in while the first call is in the queue, it
should play There are currently ONE calls ahead of you.


You are asked by your customer to generate the necessary prompts to fulfill the above
mentioned requirements by using the UC voice recording tools available on your POD.

Note: No agents need to be logged in. You dont even need to configure an extension for
IPCC.

(5 Marks)

CME Presence

SiteC PH2 button 3 should monitor the status of SiteC PH1 primary line when off-hook and
DND status. This should function also as a speed dial button to call SiteC PH1

Section 12: High Availability

12.1 Site B router high availability

You cannot use CME SRST; you must configure Call-Manager-FallBack

Make sure that voicemail functionality is restored in event of WAN failure. Voicemail forwarding
feature should work between IP phones as well as PSTN calls. When such forwarded call comes to
Cisco Unity connection, it should play users personal greeting. You are not allowed to use alternate
extension to achieve this

Make sure that the local, international and emergency calls work fine during SRST
operation.

911 (send 10 digits callerid)
local (send 7 digits callerid)
International (send callerid e164)
Make sure 4 digit call should work between SB-HQ & SB-SC during WAN failure
(send callerid e164).

Call Forward Unregistered

If HQ or SC user calls the SB Phone and if it is not registered, he should be
forwarded to SB Phone over the PSTN (For HQ to use HQ GW to call the Site B
E.164 number. For SC to use the GK to call as an international number.)
Provided a .2screenshot of a phone at siteC and the phone should display:
Forwarding from: +19723033001

OR

Make sure that HQ/SC Phones are be able to call SB PH1 using 4 digit dialing in
event of a WAN failure.
When you call from HQ Phones calls should be routed through HQ Gateway.
When you call from SC Phones calls should be routed through the GK and then HQ
Gateway.

Provided 2 screenshots of SiteB Phone 1:

Forward
(2001)
For:+19723033001 (3...)
By :+19723033001 (3...)

Forward
(4001)
For:+19723033001 (3...)
By :+19723033001 (3...)


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NET Lab 3: 22-June-10

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