Share your CIPT1 v8.0 Experience

Thanx Voicetut Gr8 wrk.. 🙂

Follow the CCNP VOICE CIPT1 642-447 Quick Reference Guide 2edition.

hxxps://rapidshare.com/files/1694250237/CCNP_VOICE_CIPT1_642-447_Quick_Reference_2ed.rar

whats the accesibility to OS of CUCM ?
1. GUI and CLI
2. CLI
3. GUI.
4. none
5. OS can access by CISCO TAC only

CLI is not the same as the OS(kernel).

Answer: OS can access by CISCO TAC only

Anyone have this QuickRef as pdf? The JPGs are great but not easy to read on the ipad.

Thx!

pass today 825

Most questions from dump, just double check all questions with book/study guide some are not correct.

I remember this question:

What options are in enterprise parameters:

Phone URL
User Search Limit
Dependency Records

All of them.

Good luck, now to CIPT2

congrats cipt1now, can u share the dumps and reference book

thanks in advance……

Anonymous,
Check previous comments under old CCVP for CIPT1. Everything is there.

Is there a SimLab in this 642-447 Exam ??

CIPT1NOW can u tell us how many question with wrong answers in this dumps ???

Failed on 1 June Score 728
Passguide has wrong answers. Bought from official http://www.passguide.com.
30-40% are new.

sorry 1 July, not june

Hello everybody..

Try with this dump, I cheked it and I think the answer are OK:

http://www.examcollection.com/cisco/Cisco.ActualTests.642-447.v2011-07-14.by.Xenix.219q.vce.file.html

Hi All,

Does anyone know where i can get the cbt nuggets for voice tracks.

Many thanks in advance

You can use the CBT nugget for the previous CIPT v6.0

Hello friends

i need CUCM ISO for VMWARE ……Where can i find it…….

Help me plz

Did any 1 pass this test? pls update with the valid dumps .. thanks in advance 🙂 🙂

I am going to take this exam in next week

Require valid Dumps and suggestions.

Plz pass the comments who have passed this exam …

common guys no comments no helping material.

Voice tut we are in trouble…..After 7 month of launching new version 8 exam …Still we do not find valid material and dumps ….. Plz help us

CIPT1 v8.0 642-447….
any one has this paper’s dumps

what type of Questions has this exam? MCQ? D&D? Lab Sims?

1. Which two services need to be activated on a Cisco Unified Communications Manager for it to
function as the publisher in a cluster? (Choose two.)
A. CiscoDirSync
B. Cisco Extension Mobility
C. Cisco Messaging Interface
D. Cisco TFTP
E. Cisco CallManager
F. Cisco DHCP Monitor Service
2. @ anonymous
Answer is D & F

But my answer is A & E

Justification:

Examples of UFFs or subcriber services include the following:
■ Call Forward All (CFA)
■ Message Waiting Indication (MWI)
■ Privacy, Enable/Disable
■ Do Not Disturb, Enable/Disable (DND)
■ Extension Mobility Login (EM)
■ Hunt Group Login Status
■ Monitor (future use)
■ Device Mobility
■ CTI CAPF Status (Computer Telephony Integration, Certificate Authority Proxy
Function)

The services listed in Table 1-1 rely on the availability of the publisher server regardless of
the version of CUCM used.

Table 1-1 Publisher Server Required Services
Component Function
CCMAdmin Provisions everything
CCMUser Provisions user settings
BAT Provisions everything initiated by the Bulk Administration tool
TAPS Provisions everything initiated by the Tool for Auto-Registered Phone SupportCUCM
AXL Provisions everything initiated by the AVVID XML Layer service
AXIS-SOAP Enables and disables services through SOAP
CCM Inserts phones (auto-registration only)
LDAP Sync Updates end-user information ( you nedd dirsync serverice enabled)
License Audit Updates license tab

Plz comments it confusing

safari, where did you find that question #1, on which dump? Where ever it is from it is a horribly worded question. The correct answer is “none of the above.” In order for a CM server to function as the publisher it needs to be installed as the publisher. Stopping or starting Feature Services does not effect its operation as a publisher.

I’ve noticed in some of the dumps the actual questions weren’t captured properly; I think that is the case here.

@selsius
can u help me vit the dumps for this

@ selsius

It is present in Actual test ,pass guide , one of link is

http://www.examcollection.com/cisco/Cisco.ActualTests.642-447.v2011-07-14.by.Xenix.219q.vce.file.html

@M_of_D
by “vit the dumps” do you mean look for correct vs incorrect answers? If so sure, I’m scheduled to take this exam at the end of the month.

whats the exam cost of cipt1, please let me know thanks.

QUESTION 106
Which of these is a characteristic of the Cisco Unified Communications Manager Bulk Administration Tool in
Cisco Unified Communications Manager 6.0?
A. Cisco Unified Communications Manager Auto-Register Phone Tool is part of Bulk Administration.
B. Cisco Unified Communications Manager BAT needs to be installed from the plug-in page.
C. Cisco Unified Communications Manager BAT pages are available from the Serviceability page.
“Pass Any Exam. Any Time.” – http://www.actualtests.com 53
Cisco 642-447: Practice Exam
D. Cisco Unified Communications Manager Auto-Register Phone Tool requires a Customer Response System.
Answer: A or D

Chosse only one from two both are correct but ( i think best suitable is answer D but in dumps is A )

Q106
I would also prefer choose answer “d”, but like safari says, “a” must be also correct.

Is this Xenix.219q.vce file enough to pass the exam ?

May I know how to open the vce file? Where do i get free vce software?

another confusing question

If no SIP dial rules are configured on an IP phone, at what point in the collection of digits does a Type A SIP
phone send digits to the Cisco Unified Communications Manager? (Choose two.)
A. when the interdigit timer expires
B. when the collected digits match a SIP dial rule
C. when the user presses the Dial softkey
D. as each digit is collected (it is sent for analysis)
E. when the user presses the # key
Answer: AC or CE

In this Question i prefer C&E

on 4-98 you can read this:

User input on Type-A SIP Phones / No SIP Dial Rules Configured on the Phone:
• Phone accumulates all user input events until # or Dial
softkey is pressed (similar to cell phones).

@ Andew

which book you refered , Please share

@ safari,

look to this thread :
http://www.examcollection.com/cisco/Cisco.ActualTests.642-447.v2011-06-22.by.Alex.216q.vce.file.html

here the post from: icciev from Unknown – Jul 30 2011, 6:40 PM

Question: 53
The Ajax Corporation is designing an IP telephony network using Cisco MCS 7845 Series
servers, each one capable of supporting 7500 devices. The design must meet these
requirements:
♦ be cost-effective
♦ support up to 7500 phones
♦ provide a minimal level of redundancy
Which configuration will meet Ajax Corporation needs?
A. two Cisco Unified Communications Manager servers:
• 1 publisher and TFTP server combined
• 1 primary subscriber
B. three Cisco Unified Communications Manager servers:
• 1 publisher and TFTP server combined
• 1 primary subscriber
• 1 backup subscriber
C. four Cisco Unified Communications Manager servers:
• 1 publisher
• 1 TFTP server
• 1 primary subscriber
• 1 backup subscriber
D. five Cisco Unified Communications Manager servers
• 1 publisher
• 1 TFTP server
• 1 primary subscriber
• 2 backup subscribers
Answer: A

For Me its Option ( B ) :

Cisco recomended if u have more than 1000 phone than not put call processing load and Back up load on Publisher ( So 1 publisher with tftp )

1 subscriber primary make call processing of upto 7500 phone
1 subcriber backup if primary goes down come in action.

These are minimal redundancy solution and cost effective i think…

Thanks Anonymous

Q 106 – D is the correct Answer. The Auto Registration Script gets down loaded from the BAT Admin and then uploaded to the CRS server

Type A SIP phone w/ no dial rules requires Dial softkey or # key – answer = C+E

Q 53 – B is the correct answer, safari is correct about the no more then 1000 phones registered to the publisher

Somebody has the PDF from p4s, the link is down…

http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_27069660.html

option of this question are

A. +492288224002
B. +00492288224002
C. +0049I288224002
D. +49128822400
E. +49122288224002
F. +491288224002

some are says B is correct

Some says F is correct

Can some one put correct answer with verified explanation and Justification

sorry i forget to put question

http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_27069660.html
option of this question are

What is the output displayed on the IP phone when the incoming calling number is
00492288224002?

A. +492288224002
B. +00492288224002
C. +0049I288224002
D. +49128822400
E. +49122288224002
F. +491288224002
some are says B is correct
Some says F is correct
Can some one put correct answer with verified explanation and Justification

Question: 21
Which three of these are characteristics of a Cisco Unified Communications Manager
software-based Media Termination Point? (Choose three.)
A. The codec and packetization of both call legs must be identical.
B. This Media Termination Point type does not require any DSP resources on the Cisco router.
C. DSP resources are required.
D. It can provide G.711 mu-law to G.711 a-law conversion and vice versa.
E. It can provide packetization conversion for a given codec (for example, if one call leg is
using a 20-ms sample size, but the other call leg is using a 30-ms sample size).
F. This Media Termination Point type is typically used for RSVP agent configurations or Cisco
Unified Border Element media flow-through configurations.
Answer: B, E, F or Bdf

My answer is A,B,F

Justification : Re-Packetization of a Stream

An MTP can be used to transcode G.711 a-law audio packets to G.711 mu-law packets and vice versa, or it can be used to bridge two connections that utilize different packetization periods (different sample sizes). Note that re-packetization requires DSP resources in a Cisco IOS MTP.

Refrence :http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/media.html#wp1046314

But in ref guide for cipt1 642-447 for v 8 says

Software MTP canot be used for any transcoding other than G.711 mu-law to G.711 a-law

They are typically used for Rsvp agent configurations and provide supplementry service with H.323 v 1 .

Confusing : whose explanation is correct ……..????????????????

Somebody has valid dumps?

Hey all

I passed this exam last week Xinis dump is the most effecient dump to get a passing grade till now 🙂

How much this dumps is valid

how many new and modified question u faced

Plz let me know em also going to take this exam in 2 0r 3 days

@Tyler
in the Xenix dump are the Questions and aswer correct? or only are the Questions correct?

@ Tyler

How many marks u got, How many question in your exam are out of dumps or modified…

QUESTION 93
Which IP phone hardening technique will prevent call signaling and media stream tampering?
A. disable GARP
B. disable PC. to-voice VLAN access
C. use MIC firmware images signed by Cisco
D. store IP phone configuration files on the SFTP server
E. deploy authentication and encryption between IP phones and Cisco Unified Communications Manager
Answer: E or B

Q 93, I would say E

Q 21 – B,D,E

Your are rite , I have confirmed these questions with same answers …

What abt my Question which i put on 17 Aug

sorry i forget to put question
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_27069660.html
option of this question are
What is the output displayed on the IP phone when the incoming calling number is
00492288224002?
A. +492288224002
B. +00492288224002
C. +0049I288224002
D. +49128822400
E. +49122288224002
F. +491288224002
some are says B is correct
Some says F is correct
Can some one put correct answer with verified explanation and Justification

Please first click on link for senerio.

safari, I would say the answer is F because 00 indicates and international number which means we strip the 00, add a + then use the International CSS to match the Trans_CLID pattern which strips the +4922 and prefixes +4912 making the result +491288224002

Q96
In a Cisco Unified Communication Manager 8.0 cluster, how is database replication accomplished to run-time data?

The Xenrix said “Replicaiton is a mesh from sub to sub and sub to pub.

I think the answer is is a hybrid using both hierarchical and mesh process.

Any idea?

Forgot to say..based on SRND 8x

QUESTION 96
What is the minimum number of partitions that must be defined given the dial-plan rules listed below?
– All employees can call local and service numbers.
– Managers can call long-distance and international numbers.
– Executives can call all PSTN numbers including premium numbers.
– Only administrative assistants can call executives.
– Incoming calls can only be routed to phones, not to trunks.
A. 3
B. 4
C. 5
D. 6
E. 7
F. 8

The dump says C. I would say B 4 because the non-executive phones could be in the null/none partition while the route patterns point to the trunk all have different partitions already so a “phones partition” is not needed; there for only 4 partitions are needed.

As for the first question 96 posed by anonymous – should be Mesh for run time information

Selsius,

So you said that the answer in Xenix is right?

Good evening all,

I found a lot of difference between the Xenix dump (provided in voicetut) and the dump, provided http://certcollection.org/forum/topic/122881-cipt-1/.

Any idea which is more efficient?

@anonymous, Yes Xenix is correct for that one

Thanks selsius,

And one more question concerning Q99

Where is toll-fraud prevention configured in CUCM?
A. Partitions and CSS
B. Enterprise Parameters
C. Route Patterns
D. Service Parameters

On first look it’s seems that all are right? Xenix said D? What’s your opinion and why?

@safari,

About question:

another confusing question

If no SIP dial rules are configured on an IP phone, at what point in the collection of digits does a Type A SIP
phone send digits to the Cisco Unified Communications Manager? (Choose two.)
A. when the interdigit timer expires
B. when the collected digits match a SIP dial rule
C. when the user presses the Dial softkey
D. as each digit is collected (it is sent for analysis)
E. when the user presses the # key

Answers are C and E. Please refer to Cisco Press Implementing Cisco Unified Comm. Manager Part 1:

Type A SIP Phones: No Dial Rules
Type A phones without SIP dial rules (default) do not deliver a dial tone to the calling party
when the calling party goes off-hook with the handset, speakerphone, or headset. All digits
are sent after the user completes dialing and clicks the Dial softkey. This function is similar
to the Send button used on cellular phones.
Figure 11-8 illustrates a user making a call to extension 1000. The user has to dial 1000
followed by clicking the Dial softkey or the # key. The phone then sends a SIP INVITE
message to CUCM for digit analysis.

@SimplyCasual
I think D is the right answer. The reason I say this is that I was thumbing from the CIPT course material from Cisco and in it, it says how setting service parameters to effect how the forwarding and conferencing of off-net calls can be used to prevent toll-fraud. So based on that I would agree with D.

Thanks. I think in this way too.

QUESTION 196
Within MLA, what is the relationship between applications, privileges, and roles?
A. Privileges and application resources are applied to roles.
B. Users are assigned privileges, which are associated with groups, which, in turn, contain roles and
applications.
C. Application resources are assigned to roles, and privileges are assigned to applications.
D. Applications are associated with groups. Roles and privileges are assigned to applications.

When I look at this, all 4 answers seem wrong. Xenix says “C”. Any thoughts?

QUESTION 117
Which three characteristics are used to determine which devices go into a device pool? (Choose three.)
A. Device type
B. Class of service
C. Geographic proximity
D. Extension mobility CSS
E. User hold MOH source
F. auto-registration CSS

Xenix says CEF. I say CF and that this one only has 2 right answers.

Hi,

And what about these two questions:

http://imageshack.us/photo/my-images/845/63193877.png/
http://imageshack.us/photo/my-images/716/76267929.png/

Are they correct and can you explain them to me?

selsius,

Check this out:

Each role refers to exactly one application, and each application has one or more
resources. Access privileges are configured per application resource in the role
configuration. Roles are assigned to user groups.

SimpCas,

The first one, since the type is defined as unknown we apply the settings for unknown which in this case means we do no digit manipulation. The answer is correct.
For the second one Software MTPs, although not transcoders can translate from G711a to G711u as well as being a resource to capture and repacketize two RTP streams transmitting at different rates. The Answer is correct

Thanks! Sorry for the SPAM, but what about this:

http://imageshack.us/photo/my-images/695/unledswk.png/

Why the answer’s B? In HW_CFB_1 you have opportunity for 2conf?

SC, CM will load balance between similar Media Resources in the same MRG

Damn it, you’re right! 🙂 Did u passed the exam?

@ Selsius

Oh Great i also so confused about SC question ……But now i got its Logic Thanks Selsius

Long ago i had asked this question …….. Can anyone have explanation ????

1. Which two services need to be activated on a Cisco Unified Communications Manager for it to
function as the publisher in a cluster? (Choose two.)
A. CiscoDirSync
B. Cisco Extension Mobility
C. Cisco Messaging Interface
D. Cisco TFTP
E. Cisco CallManager
F. Cisco DHCP Monitor Service

Answer is D & F
But my answer is A & E
Justification:
Examples of UFFs or subcriber ( Or Phone services ) services include the following:
■ Call Forward All (CFA)
■ Message Waiting Indication (MWI)
■ Privacy, Enable/Disable
■ Do Not Disturb, Enable/Disable (DND)
■ Extension Mobility Login (EM)
■ Hunt Group Login Status
■ Monitor (future use)
■ Device Mobility
■ CTI CAPF Status (Computer Telephony Integration, Certificate Authority Proxy
Function)
The services listed in Table 1-1 rely on the availability of the publisher server regardless of
the version of CUCM used.

Table 1-1 Publisher Server Required Services
Component Function
CCMAdmin Provisions everything
CCMUser Provisions user settings
BAT Provisions everything initiated by the Bulk Administration tool
TAPS Provisions everything initiated by the Tool for Auto-Registered Phone SupportCUCM
AXL Provisions everything initiated by the AVVID XML Layer service
AXIS-SOAP Enables and disables services through SOAP
CCM Inserts phones (auto-registration only)
LDAP Sync Updates end-user information ( you nedd dirsync serverice enabled)
License Audit Updates license tab
Plz comments it confusing

My Second big Question Which is I think wrong in Xenix in one place

Question 2 :

Refer to the following Exhibits.

http://imageshack.us/photo/my-images/805/qi1.jpg/

Assume that route pattern for international calls is assigned to the PSTN_Pt partition. Alter applying the CSSs
shown in the exhibit to a phone and placing a call to 00014087071222, which of the following statements is
true?
A. The call will be blocked because the line and device CSS will be combined and partitions in the device CSS
will take precedence
B. The call will be permitted because the line and device CSS will be combined and partitions in the line CSS
will take precedence
C. The call will be blocked because any translation pattern that is blocked will take precedence
D. Only calls from the primary line will be permitted Secondary lines will be blocked.

Xenix Says B:

I think in this particular scenerio answer is A

Because PHONE DEVICE CSS listed first

I will also quote refrence here from please read the last wording of this example ….

http://imageshack.us/photo/my-images/545/unlednte.jpg/

What do u think guys????

About Q2…the line CSS always take priority over the device CSS.

About Q1..in Xenix the answers are A and F?

I’m scheduled to take the exam on Friday

Q2 – The answer is B. When a CSS is on both the Line and the Device they get combined into 1 big CSS with the partitions of the Line CSS first followed by the Device CSS partitions.

Q1 – This question I can only assume was not captured properly by the person who made the Dump. The question makes no sense. THere is no logical answer for it.

Good luck, mate. Please update us with your score.

All,

I’ve another question. Can you confirm that this is the right answer? I cannot test this in my lab CUCM.

http://imageshack.us/photo/my-images/535/unledgi.png/

Hi All,

Can anyone please clarify on below question.

If an IP phone loses connectivity with its primary server and registers with its backup server, what happens when connectivity to the primary server is restored?
A. The IP phone will try to re-establish a connection to the primary server every 90 seconds.
B. The IP phone will continuously try to re-establish a connection with the primary server; if
successful, the IP phone will re-register with the primary.
C. Once the IP phone registers with the backup server, the administrator will need to reset the IP
phone for it to re-register with the primary server.
D. Once connectivity is re-established with the primary server, the IP phone will wait until there
have been three successful TCPkeepalive exchanges before it will re-register with the primary
server.
As per AT Answer: B
I think it should be D.

any comments from your side?

One more to clarify

QUESTION NO: 25

Refer to the exhibit. When Phone B places Phone A on hold which audio stream will be heard and from which server will it be delivered? (Choose two.)

Phone B :- User hold Audio 4
1. Priority MOH server B

Phone A :- User hold Audio 1
1. Priority MOH Server A

MOH Server B – Audio 1, Audio 2, Audio 3, Audio 4
MOH Server A – Audio 1, Audio 2, Audio 3

A. Audio 1
B. Server A
C. Server B
D. Audio 4
E. Audio 2
F. Audio 3
G. No audio will be delivered

As per AT and Xenix Answer is: B,G.

I feel it should be C, D

Please comment.

@Raven, For the question about the phone restoring connectivity to the primary server, I first thought the answer was D too. Then I read the question more carefully. The phone will wait for 3 Call Manager keepalives before re-registering with the primary server. The answer D says 3 TCP keepalives. Because of that it should be B.

@Raven – I just tried the MOH one in my lab here at my home. B,G is correct. The MRGL on phone A determines which server it should go to. The Audio Source on B determines which audio stream should be heard. And as I just tested, if the requested audio source does not exist on the specified server, nothing is heard, no audio is delivered.

QUESTION 120
ABC Company has four locations total, all connected via an IP WAN. Each location has a Cisco Unified
Communications Manager cluster, and the headquarters cluster is connected to all other clusters via
intercluster trunks. ABC wants to allow up to eight simultaneous calls across the IP WAN. To call between
sites, an employee dials a 2-digit access code plus a 4-digit extension. If more than eight calls are made, or if
the IP WAN is not available, calls should be sent to ABC Company’s preferred long-distance carrier. If the trunk
to the long-distance carrier is not available, the local provider should be used. The long-distance carrier
requires 10 digits to complete a long- distance call. The local provider requires 11 digits to complete a longdistance
call but only 7 digits for local calls.
What is the minimum number of route lists required at the headquarters location?
A. 1
B. 2
C. 3
D. 4
E. 5

The dump answer is A. I say C. My reason is that although the basic function of call routing could be performed with one route list, to do correct digit manipulation of the called number when going out to the PSTN would require 3 Route Lists. Each remote site would need it’s own digit manipulation rules.
THoughts?

@SC Yes, the CM will give an error if you try to make a 4th CM a member of a CM Group

More thoughts on question 120. I suppose if global transformation patterns were configured and transformation CSSs were applied on the Gateways then it could be done with 1 Route List.

QUESTION 16
Which two Cisco Unified Communications Manager BAT features provide a robust alternative to DRF? (Choose
two.)
A. performs bulk transactions for Cisco Unified Communications Manager
B. exports data (phones, users, gateways, and so on); exported files can be modified and re- imported
C. supports globalization
D. has an import and export function that can be used to move data records from one Cisco Unified
Communications Manager cluster to another
E. exports data (phones, users, gateways, and so on); exported files can be modified and re- imported in the
Active Directory database
Correct Answer: AD

I would say AB. Anyone know where the AD answer comes from?

@ Selsius
Good Luck on your exam. I have just started studying the CIPT-1 exam. Any particular suggestion you may have to BEST prepare for this exam. How long did you prepare for before deciding to register.

By the way, thanks a lot for helping every one out. Specially with the high ambiguity in the Questions and Answers, Collaboration is a MUST!

Any suggestion will be greatly appreciated.

Good Luck selsius

If possible Plz note new questions 🙂

Best wishes for you …..And i know you will definitely pass this exam with good points..

Q120

IT SHOULD BE= A ……..BECAUSE IN SINGLE ROUTE LIST YOU MENTION THE PATH TO ROUTE PATTREN e.g ipwan , PSTN, ANY OTHER SOURCE BUT YOU MENTION ALL THESE IN SINGLE ROUTE LIST …….. fIRST IT CHECK ROUTE LIST THEN FIND TO MATCH ROUTE PATTREN IN DEVICES PRESENT IN ROUTE GROUP.

IF THIS QUESTION IS ABT ROUTE GROUP THEN THE ANSWER IS C……

@Selsius
Thanks buddy for your reply 🙂 and wish you all the best for your exam.

I am testing in 1 Hour.
I used Xenix.
I took the CIPT1v6 a couple of years ago and recognized a lot of Qs from v6 exam on Xenix.
Hope my experience with v6 Exam and Xenix will get me through.
I will let you guys know.

QUESTION 96
What is the minimum number of partitions that must be defined given the dial-plan rules listed below?
– All employees can call local and service numbers.
– Managers can call long-distance and international numbers.
– Executives can call all PSTN numbers including premium numbers.
– Only administrative assistants can call executives.
– Incoming calls can only be routed to phones, not to trunks.
A. 3
B. 4
C. 5
D. 6
E. 7
F. 8
The dump says C. I would say B 4 because the non-executive phones could be in the null/none partition while the route patterns point to the trunk all have different partitions already so a “phones partition” is not needed; there for only 4 partitions are needed.
Any Thoughts?

Passed Today. 860
I had previously passed the CIPT1v6 so I had some experience, but I did use Xenix.
A couple of questions were not in the Xenix, but I did find them in the comments from other guys/gals in the post.

thx for you info Crybaby

CryBaby,

Where we can see these questions?

All,

And what about that:

Which two services need to be activated on a Cisco Unified Communications Manager for it to
function as the publisher in a cluster? (Choose two.)
A. CiscoDirSync
B. Cisco Extension Mobility
C. Cisco Messaging Interface
D. Cisco TFTP
E. Cisco CallManager
F. Cisco DHCP Monitor Service

Xenix said A and F, somebody says D and F?

Hello All,

Which two services need to be activated on a Cisco Unified Communications Manager for it to
function as the publisher in a cluster? (Choose two.)
A. CiscoDirSync
B. Cisco Extension Mobility
C. Cisco Messaging Interface
D. Cisco TFTP
E. Cisco CallManager
F. Cisco DHCP Monitor Service

I think answer to this should be A & E, here is the explanation.

From CUCM 8 SRND

Cisco CallManager :- Unified CM clusters only: If you deactivate the Cisco CallManager or CTIManager services in Service Activation, the Cisco Unified Communications Manager server where you deactivated the service no longer exists in the database, which means that you cannot choose that Cisco Unified Communications Manager server for configuration operations in Cisco Unified Communications Manager Administration because it does not display in the graphical user interface (GUI).

Cisco DirSync – Activate only on the first node.

Cisco DHCP Monitor Service – Activate this service on the node that has DHCP enabled. (Since we can use dedicated DHCP server no need to activate this service on Publisher)

Cisco Messaging Interface – Activate on only one node in the cluster. Do not activate this service if you plan to use Cisco Unity voice-messaging system. (It should be activated on any one node no need to activate it on PUB)

Cisco Extension Mobility – Activate on all nodes in the cluster.

Cisco TFTP – If you have more than one node in the cluster, activate this service on one node that is dedicated specifically for the Cisco TFTP service. Configure Option 150 if you activate this service on more than one node in the cluster.
(We can have TFTP enabled only on SUB as well so I dont think this could be the answer).

What you say?

@ Revan

I also confident on A & E ………