Pass Your Cisco 300-075 Exam Easy!

The premium dump 161q is still valid. Passed today.

QUESTION NO: 26
Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What
value should be entered into the gatekeeper to support this bandwidth?
Cisco 300-075 Exam
"Pass Any Exam. Any Time." - www.actualtests.com 20
A. 768 kbps
B. 384 kbps
C. 512 kbps
D. 192 kbps
Answer: B
Explanation:
Incorrect answer: A, C, D
A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This
sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video
rate increases to maintain a total bandwidth of 384 kb/s.
Link:
http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.ht
ml#wp1059726
===I think the corect is Answer A
http://flylib.com/books/en/2.110.1.217/1/

Hi Kazee, look top of this page: 300-075 Premium VCE File...and Download

Hi Friends;

Have you any update for dump. Who entered the exam in today?

Many Thanks

Planning to give CIPv2 exam, can anyone let me know where can I get the 161q dumps?

Passed yesterday with a score 897. All questions from 161q, refer to the comments in this forum.
thanks for everyone and good luck!

Passed yesterday with a score 9XX. All questions from 161q and 2 questions from out of 161q, Which is already discussed everyone in this forum.

What are the tasks required to route calls between H323 and SIP ENDPOINTS and vice versa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
ANSWER: C and E. EXPLANATION: You cannot have A and C on at the same time, and E has to be on for H323 to work at all.

By default, the VCS acts as a SIP-H.323 and H.323-SIP gateway but only if at least one of the endpoints that are involved in the call is locally registered. You can change this setting so that the VCS will act as a SIP-H.323 gateway regardless of whether the endpoints involved are locally registered. You also have the option to disable interworking completely.

The options for the H.323 <-> SIP interworking mode are:

Off: the VCS will not act as a SIP-H.323 gateway.
Registered only: the VCS will act as a SIP-H.323 gateway but only if at least one of the endpoints is locally registered.
On: the VCS will act as a SIP-H.323 gateway regardless of whether the endpoints are locally registered.

You are recommended to leave this setting as Registered only (where calls are interworked only if at least one of the endpoints is locally registered). Unless your network is correctly configured, setting it to On (where all calls can be interworked) may result in unnecessary interworking, for example where a call between two H.323 endpoints is made over SIP, or vice versa.

@ Sham1629

Congratulations. If you remember any questions, Please share the questions.

I passed my 300-075,Scored 869 Border and lucky though,161Q are valid but do check questions and answers.
Better do your own research and more on VCS.

@ Joseph

It is about 161q dump

Please can you share with us the last questions ? Which VCE file are yoy using to check the question number and answers ?

Thanks

@Hari.

Sorry. Yes I got this question in exam and my answer was A & D.

@Hari

No.

Hello,

Today I passed with 906 marks in second attempt.

Really really appreciate this community!
And I really recommend to read through all comment here before exam.

No new questions in my exam.
but I got all questions which corinth mentioned.
I am not sure correct answer, but you may want to reviews answers well.

good luck and many thanks!

Join our COLLABORATION

Passed the exam today. 161q and feedback on this site are valid, no new questions. However, some folks here are posting incorrect answers...verify all of your answers!

Passed exam score 897. Dumps are still valid and everything from 161Q. No new questions. Only thing is that we need to refer for correct answers. Blogs update on this portal was very helpful. Without this i couldn't make it. Again thank you for all.

Guys I am really sorry but unfortunately I dont remember those new questions...

@Shem,

Planned to take exam tomorrow morning, please share the new question.

@sham1629

If you remember questions please share, planned to take exam tomorrow

@ sham1629 @ saintcisco

Now it have 8-10 new questions?..Till 24th July everyone was speaking about 2 new questions.

If you can recollect, Please share what are the additional questions other than 161q + 2 discussed new questions.

@sham1629

What kind of new questions have you seen?
Could you describe them ?

@ sherazkk

You got Q147 in your exam?

@Hari:

According to me the answer is
A&D

Attempted exam but failed yesterday...
Almost Question are from 161Q but 8-10 new questions though.

I passed today with 869 i use 161Q and @Ramesh comment. 161Q is valid but answers not valid. Be careful. Thanks again @Ramesh

@ Hi-Octane
@ Corinths@ Anco
@ Canada
@ Pass
@ sherazkk
Any of you got below question in your exam?..If Yes, What is the correct answer

Which two steps must you take when implementing TEHO in your environment? (Choose two.)

A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.

@steve:
These were the CAC questions I can remember, its possible there has been more of them in the test. Honestly, i have had just 57% score report on CAC, so some of the answers I have provided must have been wrong..


An engineer must resolve a [[VIDEO]] call failure issue. When using RTMT, the engineer notices that the Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option is the cause of the call failure?
B.lack of video bandwidth

In Cisco Unified Communications Manager, where do you configure the default bit rate for audio and video devices?
C. Cisco CallManager service under Service Parameter Configuration

Which statement is true when considering a Cisco VoIP environment for regional configuration?
B. G.729 requires 24K of bandwidth per call.

When you use the Query wizard to configure the trace and log central feature to collect install logs, if you have servers in a cluster in a different time zone, which time is used?
A. TLC adjusts the time change appropriately.

Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
D. a hardware transcoder -LTI Transcoder Resource = Local Transcoding Interface
E. DSP resources -DSP PVDM Resource

Which function can be implemented without MTP resources?
e) Multicast MoH
(sorry, cannot remember all the answers, just the one i have selected)

Hello Guys

Today I passed the exam with 916
I hope it's helpful for you
thanks everybody in this forum for sharing the questions and answers

Friends,
on which file or document should I start as the baseline? Is this 102q vce file still valid? I understand that many thanksful people posted right answers and explanation on this thread, but really need to know which dump or vce should I start.

Thanks!

@Juan Yes the answer is C. There is a "sip identity header" which carries the information. It is the more precise answer rather than the vague 'header request' option. I will answer C in exam.

Hello i will take the exam next week, any hints which file can I study? All the best. Thank you.

Passed last friday, questions valid, but not all the answers!
You should really make a study of the course materials and still then it's hard to find the right answers.

What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

Could it be C and D? I read through several Cisco docs (I didn't have the actual servers to log into)...it looks like yes "registered only" is better than "on". Definitely C is the answer, but as per the cisco guide, the sip mode is on by default and stays on (we don't turn if off on any step).

Thanks Everyone and examcollection. I passed my exam Today with 896.

How many questions are in the dump file Cisco.ActualTests.300-075.v2015-10-28.by.Ace.102q.vce. I've 102 questions in total. Can someone help me.

For q133:
____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D


I think the answer is C, not D. Referring to the Cisco doc:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html

What do you think?

Hello, passed 906, second attempt!

63%
75%
100%
90%
100%
100%
100%
43% this one always poor, may be we need also correct some answers from DUMP or here in disscution


All Questions are from dump and Comments, please read all comment here, I think it is enought to pass!

Some questions are not 100 known, please read for them in docs

for exmaple

What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

I thought to ansewr AC

But after reading what @JB said anwered CE
It stays that you have to use two ansewrs

but REGISTERED ONLY or ON cant be set at same time, read docs

Also one more question about

Mobility and Presence when WAN goes down
now there are new option BE6000 in fallback
Of cource BE6000 will provide Mobility Presence
I chose CME in SRST in voice gateway
because it asked in BRANCH,
@Faquejai on page 2 was writing about it

Please read all commnts here, and read docs for some questions which are not having exact ansewrs

thank to all who shared exam, and goodluck to all who will take exam

I passed the exam in first attempt with 906 marks.
Special thanks to @anon12345, @corinth, @Hi-Octane and @suman for their guidance.

@Dave/Ron

"
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

why the answer should be AE?

I suspect it to be A and C"

What you're looking at in the answers is the pathway you would take in the VCS-E menu to make configuration changes.

The interworking config is, by default, set to "Registered Only." One of the earlier posters stated they chose A & E, but based on the question breakdown given at the end of the exam and the percentage they received, they believe the A&E answer is wrong.

Given that the question asks what is REQUIRED configuration, not optimal configuration (which IMO would be 'On', I believe one of the answers to be C.

Both of you said A & C, but given how configuration is handled in VCS-E, it has to be one or the other, Configuration cannot be set to both On and Registered Only.

I believe the second answer to this question is E, H323. When you hover over the information Icon under H323 config in VCS-E, it reads "Determines whether or not the VCS will provide H.323 gatekeeper functionality."

When you do the same for SIP, it reads "Determines whether or not the VCS will provide SIP registrar and SIP proxy functionality.
This mode must be enabled in order to use either the Presence Server or the Presence User Agent."

Therefore, I believe the correct answer to be C and E.

corinth,
What was you CAC questions if u remeber?? everyone i see get low marks on CAC

Just passed the exam with 910 and got lucky 4th time.

Thanks all for your support.
Much appreciated

I have the same doubt of @Dave.

For q133:
____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D


I think the answer is C, not D. Referring to the Cisco doc:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html

What do you think?

Hello Friends,

Could you please explain question

What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

why the answer should be AE?

I suspect it to be A and C

Hello Friends,

Could someone explain why this has to go with Option A

QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco UNITED CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.

Guys,
I just passed with score of 925. Here are the questions I have reviewed myself and are different from what has already been published in this forum.


Which two commands verify Cisco IP Phone registration? (Choose two.)
Changed question:
What are the two commands, one of which can be used to verify SIP phone registration and one can be used to verify SCCP phone registration in CME?
A. show telephony-service ephone-dn
B. show voice register session-server
C. Show ephone registered
D. show ccm-manager hosts
E. show sip-ua status registrar
Answers remains the same: C, E


Refer to the exhibit (screenshot of Transform where some field are popluted).
Pattern type : Regex
Pattern string : ccnpcolab
Pattern Behaviour : Add Suffix
Replace string : @cisco.com
State : Disabled
A call is initiated from endpoint to address "ccnpcolab" from a VCS where the above configuration is applied. Which of the below statements is true?
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com
Answer: C (note the status of transform is disabled, which means the dialed URI is NOT BEING transformed and is sent in its original form)


Which two statements about Cisco Unified Communications Manager Extension Mobility are true? (Choose two.)
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. When Logged off the device can be set to use an autogenerated profile or a user defined profile
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time.
Answer: C, D (100% score report on this topic)


QUESTION NO: 118 This is no more valid question, it now says:
How many nodes can a phone establish a SCCP connection to at the same time?
C. 1


Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.
Answer I selected: D
-but due to my 75% score report on VCS I would say that better is to go with option A


Which option describes the effect of this configuration? (two identical configurations of CME router)
A. It implements Cisco United CME redundancy. (the word UNITED is there by purpose, meaning this is false answer)
B. It configures a standby Cisco Unified E.
C. It configures failover.
D. It implements Cisco IOS redundancy.
E. It creates dial peers.
F. It implements HSRP.
Answer: C (100% score report on this topic)


You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phones located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two.)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway.
C. Phones at the remote site are assigned to the incorrect device pool.
D. The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
Answer: A,C


Which function can be implemented without MTP resources?
-regarding this question - it comes in multiple combinations, but keep in mind
-MTP required:
DTMF relay conversion
DTMF inband RTP-NTE (rfc2833)
terminating a media stream that uses the same codec
H.323 Outbound Fast Start
SIP early offer
IPv4 to IPv6 conversion
-MTP NOT required:
multicast music on hold
delayed offer h.323
SIP delay offer


Which statement about when user A calls user С using SIP are true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
Answer: B !!!!
-note: i have selected A in the exam, aparently i was wrong due to lower score report on this topic


Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit numbers. Which option should be configured if you want outgoing calls from 4-digit internal directory number to be presented as a 10-digit number?
A.calling party transformation pattern filed in Route Pattern
B.AAR group
C.translation pattern
D.route pattern
Answer: A (has changed and makes more sense now)


What are the tasks required to route calls between H323 and SIP ENDPOINTS and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
My answer AE, but apparently this combination is wrong (due to lower VCS score report)


Hope this helps :-)

Hello!

Can you please explain question

What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

why answer AE?

Sorry guys forgot to mention the corrections i made were from the exam collections premium 161 dumps where all questions are valid.

Good luck.

@Juan Below are my answers for 4 questions you have asked. But none of the below were in my exam.

QUESTION NO: 133 (did not get in the exam)
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which
result is achieved by enabling this option?
A.Resolve FQDN using DNS type SRV record.
B.Resolve FQDN using DNS type A record.
C.Ensure FQDN is used in SIP Identity header.
D.Ensure FQDN is used in SIP Request header.
Answer: D

QUESTION NO: 142
A presales engineer is working on a quote for a major customer and must evaluate how many Cisco VCS Expressway traversal call licenses for which to plan. Calls to and from which three
routes must the engineer include in the tally? (Choose three.)
A.gateway
B.Cisco 9971 Endpoint
C.border controllers
D.gatekeeper
E.SIP trunk
F.VCS
Answer: C,D,F

QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.
Answer: D,E

QUESTION NO: 156
An engineer has configured a Cisco EX60 to register with a Cisco VCS-C, but the device is not showing up as registered. During troubleshooting, which component will the engineer likely find
missing in the configuration?
A.gatekeeper
B.MCU
C.default gateway
D.TMS
E.DNS
Answer: A

Hello All

PASSED: 910

I PASSED TODAY IN UK (18/7/16) WITH A HEALTHY 910, yes scary huh. Man what a ride this beast has been. My 2nd attempt.

First i would like to thank all the guys who took time out to help us specifically Guillaume and Anon12345 who laid the platform and others like Adam, Larry, Steve, Calvin and whoever i missed. This forum has been very helpful.

OK i have spent a bit of time going through every correction for you and the below should be enough for all of you to pass. I have only made corrections to the answers i felt that are wrong based on other comments. the other answer feel correct to me so i have left those. Please analyze your own answers but below are my Personal corrections.

Below are my corrections which should get you guys through this HORRIBLE exam.


57: E
-> The DX650's MAC address is incorrect in the Cisco UCM.

68: BE
-> Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
-> Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.

80: B
-> 34(100010)

83: c
-> 2 servers

109: B
-> G.729 requires 24K of bandwidth per call.

115: D
-> Transform

116: CD
-> device can adopt a user profile even when no user is logged in.
-> device profile has most of the same attributes as a physical device.

118: D
-> 2 nodes

120: A
-> The traversal zone on the VCS Control does not have a search rule configured.

121: Qestion has been changed to below
What are requirements for hardware MTP on Cisco IOS routers?
a. PVDM or DSP resource
b. LTI local transcode resource
c. ref2833
d. one audio codec
e. T1 PRI card
Answer AB

122: AB
-> DX-650
-> Cisco Jabber Desktop

123: BE
Answers changed. Finally CISCO released their mistake.
-> Answer B now reads The NAT device is allowing only RTP/RTCP ports from DMZ to internal network. so select this.
-> The router does not have a route back from the DMZ to the internal network.

124: F
-> registration. Question only asks for one answer

125: ACE
-> Configure voice register pool.
-> Configure an SRST reference.
-> Configure the SIP registrar.

127: B
-> Transform

128: B
-> SCCP Gateway

129: E
-> BFCP

130: Careful question has been changed to Engineer must resolve a "VIDEO" call failure issue.
-> I selected B: Lack of video BW

131: Answer is B (BE CAREFUL AS THEY MADE MORE ANSWERS SIMILAR TO THIS like Client zone and server zone, DONT SELECT THIS)

132: C
-> LRG

133: D
-> Ensure FQDN is used in SIP Request header.

134: DE
-> Apply registration, authentication, and media encryption policies.
-> Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of BW

135: D
-> when you require administrative access to the Cisco Expressway Edge from the Internet

136: A
-> It implements Cisco Unified CME redundancy.

138: ABE
-> Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
-> Verify that all phones are registered to a second subscriber server.
-> Verify that media resources fail over to a secondary subscriber server when the publisher fails.

139: Should be A based on the question "best QOS" but it think CISCO got this wrong
-> I selected C: 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning

140: BEF
-> hosted DN patterns
-> the SIP or H.323 trunk
-> hosted DN groups

142: CDF
-> border controllers
-> gatekeeper
-> VCS

144: B
-> Create a block learned pattern.

145: B
-> AAR is routing some of the calls.

146: FG (This question only asks for two answers)
-> Assign directory URIs to users.
-> Configure the SIP profile.

147: I wasn't asked this one but i think CISCO got it wrong but if i was asked i would select AF (your risk)
-> Implement local failover
-> Implement centralized failover

148: AC
Answer has changed
-> The site has exceeded the number of SRST endpoints supported by the voice gateway.
-> Phones at the remote site are assigned to the incorrect device pool.
C now reads "Some Phones at the remote site are assigned a device pool without SRST reference"
so select this one. CISCO finally realized their mistake.

149: DE
Answer have been changed to the below.
A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833)
D. Sip delay offer
E. Multicast MOH
-> I selected DE

151: BCE
-> TEHO
-> CER
-> AAR

152: CE
-> show ephone registered
-> show sip-ua status registrar

153: F (Question only asks for one answer: DNS sever)

154: C
-> The user can log in to only one device at a time.)

155: D
-> Cisco Unified Communications Manager Express in SRST mode

158: A
-> calling party transformation pattern

160: I selected A
-> SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.

161: BDF
-> The Tomcat certificates do not match.
-> The ILS authentication password does not match.
-> One cluster is using TLS certificate, and the other is using Password.

-------------------------------------------------------
2 EXTRA questions.

An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are
three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled

A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com

Answer is C


What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On

Answer AE


Good luck guys, i really hope this post of mine should be enough for all of you to pass this crap and we no longer need to keep asking one question at at a time to get the correct answers

Thanks.

I've seen a few people state that they verified their answers on Cisco.com with different docs. Could you provide links to the docs? Thanks!

@CCNPCollab
To answer your question, the dump and the test have the same typo.
QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco UNITED CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.
Answer: A

So A is incorrect. The test writer put that mispelling there on purpose. It's not an accident.
So, go with Failover.

Hello! Please update who recently passed or failed exam in past few days
I failed exam last week only few points

Hi.

Please, could you put the questions q133, q142, q147 and q156?

Thank you very much!

Hi.
Please, could you share the following questions?

q133, q142, q147 and q156

thank you very much

passed in second try with 869. Thank u all for your inputs

QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco Unified CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.
Answer: A

Why people think the answer is A or C? any specifics

Hi Ramesh,

QUESTION NO: 118
How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2

Has the above question changed to how many SCCP phones can register at one time?

No this question did not changed and it's not clear to me. I answered "2".


QUESTION NO: 130
An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?

A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources

did this question read as "video call" is the answer still A?

--> I answered "lack of video bandwidth

QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning

Did you answer A or C?
--> C

QUESTION NO: 155
Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?
A.SRST without MGCP fallback
B.SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C.SRST with MGCP fallback
D.Cisco Unified Communications Manager Express in SRST mode

Did you answer C or D?

--> D. You can find this in the cisco documentation.

@Guillaume

Thanks indeed for the answers really appreciate. I will be attempting again hopefully next week.

Thanks!

Hi Guillaume
Congrats on passing, could you tell me what you answered for the two below please? if you got them in your exam? or your thoughts on the correct answers that "CISCO would recognize as correct"

QUESTION NO: 118
How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2

Has the above question changed to how many SCCP phones can register at one time?
Is the answer still C?

QUESTION NO: 130
An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
Cisco 300-075 Exam
"Pass Any Exam. Any Time." - www.actualtests.com 115
A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources

did this question read as "video call" is the answer still A?

QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning

Did you answer A or C?

QUESTION NO: 155
Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?
A.SRST without MGCP fallback
B.SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C.SRST with MGCP fallback
D.Cisco Unified Communications Manager Express in SRST mode

Did you answer C or D?

@Hi_Octane
Here are my answers for the questions you asked:
109: answer has changed, i selected "The site has exceeded the number of SRST endpoints supported by the voice GW" and "Some Phones at the remote site are assigned a device pool without SRST reference"
116: CD
120: I selected A (traversal zone has no search rule configured) because issue is for external calls.
123: Answer has changed so i selected "The nat device is allowing only RTP/RTCP ports from DMZ to internal network" and "The router does not have a route back from the DMZ to the internal network"
132: C (LRG) answer has changed and it's more clear now, it do not mention "call limit" anymore so AAR is not relevant anymore.
136: A (CME redudancy)
146: you have to select only two answers and it's more easy. Answers are "configure SIP profile" and "assign directory URI to users"
147: i did not hit that one but even in the dump question is not clear at all so i'm sorry but i can't help on this one.
149: answer has changed and it's more clear, it asks for 2 answers and i selected "MOH" and "sip delayed offer"
153: answers has changed, only 4 choices now (ldap server, dhcp, cucm ip, and DNS) so it's clearly DNS.
156: I selected A (gatekeeper) because gatekeeper VCS is the component in charge for registration.
160: only one answer needed and i selected "SIP TLS" because they ask for "SIP call". I added a comment during the exam because it was not clear to me. RTP/RTCP ports need to be open too if you want to get video and voice media.

@Guillaume

Congrats on passing this tough exam.

Can you please help for the following questions answers?

109,116,120,123,132,136,146,147,149,153,156,160

Although some people has given answers for above but would appreciate if you can share what you have chosen. It will be great.

Many Thanks!

Thanks

Hello Everyone,

I was able to pass this beast of an exam on my 4th attempt today, scored 888...

Here are the answers I chose for the 2 new questions. Please feel free to reach out and I will help anyone that is testing. Thanks to all of those who helped during this journey

2 new questions....



H.323-SIP internetworking mode Registered Only


Protocol>Sip>Sip on


2nd new question i selected "call is sent as ccnpcollab" because transform is in disabled state and is just ignored

Exam booked for Monday. Here goes. Will be studying all weekend.

Hi all,

Just passed this crap with 906.
All questions except the 2 already mentionned are in the 161 premium file.
Some questions were rewrited and it's more clear.
First new question
I selected :
Config-protocols-Interworking-On
Config-protocols-H323-H323Mode-On

2nd new question i selected "call is sent as ccnpcollab" because transform is in disabled state and is just ignored.

During exam do not hesitate to comment when the questions are not clear.

Feel free to ask for other questions.
Good luck !

@ MERCE

I think it should be:
Config-protocols-Interworking-On
Config-protocols-H323-H323Mode-On

Regards.

@Larry

Much appreciated

Merce

Mi respuesta de segund pregunta fue Internetworking-on y Internetworking-registered-only

pero no estoy seguro de que es la respuesta correcta

Hi Merce

http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/admin_guide/Cisco_VCS_Administrator_Guide_X7-2.pdf page 170 says:

State Indicates if the transform is enabled or not. Use this setting when making or testing
configuration changes, or to temporarily
enable or disable certain rules. Any
disabled rules still appear in the rules list
but are ignored.

Hi, I'm going to try to pass the exam tomorrow. I need help. Larry or someone, can you tell us the correct answers?
Why do you think that in the new question ( C. sent as CCNPCOLAB) is the correct?

I friend told me a other new question too:

What are the tasks required to route calls from H323 to SIP and viceversa?
Config-protocols-Interworking-On
Config-protocols-Interworking-Off
Config-protocols-Interworking-registered only
Config-protocols-Sip-Config-Mode-On
Config-protocols-H323-H323Mode-On

¿do you know what are the two answers?

thanks

Passed with 89x.
New question (mentioned earlier)
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string :
Pattern Behaviour :add suffix
Replace string : @cisco.com
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com

My answer: C (snapshot from transform with state: disabled)

Hi All,

3rd time failed with 845 Marks feeling so depressed.
Anyways can anyone please confirm for the following Questions answers?

146, 147,149,156,160, 132

no idea why still not able to achieve anything above 35% in Collaboration Edge (VCS-E) section and 40% in Implementing bandwidth Mgt and CAC on CUCM section.

@Anon12345 and all who has passed please share the right answers.

For new question i chose option
There is a new question that someone earlier had mentioned.
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com

Option A

Please help us to pass this tough exam.

Adam
I just passed the exam. I will post the questions and answers from 356 dump that I think will help you to night. I don't understand why those individuals never responded to your questions.

Hi All! I posted earlier, but it looks like my post didn't make it. So, If you see my post twice, I apologize.
I just passed, with an 880 on my exam. This forum has been so helpful, that I wanted to also add some more helpful information.

- First off, the 161 dump is valid, as far as questions. The answers need to be checked!

- For anyone just looking to find the dumps, memorize them, answers and take the test, good luck! I failed my first test, even though I had checked some of the answers on my own. After I failed the test, I went back through and tried to look up as many questions on my own to verify. I learned ALOT!

- @anon12345, @Steve and @Mohan, thank you for the wealth of information you shared!

- I recommend starting with what @anon2345 posted.

Now for some more specific information, to help those who have been debating some of the more difficult questions to verify.

Steve and Mohan both reported back with some new information. There are some test questions that have been updated. From what they have reported, it looks like maybe the original version of those questions either didn't have a right answer, or the right answer was almost as vague as the other wrong answers. Check what they say about those question. I think they will help you, but still check their answers. I had already gone through and verified ALL of the questions I remembered seeing on the test, then verifed the ones they also mentioned. Which was a difference of about 20 questions. Most of my answers matched theirs. But not all.

Question 123 according to the dumps is pretty much impossible. I want to report what I saw, because it gives you two possible right answers. It will help all those who have been debating the question since I've been studying for this exam.
*********************************
QUESTION 123
Refer to the exhibit.

An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:

•User A can hear user В and vice versa.
•User A can hear user C, however user С cannot hear user A.
•User В can heat user C, however user С cannot hear user В.

Which two properties are the most likely reasons for this issue? (Choose two.)

A.The Cisco EX60 default gateway of user С is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.
*********************************
Everyone can agree that E is correct. But none of the others match the direction of traffic, where audio is NOT moving from the internal network to the DMZ.
On the test, they changed B.
It now reads
B.The NAT device is allowing only RTP/RTCP ports from the DMZ to the internal network.

I chose B and E.

The first time I took the test, I thought I had the questions memorized. Then, after failing, taking the time to learn the questions, and also reading what Steve and Mohan added after I failed, helped me not only verify the old answers, but learn why some of the questions I thought I knew, just didn't seem to have the right answers. They modified them!
Mohan pointed out a new one, with a picture, and being asked about how the traffic will be routed.
*************************************
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled

A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com
************************************
I chosed same as Mohan. A.
Because state was disabled.

One last bit, I hope this helps save some time.
* The last 53 questions of the dump, are were I saw about 2/3 of the exam, and they are discussed at great length here. Check out what people say, but still verify!
* In looking up the questions, and the correct answers online, I found the correct answers WORD-FOR-WORD on Cisco online documentation. Most of them are in tables, or setup guides. Most of there were "Deployment Guides" or "Administration Guides."
* Someone else mentioned that Questions 57 - 68 aren't on the exam. I can comfirm that I didn't see ANY of those questions either time I took the exam.
* Any questions from the first part of the test, that don't involve VCS topics, look like they were pulled from an old CCNP CIPT2 (642-457) test. I even had to correct a couple of questions from THIS dump, based on the old dump.
* Look at the question list that people posted they saw, and add that to your list of questions to verify.

@Adam
In question 146 I would choose:
C.Associate the directory URIs to directory numbers.
F.Assign directory URIs to users.
Because it says URI calling within the "same cluster"
If someone else has a different option I would like to hear it.

failed today even though i reviewed most answer below and corrected it in my file.

I passed on my second try today! It has been a hard road.
For those who are here just to grab the dumps, memorize and pass, good luck.
I can verify that the question on the 161 dump, cover most of what is on the test.
I do want to pay it forward, though, with the help of those who have succeeded and reported, in hopes that it helps others.

First of all. @anon12345, @Steve and @Mohan. Thank you thank you thank you for your input! Bringing attention to the meat of the questions we all saw, helped me.
I got over 880 on my score!

My method for prepping:
I went through and marked all the questions that I got. I then verified the sources sited from the dumps. Or I looked it up. There were about 10 questions that were so vague, it was really hard to find a definitive answer. BUT! Many questions, I was able to find the verbage, almost word-for-word.
Anon12345 is a great place to start.

The first part of the test dump mostly pulls questions from the old CCNP Voice test, 642-457. There is an online version that confirms most of the questions from the dumps. There were a few questions that had a different answer, so I went with the 642-457 answers.

Once I had my answers pretty well confirmed, or changed from my research, I then compared what I had, to Anon12345, Steve and Mohan's answers. What they posted was pretty helpful. But as Anon12345 said, don't take their answers for 100%. Verify!

That said, here are a couple of specifics.
Mohan and Steve provided confirmation and details of new questions. This was a HUGE help!

Question 123 has been a huge pain, because only 1 of the 5 choices made sence!

Refer to the exhibit.

An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:

•User A can hear user В and vice versa.
•User A can hear user C, however user С cannot hear user A.
•User В can heat user C, however user С cannot hear user В.

Which two properties are the most likely reasons for this issue? (Choose two.)

A.The Cisco EX60 default gateway of user С is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.

Regardless of what dumps say, the fixed option.
E was always correct.
B, however now reads...
"The NAT device is allowing only RTP/RTCP ports from the DMZ to the internal network." Which follows with the fact that audio is not getting to the internal network.
My answer, is B and E from the list above.

Pay attention to the changes that Steve and Mohan have mentioned. I can confirm that all of the new changes replaced the old questions.

@D4D
__Question 148___

You have deployed a Cisco 2821 ISR to poerform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phone located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two)

A. The site has exceeded the number of SRST endpoints supported by the voice gateway.

B. The ccm-manager fallback command is configured incorrectly on the voice gateway

C. Phone at the remote site are assigned to the incorrect device pool

D. The ccm-manager fallback-mgcp cimmand is configured incorrectly on the voice gateway.

E. The site has exceeded the number of simultaneous calls allowed in SRST mode.


Note: Dump answer are B & D but other said that the answer are A & E.

----
I've deployed SRST dozens and dozens of times.
C is correct:
C. Phone at the remote site are assigned to the incorrect device pool

The device pool contains SRST reference. If phone doesn't belong to the srst device pool, that phone won't be in SRST if connection to CUCM is lost.

"A" is 2nd choice

@Mohan@Steve@D4D@Kiki

Hello Everyone,
I wanted to reach out to the users who have passed. I failed again today with a 845, and have questions about the below. There were 2 new questions that I put at the bottom of my list. I would GREATLY APPRECIATE ANY INSIGHT ON THESE QUESTIONS AS I AM GOING TO TAKE THE EXAM AGAIN NEXT WEEK....

Q.109
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A.G.711 requires 128K of bandwidth per call.
B.G.729 requires 24K of bandwidth per call.
C.The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D.To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
Answer I selected: B

QUESTION NO: 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video
endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A.The traversal zone on the VCS Control does not have a search rule configured.
B.The access control list on the VCS Control must be updated with the IP for the external users.
C.When a traversal zone is set up on VCS Control only outbound calls are possible.
D.The local zone on the VCS Control does not have a search rule configured.
Answer I selected: D
I have seen some users select A

Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio codec; E. T1 PRI
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder -LTI Transcoder Resource
E.DSP resources -DSP PVDM Resource
F.a T1 card
Answer: D,E

QUESTION NO: 123
Refer to the exhibit.
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versa.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.
Answers I selected: A,E

QUESTION NO: 132
An engineer is performing an international multisite deployment and wants to create an effective backup method to access TEHO destinations in case the call limit triggers. Which technology
provides this ability?
A.AAR
B.CFUR
C.LRG
D.SRST
Answers I selected: C.LRG
I have seen several users answer A.AAR

____QUESTION 134____
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
ANSWER: D,E

Q.135
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?
A.when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B.when you require encrypted calls to endpoints on your corporate LAN
C.when you want to enable calls to web applications by using HTTP
D.when you require administrative access to the Cisco Expressway Edge from the Internet
Answer:D
I have seen "A" as answer by many on this community

QUESTION NO: 146 in Exam options are reduced to choose 2
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish this configuration? (Choose four.)
A.Configure SIP route patterns.
B.Configure the directory URI partition and calling search space.
C.Associate the directory URIs to directory numbers.
D.Activate the URI service in Cisco Unified Serviceability.
E.Configure SIP trunk.
F.Assign directory URIs to users.
G.Configure the SIP profile.
H.Configure the URI service parameters.
Answers I chose: F,G

Should it be the SIP Trunk or SIP Profile??
URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html

I failed today for the 3rd time with a 845.
Here are my scores
VCS Control: 88%
Collab Edge: 50%
CUCM Video Service: 57%
Centralized Call Processing Redundancy: 60%
Multi-site Dial Plan: 100%
CCD/ILS: 100%
Video Mobility: 100%
Bandwidth and CAC: 29%
There is a new question that someone earlier had mentioned.
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as CCNPCOLAB@cisco.com

I selected A since the state is disabled as another user on the board had commented.
Any advise from anyone who has passed can give me to bring up these scores. I was only a question or 2 away from taming this beast.

Could someone help me to answer the below
QUESTION 123
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user В and vice versa.
User A can hear user C, however user С cannot hear user A.
User В can heat user C, however user С cannot hear user В.

Which two properties are the most likely reasons for this issue? (Choose two.)
A. The Cisco EX60 default gateway of user С is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E. The router does not have a route back from the DMZ to the internal network.

Could someone help me to understand the reason for an answer on this...
QUESTION 120

Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?

A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.

@Kiki

Thank you.

Passed today with 878.
Please Confirm Answers and cross verify with documents.




1- VCS Control: 88%
2- Collaboration Edge(VCS Expressway): 75%
3- Configure CUCM Video Service parameters: 100%
4- Describe and Implement Centralized Call Processing Redundancy: 80%
5- Describe and configure a Multi-Site Dial Plan for CUCM: 88%
6- Implement Call Control Discovery/ILS: 83%
7- Implement Video Mobility Features: 83%
8- Implement Bandwidth Management and Call Admission Control on CUCM: 29%





New Question : I dont remember Exactly question they asked in Exam

An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are

three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"

Mode : .....

Pattern type : Regex

Pattern string : @cisco.com

Pattern Behaviour :.......

Replace string : ccnpcollab

Target : .....

State : Disabled

A. Can not route call

B.Sent to Cisco.com

C. sent as CCNPCOLAB

D. Sent as CCNPCOLAB@cisco.com


I selected A since the state is disabled.
Not sure whether is right or wrong please check guides.


QUESTION NO: 1

Which parameter should be set to prevent H.323 endpoints from registering to Cisco
TelePresence Video Communication Server automatically?



A.
On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Discover to off.

B.
On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Registration to off.

C.
On the VCS, navigate to Configuration, Registration, Allow List, and set Auto Registration to

off.

D.
On the VCS, navigate to Configuration, Registration, Configuration, and set Auto Registration

to off.



Answer: A


QUESTION NO: 7

Which three statements about configuring an encrypted trunk between Cisco TelePresence Video
Communication Server and Cisco Unified Communications Manager are true? (Choose three.)



A.
The root CA of the VCS server certificate must be loaded in Cisco Unified Communications
Manager.

B.
A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP.

C.
The Cisco Unified Communications Manager trunk configuration must have the destination port

set to 5061.

D.
A SIP trunk security profile must be configured with Device Security Mode set to TLS.

E.
A SIP trunk security profile must be configured with the X.509 Subject Name from the VCS
certificate.

F.
The Cisco Unified Communications Manager zone configured in VCS must have SIP
authentication trust mode set to On.

G.
The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set

to Off.



Answer: A,C,E


QUESTION NO: 9

Which two actions ensure that the call load from Cisco TelePresence Video Communication

Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes?

(Choose two.)


A.
Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.

B.
Create a single traversal client zone in VCS with the Unified CM nodes listed as location

peer addresses.

C.
Create one neighbor zone in VCS for each Unified CM node.

D.
Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.

E.
In VCS set Unified Communications mode to Mobile and remote access and configure each
Unified CM node.



Answer: A,D



QUESTION NO: 13

If delegated credentials checking has been enabled and remote workers can register to the VCS

Expressway, which statement is true?


A.
H.323 message credential checks are delegated.

B.
SIP registration proxy mode is set to On in the VCS Expressway.

C.
A secure neighbor zone has been configured between the VCS Expressway and the VCS Control.

D.
SIP registration proxy mode is set to Off in the VCS Expressway.



Answer: D



QUESTION NO: 14

Which two options should be used to create a secure traversal zone between the Expressway-C
and Expressway-E? (Choose two.)


A.
Expressway-C and Expressway-E must trust each other's server certificate.

B.
One Cisco Unified Communications traversal zone for H.323 and SIP connections.

C.
A separate pair of traversal zones must be configured if an H.323 connection is required and
Interworking is disabled.

D.
Enable username and password authentication verification on Expressway-E.

E.
Create a set of username and password on each of the Expressway-C and Expressway-E to
authenticate the neighboring peer.


Answer: A,C
Explanation:



QUESTION NO: 15

Which two statements regarding you configuring a traversal server and traversal client

relationship are true? (Choose two.)


A.
VCS supports only the H.460.18/19 protocol for H.323 traversal calls.

B.
VCS supports either the Assent or the H.460.18/19 protocol for H.323 traversal calls.

C.
VCS supports either the Assent or the H.460.18/19 protocol for SIP traversal calls.

D. If the Assent protocol is configured, a TCP/TLS connection is established from the

traversal client to the traversal server for SIP signaling.

E.
A VCS Expressway located in the public network or DMZ acts as the firewall traversal client.



Answer: B,D




QUESTION NO: 16

What is the standard Layer 3 DSCP media packet value that should be set for Cisco

TelePresence endpoints?


A.
CS3 (24)

B.
EF (46)

C.
AF41 (34)

D.
CS4 (32)



Answer: D



QUESTION NO: 17

When you configure QoS on VCS, which settings do you apply if traffic through the VCS should

be tagged with DSCP AF41?


A.
Set QoS mode to DiffServ and tag value 32.

B.
Set QoS mode to IntServ and tag value to 34.

C.
Set QoS mode to DiffServ and tag value 34.

D.
Set QoS mode to IntServ and tag value to 32.

E.
Set QoS mode to ToS and tag value to 32.



Answer: C


QUESTION NO: 22

Which two options are valid service parameter settings that are used to set up proper video

QoS
behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)



A.
DSCP for Video Calls when RSVP Fails

B.
Default Intraregion Min Video Call Bit Rate (Includes Audio)

C.
Default Interregion Max Video Call Bit Rate (Includes Audio)

D.
DSCP for Video Signaling

E.
DSCP for Video Signaling when RSVP Fails



Answer: A,C


QUESTION NO: 34

Which two options should be selected in the SIP trunk security profile that affect the SIP

trunk pointing to the VCS? (Choose two.)


A.
Accept Unsolicited Notification

B.
Enable Application Level Authorization

C.
Accept Out-of-Dialog REFER

D.
Accept Replaces Header

E.
Accept Presence Subscription



Answer: A,D


QUESTION NO: 35

Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with
video endpoints registered on both systems. Users find that video endpoints registered on

Call
manager can call each other and likewise for the endpoints registered on the VCS server. The
administrator for Company X realizes he needs a SIP trunk between the two systems for any

video
endpoint to call any other video endpoint. Which two steps must the administrator take to add

the
SIP trunk? (Choose two.)


A.
Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the

VCS server.

B.
Set up a subzone on Cisco UCM with the peer address to the VCS cluster.

C.
Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option

VCS Configuration > Zones > zone.

D.
Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and
Transport set to TCP.

E.
Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco
UCM to communicate.


Answer: A,C




QUESTION NO: 48

Which statement about setting up FindMe in Cisco TelePresence Video Communication Server is

true?


A.
Users are allowed to delete or change the address of their principal devices.

B.
Endpoints should register with an alias that is the same as an existing FindMe ID.

C.
If VCS is using Cisco TMS provisioning, users manage their FindMe accounts via VCS.

D.
A VCS cluster name must be configured.

Answer: D



QUESTION NO: 74

Company X currently uses a Cisco Unified Communications Manager, which has been configured
for IP desk phones and Jabber soft phones. Users report however that whenever they are out of
the office, a VPN must be set up before their Jabber client can be used. The administrator

for
Company X has deployed a Collaboration Expressway server at the edge of the network in an
attempt to remove the need for VPN when doing voice. However, devices outside cannot

register.

Which two additional steps are needed to complete this deployment? (Choose two.)

A.
A SIP trunk has to be set up between the Expressway-C and Cisco UCM.

B.
An additional interface must be enabled on the Cisco UCM and placed in the same subnet at the
Expressway.

C.
The customer firewall must be configured with any rule for the IP address of the external

Jabber
client.

D.
The Expressway server needs a neighbor zone created that points to Cisco UCM.

E.
Jabber cannot connect to Cisco UCM unless it is on the same network or a VPN is set up from
outside.


Answer: A,D


QUESTION NO: 75

A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway
in the DMZ to facilitate VPN-less SIP calls with users outside of the network. However, the

users
report that calls via the VCS are erratic and not very consistent.


What must the administrator configure on the firewall to stabilize this deployment?


A.
The VCS Control should not be on the LAN, but it must be located in the DMZ with the
Expressway.

B.
The firewall at Company X must have all SIP ALG functions disabled.

C.
The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the

same
network that the VCS Control is on.

D.
A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway

and
the VCS Control servers.

Answer: B




QUESTION NO: 78

Company X wants to implement RSVP-based Call Admission Control and move away from the current

location-based configuration.


Where does the administrator go to create a default profile?


A.
System > Call Manager >Clusterwide> Service Parameters > RSVP

B.
System > Service Parameters > RSVP

C.
System > Service Parameters > Call Manager >Clusterwide parameters > RSVP

D.
on each MGCP gateway at all remote locations



Answer: C


QUESTION NO: 79

Where can you change the clusterwide DSCP setting for Cisco Unified Communications
Manager?



A.
enterprise parameters

B.
service parameters

C.
enterprise phone configuration

D.
Ethernet configuration



Answer: B



QUESTION NO: 81

Which two statements about remote survivability are true? (Choose two.)




A.
SRST supports more Cisco IP Phones than Cisco Unified Communications Manager Express in SRST

mode.

B.
Cisco Unified Communications Manager Express in SRST mode supports more Cisco IP Phones
than SRST.

C.
MGCP fallback is required for ISDN call preservation.

D.
MGCP fallback functions with SRST.


Answer: A,D



QUESTION NO: 84

Which three CLI commands are used when configuring H.323 call survivability for all calls?
(Choose three.)


A.
voice service voip

B.
telephony-service

C.
h323

D.
call preserve

E.
call-router h323-annexg

F.
transfer-system


Answer: A,C,D





QUESTION NO: 91

Which two statements about the use of the Intercluster Lookup Service in a multicluster
environment are true? (Choose two.)


A.
Cisco Unified Communications Manager uses the ILS to support intercluster URI dialing.

B.
ILS contains an optional directory URI replication feature that allows the clusters in an ILS

network to replicate their directory URIs to the other clusters in the ILS network.

C.
Directory URI replication does not need to be enabled individually for each cluster.

D.
To enable URI replication in a cluster, check the Exchange Directory URIs with Remote

Clusters
check box that appears in the SIP trunk configuration menu.

E.
If the ILS and directory URI replication feature is disabled on a cluster, this cluster still

accepts ILS
advertisements and directory URIs from other neighbor clusters; it just does not advertise

its local
directory URIs.


Answer: A,B
Explanation:



QUESTION NO: 93

When implementing a dial plan for multisite deployments, what must be present for SRST to

work
successfully?


A.
dial peers that address all sites in the multisite cluster

B.
translation patterns that apply to the local PSTN for each gateway

C.
incoming and outgoing COR lists

D.
configuration of the gateway as an MGCP gateway


Answer: B



QUESTION NO: 95

When using SAF, how do you prevent multiple nodes in a cluster from showing up in the Show
Advance section of the SAF Forwarder configuration?


A.
Configure the publisher node only in the SAF Forwarder configuration page.

B.
Append an @ symbol at the end of the client label value in the SAF Forwarder configuration

page.

C.
Configure the correct node in the EIGRP configuration of the gateway router that is

associated
with the Cisco Unified Communications Manager node.

D.
Configure the SAF Security Profile Configuration to support only a single node.


Answer: B



QUESTION NO: 96

Which statement about the SAF Client Control is correct?


A.
The SAF Client Control is a configurable inherent component of Cisco Unified Communications
Manager.

B.
The SAF Client Control is a non-configurable inherent component of Cisco Unified
Communications Manager.

C.
The SAF Client Control is a non-configurable inherent component of the Cisco IOS Routers.

D.
The SAF Client Control is a configurable inherent component of the Cisco IOS Routers.


Answer: B




QUESTION NO: 103

How many Cisco Unified Mobility destinations can be configured per user?


A.
1

B.
10

C.
4

D.
6


Answer: B




QUESTION NO: 104
When configuring Cisco Unified Mobility, which parameter defines the access control for a

call that
reaches out to a remote destination?


A.
Calling Party Transformation Calling Search Space under Remote Destination Profile

Information

B.
User Local under Remote Destination Profile Information

C.
Rerouting Calling Search Space under Remote Destination Profile Information

D.
Rerouting Calling Search Space under Remote Destination information

E.
Calling Search Space under Phone Configuration


Answer: C



QUESTION NO: 106

In Cisco Unified Communications Manager, where do you configure the default bit rate for

audio
and video devices?


A.
Enterprise Parameters

B.
Region under Region Information

C.
Cisco CallManager service under Service Parameter Configuration

D.
Enterprise Phone Configuration


Answer: C





QUESTION NO: 109

Which statement is true when considering a Cisco VoIP environment for regional configuration?


A.
G.711 requires 128K of bandwidth per call.

B.
G.729 requires 24K of bandwidth per call.

C.
The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.


D.
To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and
only use G.711 between regions.


Answer: B




QUESTION NO: 114

What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified
Communications Manager?


A.
EF/46

B.
CS6/48

C.
AF41/34

D.
CS3/24

E.
CS4/32


Answer: E


QUESTION NO: 115 As per Dumps Answer B is wrong Correct answer is D

A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?

A.
search rules

B.
SIP route pattern

C.
policy service

D.
transform


Answer: D






QUESTION NO: 116 As Per Dumps B&C In Exam Option C was not present but still the option was

to choose 2.
I Selected B&D

Which two statements about Cisco Unified Communications Manager Extension Mobility are true?
(Choose two.)


A.
After an autogenerated device profile is created, you can associate it with one or more

users.

B.
An autogenerated device profiles can be loaded on a device at the same time as a user

profile.

C.
A device can adopt a user profile even when no user is logged in.

D.
A device profile has most of the same attributes as a physical device.

E.
Devices can be configured to allow more than one user to be logged in at the same time.

Answer: B,D


QUESTION NO: 117

When you configure a globalized dial plan, in which three ways can you enable ingress

gateways
to process calls? (Choose three.)

A.
Configure the called-party transformation settings for incoming calls on H.323 gateways.

B.
Configure translation patterns in the partitions used by the gateway calling search space.

C.
Configure SIP trunks between Cisco Unified Communications Manager clusters.

D.
Configure a remote site device pool.

E.
Configure a hunt group.

F.
Configure the gateway with prefix digits to add necessary country and region codes.


Answer: A,B,F
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/dialplan.html#wp115316
6




QUESTION NO: 118

How many nodes can a phone establish a connection to at the same time?


A.
4



B.
3

C.
1

D.
2

Answer: D


QUESTION NO: 119

Company X has a primary and a backup Cisco Unified Communications Manager instance. The
administrator had to do maintenance on the primary node and did a shutdown, which resulted in

a
failover to the backup node. What happens when the primary node comes back online?


A.
The primary node becomes the backup node.

B.
Endpoints detect that the primary is back and reregisters automatically.

C.
The backup node must be shut down first to allow the endpoints to realize that the primary

node is
online again.

D.
Nothing, the endpoints only failover when the node they lose connection to their registered

node.


Answer: B



QUESTION NO: 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To

facilitate
external calls, VCS Expressway is deployed and traversal server zone is set up there. Video
endpoints inside Company X have registered, but are unable to receive calls from outside
endpoints. Which option could be the cause of this issue?


A.
The traversal zone on the VCS Control does not have a search rule configured.

B.
The access control list on the VCS Control must be updated with the IP for the external

users.

C.
When a traversal zone is set up on VCS Control only outbound calls are possible.

D.
The local zone on the VCS Control does not have a search rule configured.


Answer: D



QUESTION NO: 121 Options are Changed
Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio

codec; E. T1 PRI
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)


A.
the same audio codec on both legs of the call

B.
an FXO card

C.
a binding IP address

D.
a hardware transcoder -LTI Transcoder Resource

E.
DSP resources -DSP PVDM Resource

F.
a T1 card


Answer: D,E




QUESTION NO: 122 As per dumps A,C is wrong A & B is the Correct answer

The Cisco Unified Communications system of a company has five types of devices:


•Cisco Jabber Desktop

•CP-7965

•DX-650

•EX-60

•MX-200




Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)


A.
DX-650

B.
Cisco Jabber Desktop

C.
CP-7965

D.
EX-60

E.
MX-200


Answer: A,C



QUESTION NO: 123 I selected B,E
I think is better to go with A, E as per previous discussion here

Refer to the exhibit.



An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:




User A can hear user and vice versa.

User A can hear user C, however user cannot hear user A.

User can heat user C, however user cannot hear user .


Which two properties are the most likely reasons for this issue? (Choose two.)


A.
The Cisco EX60 default gateway of user is missing from the network configuration.

B.
The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.

C.
The Cisco EX60 of user is not responding to requests coming from the TMS server.

D.
The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.

E.
The router does not have a route back from the DMZ to the internal network.


Answer : B,E




QUESTION NO: 124

Which three messages does a Cisco VCS use to monitor the Presence status of endpoints?
(Choose three.) Options are reduced in Exam (A. start-call; B. end-call; C. call-started; D.

registration ) Answer is D


A.
start-call

B.
in-call

C.
end-call

D.
call-ended

E.
call-started


F.
registration


Answer: B,D,F
Reference:
http://www.cisco.com/c/en/us/td/docs/telepresence/infrastructure/articles/vcs_monitors_presen

ce_
status_endpoints_kb_186.html





QUESTION NO: 125 In Dumps it is A,B,E But “ voice register global dn” I think it’s not a

valid command.

Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)


A.
Configure voice register pool.

B.
Configure voice register global dn.

C.
Configure an SRST reference.

D.
Configure a phone NTP reference.

E.
Configure the SIP registrar.

F.
Configure telephony service.


Answer: A,C,E



QUESTION NO: 126

Which three options are overlapping parameters for roaming when a device is configured for
Device Mobility? (Choose three.)


A.
device pool

B.
location

C.
network locale

D.
codec

E.
MRGL

F.
extension


Answer: B,C,E
Explanation:

The overlapping parameters for roaming-sensitive settings are Media Resource Group List,
Location, and Network Locale. The overlapping parameters for the Device Mobility-related

settings
are Calling Search Space (called Device Mobility Calling Search Space at the device pool),

AAR
Group, and AAR Calling Search Space. Overlapping parameters configured at the phone have
higher priority than settings at the home device pool and lower priority than settings at the

roaming
device pool.

Reference: https://supportforums.cisco.com/document/77096/device-mobility



QUESTION NO: 127

An engineer is working on a Cisco VCS Control routing configuration and wants users to be

able
to dial ccnpcollab and have calls routed to ccnpcollab@cisco.com. Which option achieves this
aim?


A.
search rules

B.
transforms

C.
access rules

D.
call policy


Answer: B



QUESTION NO: 128 In Dumps it is MGCP call park feature is not available in MGCP and in SCCP

call park feature is available so Correct Answer is SCCP.

An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?


A.
H.323 gateway

B.
SCCP gateway

C.
H.225 trunk

D.
MGCP gateway

E.
SIP trunk


Answer: B




QUESTION NO: 129 BFCP is Correct answer. In dumps D is Wrong answer.

An engineer must enable video desktop sharing between a Cisco Unified Communications
Manager registered video endpoint and a Cisco VCS registered video endpoint. Which protocol
must be enabled in SIP profile for VCS SIP trunk on Cisco Unified Communications Manager?


A.
RDP

B.
B. H.264

C.
C. H.224

D.
H.263



E.
BFCP


Answer: E




QUESTION NO: 130

An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?

A.
lack of audio bandwidth

B.
lack of video bandwidth

C.
lack of transcoding resources

D.
lack of audio or video bandwidth

E.
lack of conferencing resources


Answer: A



QUESTION NO: 131 In Exam Option was modified

While troubleshooting a connectivity issue between Cisco Unified Communications Manager,
Expressway-C, and Expressway-E, an engineer sees this output in the Expressway-E logs.


Event=”Authentication Failed” Service=”SIP” Src-ip=”10.50.2.1”

Src-port=”25723” Detail=”Incorrect authentication credential for user”

Protocol “TLS” Method=”OPTIONS” Level=”1”


What is the cause of this issue?


A.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Zone username/password.




B.
The Expressway-C Traversal Client username/password do not match the Expressway-E
Traversal Server username/password.

C.
The Expressway-C Traversal Client Zone username/password do not match the Expressway-E

Traversal Zone username/password.

D.
The Expressway-C Traversal Zone username/password do not match the Expressway-E

TraversalClient username/password.

E.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Client username/password.


Answer: B


QUESTION NO: 132 I Selected AAR but as per dumps.Some have suggested “AAR” & “LRG” Please

research & answer.

An engineer is performing an international multisite deployment and wants to create an

effective
backup method to access TEHO destinations in case the call limit triggers. Which technology
provides this ability?


A.
AAR

B.
CFUR

C.
LRG

D.
SRST


Answer: C


QUESTION NO: 134 As per dumps B,D is wrong. I think D,E is correct answer.

Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)


A.
Resolve names outside of the direct control of the Cisco VCS that exist on the public

Internet.

B.
Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.

C.
Traverse a firewall from a protected network to a public or DMZ network.

D.
Apply registration, authentication, and media encryption policies.

E.
Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of
bandwidth.


Answer: D,E



QUESTION NO: 135

Which situation requires TCP port 443 to be open for packets that are sourced from the

Internet
with a destination in the corporate DMZ?




A.
when video endpoints that reside on the Internet require administrative access to the Cisco
Expressway Edge

B.
when you require encrypted calls to endpoints on your corporate LAN

C.
when you want to enable calls to web applications by using HTTP

D.
when you require administrative access to the Cisco Expressway Edge from the Internet


Answer: D


QUESTION NO: 136

Refer to the exhibit.




Which option describes the effect of this configuration?


A.
It implements Cisco Unified CME redundancy.

B.
It configures a standby Cisco Unified E.

C.
It configures failover.

D.
It implements Cisco IOS redundancy.

E.
It creates dial peers.

F.
It implements HSRP.


Answer: A



QUESTION NO: 137

Which two types of trunks can support Cisco Unified Communications Manager? (Choose two.)


A.
switch port trunks

B.
PIMG trunks

C.
SIP trunks

D.
H.225 trunks

E.
CO trunks

F.
POTS trunks


Answer: C,D


QUESTION NO: 138

Which three tests can you perform to verify redundancy in the customer environment? (Choose
three.)

A.
Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is

disconnected.

B.
Verify that all phones are registered to a second subscriber server.

C.
Verify that SCCP fallback is configured in Cisco Unified Communications Manager.

D.
Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.

E.
Verify that media resources fail over to a secondary subscriber server when the publisher

fails.

F.
Verify that the H.323 redundant connection is active.



Answer: A,B,E



QUESTION NO: 139 I selected C. I think "A" will better comparing with "C"

Which option indicates the best QoS parameters for interactive video?


A.
0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning

B.
1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning

C.
1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning

D.
5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning


Answer: C



QUESTION NO: 140

Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)


A.
a calling search space

B.
hosted DN patterns

C.
translation patterns

D.
route patterns

E.
the SIP or H.323 trunk

F.
hosted DN groups


Answer: B,E,F




QUESTION NO: 141

The Cisco Unified Communications system of a company has five types of devices:


•Cisco Jabber Desktop

•CP-7965

•DX-650

•EX-60

•MX-200


Which two types of devices are affected when an engineer changes the DSCP for TelePresence
Calls service parameter? (Choose two.)


A.
a Cisco Jabber Desktop

B.
DX-650

C.
CP-7965

D.
MX-200

E.
EX-60


Answer: D,E



QUESTION NO: 143

Which three devices or applications support call preservation? (Choose three.)


A.
a software conference bridge

B.
Cisco Unified IP Phone 7962G

C.
an annunciator

D.
SIP trunks

E.
JTAPI applications

F.
TAPI applications

G.
CTI applications

H.
third-party H.323 endpoints


Answer: A,B,D


QUESTION NO: 144 in dumps C is Wrong correct answer is B

An engineer is configuring Global Dial Plan Replication and wants to prevent the local

cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes
this task?


A.
Create a block route pattern.

B.
Create a block learned pattern.

C.
Create a block transformation pattern.

D.
Create a block translation pattern.


Answer: B



QUESTION NO: 145

An administrator is visiting a remote site that has on-net calls with headquarters and one

voice
gateway for PSTN calls. When using RTMT to monitor the bandwidth utilization of the remote

site,
the administrator notices the OutOfResources counter for the site in LBM has been increasing
slowly in last two weeks, but no call failure reports have been sent from this site. Which

description
about this issue is true?


A.
The bandwidth settings of the site are fulfilling on-net call volume.

B.
AAR is routing some of the calls.

C.
The location-based CAC does not work properly.

D.
The LBM service is malfunctioning.


Answer: B



QUESTION NO: 146 in Exam options are reduced to choose 2
I selected G,C from the below options

An engineer is configuring URI calling within the same cluster. Which four actions must be

taken to
accomplish this configuration? (Choose four.)


A.
Configure SIP route patterns.

B.
Configure the directory URI partition and calling search space.

C.
Associate the directory URIs to directory numbers.

D.
Activate the URI service in Cisco Unified Serviceability.

E.
Configure SIP trunk.

F.
Assign directory URIs to users.

G.
Configure the SIP profile.

H.
Configure the URI service parameters.


Answer: B,C,F,H



QUESTION NO: 148 I Selected A,E

You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site.
During a network failure between the remote site and the central office, some of the phones
located at the remote site are unable to make phone calls. Which two options are potential

causes
of the problem? (Choose two.)


A.
The site has exceeded the number of SRST endpoints supported by the voice gateway.

B.
The ccm-manager fallback command is configured incorrectly on the voice gateway.

C.
Phones at the remote site are assigned to the incorrect device pool.

D.
The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.

E.
The site has exceeded the number of simultaneous calls allowed in SRST mode.


Answer: A,E


QUESTION NO: 149

Options Modified B,D Is the Correct answer from modified option
(A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833);
D. Sip delay offer )

Which function can be implemented without MTP resources?


A.
DTMF relay conversion

B.
terminating a media stream that uses the same codec

C.
music on hold

D.
SIP early offer

Answer: B




QUESTION NO: 150

An engineer is setting up a Cisco VCS Cluster with SIP endpoints only. While configuring the
Cisco VCS peers, which signaling protocol is used between peers to determine the best route

for
calls?


A.
SIP

B.
H.323

C.
SCCP

D.
MGCP


Answer: B
Reference:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X87/Cis

co-VCS-Cluster-Creation-and-Maintenance-Deployment-Guide-X8-7.pdf
(page 4, basic Configuration is done, third point)




QUESTION NO: 151 Please check the documents I think SAF will not be a dialing function.
Which three globalization dialing functions are enhanced in Cisco Unified Communications
Manager 7.x and later? (Choose three.)

A.
MGRL

B.
TEHO

C.
CER

D.
AAR

E.
SAF

F.
click-to-call

Answer: B,C,D


QUESTION NO: 152 One command is to verify SCCP phone and another command to view SIP IP

phones In dumps it wrong answer C,E is Correct answer

Which two commands verify Cisco IP Phone registration? (Choose two.)


A.
show telephony-service ephone-dn

B.
show voice register session-server

C.
Show ephone registered

D.
showccm-manager hosts

E.
show sip-ua status registrar


Answer: C,E
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/troubleshooting/guide/ts_phreg.html
(see the steps)



QUESTION NO: 153 I selected F . DNS Server

In Exam options are reduced to choose 1,


Refer to the exhibit.




An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP
address and the system name. After logging into the Cisco VCS Expressway admin page, the
engineer sees this output. Which four options must be configured to complete the Cisco VCS
Expressway system configuration? (Choose four.)


A.
NTP server

B.
SIP server

C.
LDAP server

D.
security certificate

E.
DHCP server

F.
DNS server

G.
SIP URI

H.
Cisco Unified Communications Manager IP address



Answer: A,C,D,F




QUESTION NO: 154

After forgetting to log out of his IP phone in the main office, an Extension Mobility user is

unable to
log in to a different IP phone at a remote office. Which option is a possible reason for the

problem?

A.
The phone at the remote location is a different model than the phone in the user’s main

office.

B.
The user’s Extension Mobility profile is misconfigured.

C.
The user can log in to only one device at a time.

D.
The device pool is misconfigured.


Answer: C



QUESTION NO: 155

Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?


A.
SRST without MGCP fallback

B.
SRST with VoIP dial peers to Cisco Unified Communications Manager Express

C.
SRST with MGCP fallback

D.
Cisco Unified Communications Manager Express in SRST mode


Answer: D



QUESTION NO: 157

An engineer is configuring a new DX-80 in Cisco Unified Communications Manager. Where can an
engineer verify the default DSCP value of AF41?


A.
enterprise phone configuration

B.
common phone profile

C.
service parameters

D.
enterprise parameters


Answer: C



QUESTION NO: 158

Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit
numbers. Which option should be configured if you want outgoing calls from 4-digit internal

directory number to be presented as a 10-digit number?

A.
calling party transformation pattern

B.
AAR group

C.
translation pattern

D.
route pattern


Answer: A



QUESTION NO: 159

Which three configuration settings are included in a default region configuration? (Choose

three.)


A.
Immersive Bandwidth

B.
Video Call Bandwidth

C.
Audio Codec

D.
Link Loss Type

E.
Real Time Protocol

F.
Location Description


Answer: B,C,D
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/bccm-

801cm/b02regio.html#wp1077135


QUESTION NO: 160 In Exam Options to choose 1.

I selected "A"


Refer to the exhibit.



Which three statements about when user A calls user using SIP are true? (Choose three.)


A.
SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.

B.
Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking

option key.

C.
Cisco VCS Control and Cisco VCS Expressway support static NAT.

D.
Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.

E.
RT and RTCP ports must be opened at the firewall from internal to DMZ and vice versa

F.
The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.

Answer: A,B,E



QUESTION NO: 161

When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose

three.)


A.
The SIP route patterns have not been properly configured.

B.
The Tomcat certificates do not match.

C.
The Cisco Unified Resource Identifier service needs a restart.

D.
The ILS authentication password does not match.

E.
The cluster ID does not match.

F.
One cluster is using TLS certificate, and the other is using Password.


Answer: B,D,F

@Juan 102q is not valid
Please Check 161q Dumps

Question 123
@KMX
Thank you for breaking down the different options. My question though, is that B is saying that RTP/RTCP traffic is only allowed to flow towards user C. The problem states that User C is the one who cannot hear. Shouldn't it be the other way around?
It really seems like E is the only valid option.

Based on Steve's questions from USA what answers can you give for these??
What steps are needed to set up h.323 to SIP and vise versa, pick two.
a. Protocol>Sip>Sip on
b. H.323-SIP internetworking mode On
c. H.323-SIP internetworking mode Off
d. H.323-SIP internetworking mode Registered Only
e. Protocol>H.323>On
f. Protocol>Sip>configuration>Sip On

An engineer must resolve a video call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources

Could you confirm if this dump 102q is valid?

@CCNPCollab
Question N0: 109
As far as I know CUCM calculates bandwidth with headers (g.729: 8k+16k headers = 24k total bandwidth) in the Location section for sure. I do not know if it calculates bandwidth similarly in Region too...

123. BE
A is wrong because A hears C, so C has correct default gateway.
C is wrong because TMS is needed for conferences.
D - problem is not related to call setup (INVITE)

131. Choice between very similar answers
The Expressway-C Traversal Client username/password do not match the Expressway-E Traversal Server username/password
The Expressway-C Traversal client zone username/password do not match the Expressway-E Traversal server zone username/password
I choose second. Citation from VCS admin guide: “The traversal server zone for the VCS client must be configured with the client's authentication”. So the password is configured for the zone, not for the client or server.

160 (select ONE, answers were slightly changed)
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa.

Ignore last comment. Was meant to be posted in 300-207. Sorry

Congratulations to all those who have passed.
Just a confirmation of ne of the D&D's

Risk Rating Calculation
Risk rating is a quantitative measure of your network's threat level before IPS mitigation. For each event fired by IPS signatures, Cisco IPS Sensor Software calculates a risk rating number. The factors used to calculate risk rating are:

• Signature fidelity rating: This IPS-generated variable indicates the degree of attack certainty.

• Attack severity rating: This IPS-generated variable indicates the amount of damage an attack can cause.

• Target value rating: This user-defined variable indicates the criticality of the attack target. This is the only factor in risk rating that is routinely maintained by the user. You can assign a target value rating per IP address in Cisco IPS Device Manager or Cisco Security Manager. The target value rating can raise or lower the overall risk rating for a network device. You can assign the following target values:

– 75: Low asset value

– 100: Medium asset value

– 200: Mission-critical asset value

• Attack relevancy rating: This IPS-generated value indicates the vulnerability of the attack target.

• Promiscuous delta: The risk rating of an IPS deployed in promiscuous mode is reduced by the promiscuous delta. This is because promiscuous sensing is less accurate than inline sensing. The promiscuous delta can be configured on a per-signature basis, with a value range of 0 to 30. (The promiscuous delta was introduced in Cisco IPS Sensor Software Version 6.0.)

• Watch list rating: This IPS-generated value is based on data found in the Cisco Security Agent watch list. The Cisco Security Agent watch list contains IP addresses of devices involved in network scans or possibly contaminated by viruses or worms. If an attacker is found on the watch list, the watch list rating for that attacker is added to the risk rating. The value for this factor is between 0 and 35. (The watch list rating was introduced in Cisco IPS Sensor Software Version 6.0.)

Could someone help me answer the below question please.

Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.

Is it B or C? If so why? any particular reason...

Thanks.

Passed today with 897. Same questions in 161 just be careful on your answers.

@Adam:
Cisco CallManager Extension Mobility supports only one login at a time on a device. Subsequent logins will fail.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a0080153e60.html

@Scotty and @Nono.
Can you collect-gather all the correct answers for all the post you have replied and you believe that are wrong?? It would be much easier if you posted all together nice and clean. Thank you in advance

Placing my exam tomorrow. Hope the questions in 161 still the same.

@Hari

Can you give me your advice on this question, this another grey area..

After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the problem?

A. The phone at the remote location is a different model than the phone in the user's main office.

B. The user's Extension Mobility profile is misconfigured

C. The user can log in to only one device at a time.

D. The device pool is misconfigured.

Note: Dump answer is A but other said that the answer is C.

Third time's a charm! Passed (barely!) with a 869 this morning.
Here's the breakdown of the sections:

VCS Control: 75%
Collab Edge: 50%
CUCM Video Service: 100%
Centralized Call Processing Redundancy: 80%
Multi-site Dial Plan: 700%
CCD/ILS: 100%
Video Mobility: 83%
Bandwidth and CAC: 29% (proof that this test is jacked, this is part of my job and has been for years!)

Big heads up, they've changed a few questions again!
Here's my notes on what they've changed and what I answered on the debated questions that I had today:

6.) No longer asks about avoinind unneccessary intrworking, instead asks what steps are needed to set up h.323 to SIP and vise versa, pick two: (these are as best as I can remember)
a.) Protocol>Sip>Sip on
b.) H.323-SIP internetworking mode On
c.) H.323-SIP internetworking mode Off
d.) H.323-SIP internetworking mode Registered Only
e.) Protocol>H.323>On
f.) Protocol>Sip>configuration>Sip On
I chose D and F.

109.) I answered B

116.)Options are slightly different:
a.) After an Autogenerated device profile is created you can associate it with one or more users.
b.) An autogenerated device profile can be loaded on a device at the same time as a user profile
c.) When Logged off the device can be set to use an autogenerated profile or a user defined profile.
d.) A device profile has most of the same attributes as a physical device
e.) Devices can be configured to allw more than one user to be logger in at the same time.
I answered C and D

118.) How many CUCM nodes cana Skinny phone establisha SCCP connection to at the same time?
I answered 1

120.) I answered D, Local Zone search
121.) I answered Hardware Transcoder and PVDM/DSP
122.) I answered Jabber and DX-650
123.) I answered A and E
124.) Choose 1, did not have B or D. I answered F
125.) I mistakely answered BCE when I should have answered ACE
128.) Answered B, SCCP
129.) Answered E, BFCP
130.) BIG CHANGE!!!! It now reads "An engineer must resolve a VIDEO call failure...." With that in mind I answered B, Lack of Video Bandwidth
131.) Answered B. Be careful, they changed one answer to read the exact same as B but with "Client Zone" and "Server Zone"
132.) Answered C, LRG
134.) Answered D, Apply registration, auth and policies, and E, Manage Bandwidth to restrict....
135.) Answered D
136.) Answered C
138.) Answered ABE
139.) Answered C
140.) Answered BEF
144.) Answered B, Block Learned Pattern
145.) Answered B
146.) Choose Two, got the following choices:
a) Configure SIP route patterns
b) Configure SIP trunk
c) Assign directory URIs to users
d) Configure the SIP profile
I answered C and D
148.) Answer C has been changed to "Some phones at the remote site are assigned to a device pool which does not have an SRST Reference assigned.
With that in mind I picked A and C

149.) The answers I was given:
a) DTMF Relay conversion
b)H.323 Outbound Fast Start
c) SIP early offer
d) IPv4 to IPv6 conversion
e) Multicast MoH
I answered A and D

152.) I answered C and E
153.) Pick one from C E F and H, I chose F, DNS
154.) I answered A
155.) I answered D
158.) A now reads "calling party transformation pattern filed in Route Pattern" and D now reads "Route Pattern field in Route Pattern".
I chose A

160.) Answered A
161.) Answered BDF

New Question!:

Don't remember the wording, but basically "An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com", what happens to the call.

a) Route fails
b) Sent to Cisco.com
c) sent as CCNPCOLAB
d) Sent as CCNPCOLAB@cisco.com

I chose D

I Hope this helps everyone, good luck!
If anyone has any questions about any other questions let me know and I'll see what I can help with.

@Hari:
sorry I misunderstood ur answer before

@Scotty:
to configure search rule, you need to define source and target.

the problem, call from "outside" to "inside"
source: traversal zone
target : local zone

when you done, the search rule will belong to target.

@r.zelig

premium file has all the questions. See what people have posted in here as to what the correct answers should be.

eny bady pass the sxem with the primum file ?160q

@ Adam
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder
E.DSP resources
F.a T1 card
Answer: D,E

My Understanding : Same

Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer

Answer: B

My Understanding is B and C
Reason for C: "The following restriction exists for multicast music on hold (MOH) When an MTP resource gets invoked in a call leg at a site that is using multicast MOH, the caller receives silence instead o music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH"

Reference : http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsgd-712-cm/fsmoh.html

@Chase @Hari

Can you shed some light on these 2 questions, I take my exam on July 5th and feel like I am getting very close to being ready.

Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder
E.DSP resources
F.a T1 card

Answer: D,E



Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer

Answer: B

@D4D Re q139 Below are Cisco guidelines but the question is asking which one is best and I believe A is better than C which corresponds to the Cisco guidelines

When provisioning for Interactive Video (IP Videoc onferencing) traffic, the following guidelines are recommended:
• Interactive Video traffic should be marked to DSCP AF41; excess Interactive-Video traffic can be marked down by a policer to AF42 or AF43.
• Loss should be no more than 1 %.
• One-way Latency should be no more than 150 ms.
• Jitter should be no more than 30 ms.
• Overprovision Interactive Video queues by 20% to accommodate bursts Because IP Videoconferencing (IP/VC) includes a G.711 audio codec for voice, it has the same loss, delay, and delay variation requirements as voice, but the traffic patterns of videoconferencing are radically different from voice.

@Nono, @Hari
Re q120. I don't see what the local zone has to do with receiving outside calls as this is used only for internal calls so don't think the answer could be D. I believe it is A as from a VCS control perspective all calls are outgoing, even incoming calls. The outside call hits the VCS expressway which in turn notifies the VCS Control which makes the necessary settings to make the call appear as an outgoing call. Otherwise the firewall would stop all incoming calls as they were not initiated from the internal network.

Explanation:
The traversal client (VCS Control) constantly maintains a connection via the firewall to a designated port on the
traversal server (VCS Server). This connection is kept alive by the client sending packets at regular intervals
to the server. When the traversal server receives an incoming call for the traversal client, it uses
this existing connection to send an incoming call request to the client. The client then initiates the
necessary outbound connections required for the call media and/or signaling.
This process ensures that from the firewall’s point of view, all connections are initiated from the
traversal client inside the firewall out to the traversal server.

@Chase, Scotty, Nono or anyone else who have passed.
Hey guys, you seem to be pretty good, can you help me with the following as there seems to be a lot of uncertainty

QUESTION NO: 149
Which function can be implemented without MTP resources?
A. DTMF relay conversion
B. terminating a media stream that uses the same codec
C. music on hold
D. SIP early offer

Is the correct answer B or C?

QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning

Is the correct answer A as the question implies "best"?

Also the below seems to be an extra question outside the dump?

which 2 things do not use MTP
a> h.323 fast start
b> IPV6 -IPV4 
c> DTMF inband RTP-NTE (rfc2833)
d> delayed offer h.323 

Is the correct answer B and C?

Thanks
Ramesh

guys, where can I get the 161 dump? It doesn't seem to be listed

@hari:
120: D
the main prob is "unable to receive calls from outside endpoints".
A. is the solution for unable to calls from inside to outside endpoints
C. you need to add search rule too

123: E
(the exam only ask for 1 answer)

*135,138,160: same

___Question 121___

Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)

A. the same audio codec on both legs of the call
B. an FXO card
C. a binding IP address
D. a hardware transcoder
E. DSP reources
F. a T1 card


Note: Dump answer are D & E. But upon searching a hardware transcoder is different from hardware MTP.

Hardware MTP configured on Cisco IOS routers:

+DSP resources are required. Configure this MTP type by using the maximum session hardware command. The maximum number of sessions is derived from the number of installed DSP resources on the Cisco IOS router.

+Use of the same audio codec but different packetization on both call legs is possible.


__Question 139___

Which option indicates the best QoS parameters for interactive video?

A. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning

Note: Dump answer is C but other said that it should be A?


__Question 148___

You have deployed a Cisco 2821 ISR to poerform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phone located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two)

A. The site has exceeded the number of SRST endpoints supported by the voice gateway.

B. The ccm-manager fallback command is configured incorrectly on the voice gateway

C. Phone at the remote site are assigned to the incorrect device pool

D. The ccm-manager fallback-mgcp cimmand is configured incorrectly on the voice gateway.

E. The site has exceeded the number of simultaneous calls allowed in SRST mode.


Note: Dump answer are B & D but other said that the answer are A & E.


__Question 154__

After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the problem?

A. The phone at the remote location is a different model than the phone in the user's main office.

B. The user's Extension Mobility profile is misconfigured

C. The user can log in to only one device at a time.

D. The device pool is misconfigured.

Note: Dump answer is A but other said that the answer is C.


___Question 155___

Which solution is needed to enable presence and extension monility to branch office phones during a WAN failure?

A. SRST without MGCP fallback
B. SRST with VOIP dial peers to CME
C. SRST with MGCP fallback
D. CUCM Express in SRST mode

Note: Dump answer is C but other said that the answer is D.

Any advise guys? I will take the exam this coming Monday. Thanks!

@ Chase..Thank you for the answers.
I have same opinion for 123, 135,138 and 160.

Q123: A and E
Q135: D
Q138: A B E
Q160: A

But Q120: D, Local Zone we will use for internal calls in the VCS. For external calls we need to have a traversal zone.

We have 2 options with traversal Zone(A&C).
C. When a traversal zone is set up on VCS Control only outbound calls are possible.(Not outbound only, Both inbound and outbound are possible)

So i feel it is A.

Q146 I got
b. Configure the directory uri partition and calling search space
c. Associate the directory uris to directory numbers
f. Assign directory uris to users
g. Configure the sip profile

page 2 steps 1 - 4

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.pdf

What you guys think?

@Hari
Q120: D
Q123: A and E
Q135: D
Q138: A B E
Q160: A

I see that folks are discussing questions numbered near 160, but this file says it has only 102 questions and is dated 10/2015. Are all of you discussing the same file? If not, where do I get the file you are discussing?

@ Nono, Chase, Scotty

What are your answers on the below 5 questions.

QUESTION 120

Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?

A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.


QUESTION 123

An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user В and vice versa.
User A can hear user C, however user С cannot hear user A.
User В can heat user C, however user С cannot hear user В.


Which two properties are the most likely reasons for this issue? (Choose two.)

A. The Cisco EX60 default gateway of user С is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E. The router does not have a route back from the DMZ to the internal network.

QUESTION 135

Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?

A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet

QUESTION 138

Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)

A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.

QUESTION 160

Refer to the exhibit.

Which three statements about when user A calls user С using SIP are true? (Exam asks only one)

A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.

D'oh! Looks like I'm talking to myself!

I meant to say @Chase!

@ Scotty and Nono,

Q146. Thanks for the clarification. Yes it is E and F.

@ Scotty...Q95 : Thanks for giving the answer on this.

@ Steve:

Congrats on passing, let us know if there was a grace period!

Would it be possible for you to go through your list from the 23rd and verify which answers were correct?

Thanks!

@Bari: yes

@Hari: Like Scotty says you need to check "Use Fully Qualified Domain Name in SIP Requests" in sip profile (the default is unchecked).

@Hari re question 95

Tip If more than one Cisco Unified Communications Manager node displays in the Selected Cisco Unified Communications Managers pane under the Showed Advanced section, append @ to the client label value; otherwise, errors may occur because each node uses the same client label to register with the SAF forwarder.

@Hari

Step 3 refers to assigning but the possible answer refers to configuring so I don't think it is this. I still think it is E and F as SIP profile is required as part of the Intra Cluster setup as per the referenced document. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.pdf

I passed the exam after the 3rd attempt. with 888.

Hi Nono,
Congratulations for passing this difficult exam. Can you help to double check this again.

An engineer is configuring URI calling within the same cluster. Which two actions must be taken to accomplish this configuration? (Choose two.)

A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Activate the URI service in Cisco Unified Serviceability.
D. Configure SIP trunk.
E. Assign directory URIs to users.
F. Configure the SIP profile.
G. Configure the URI service parameters.

Tricky part in the question is "within the same cluster". SIP Profile is needed for Inter Cluster URI Dialing.

So my choice is E and B

Step 1: Assign Directory URI to Users (Option E)
Step 2: Associate Directory URI with Directory Numbers (This option not available)
Step 3:Assign the default directory URI partition to an existing partition that is located in a calling search space(Option B)


Reference : http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/configure_uri_dialing.pdf

Just passed with a 906. Was my 3rd attempt. My CCNP expired Saturday, sure hope there was a grace period.

nono
Which file did you used ? premium ?

Well... i barely passed exam with 90X.

here's some modified Question i got in my exam:

An engineer is configuring URI calling within the same cluster. Which two actions must be taken to accomplish this configuration? (Choose two.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Activate the URI service in Cisco Unified Serviceability.
D. Configure SIP trunk.
E. Assign directory URIs to users.
F. Configure the SIP profile.
G. Configure the URI service parameters.

My choice at that time (E,F)
*i decided to follow configuration guide order

Which statements about when user A calls user С using SIP are true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Cisco VCS Control and Cisco VCS Expressway support static NAT.
C. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
D. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa

My choice at that time (B), but after cool down..
I think the correct answer is (A)

afaik, VCSC don't have static NAT and dual interface option
VCSE's indeed use static NAT (if you want deploy it inside NAT)
VCSE's use Advanced Networking/dual interface (if you want deploy it behind NAT) but it's not a MUST (if your NAT device support it)
RTР and RTCP ports must be opened (if it's blocked you simply don't receive any video / audio)
SIP TCP/TLS ports, this one is requirement for SIP traversal calls and the question asking for SIP.


*I hope it'll help you a bit and sorry for my bad english :D

Hi..Can someone help to find correct answers for below questions

QUESTION 75

A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway in the DMZ to facilitate VPN-less SIP calls with users outside of the
network. However, the users report that calls via the VCS are erratic and not very consistent.What must the administrator configure on the firewall to stabilize this deployment?

A. The VCS Control should not be on the LAN, but it must be located in the DMZ with the Expressway.
B. The firewall at Company X must have all SIP ALG functions disabled.
C. The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the same network that the VCS Control is on.
D. A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway and the VCS Control servers.

QUESTION 95

When using SAF, how do you prevent multiple nodes in a cluster from showing up in the Show Advance section of the SAF Forwarder configuration?

A. Configure the publisher node only in the SAF Forwarder configuration page.
B. Append an @ symbol at the end of the client label value in the SAF Forwarder configuration page.
C. Configure the correct node in the EIGRP configuration of the gateway router that is associated with the Cisco Unified Communications Manager node.
D. Configure the SAF Security Profile Configuration to support only a single node.

QUESTION 120

Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?

A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.


QUESTION 123

An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user В and vice versa.
User A can hear user C, however user С cannot hear user A.
User В can heat user C, however user С cannot hear user В.
Which two properties are the most likely reasons for this issue? (Choose two.)

A. The Cisco EX60 default gateway of user С is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E. The router does not have a route back from the DMZ to the internal network.

QUESTION 135

Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?

A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet

QUESTION 138

Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)

A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.

QUESTION 149

Which function can be implemented without MTP resources?


A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833);
D. delayed offer h.323

QUESTION 160

Which three statements about when user A calls user С using SIP are true? (Exam Asks only One)

A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.

@Ramesh
Re Q146 Hope this helps.

Step 1
Assign directory URIs to the users in your network.
Step 2
Associate the directory URIs to directory numbers by assigning both a primary extension and phone to the users in your network.
Step 3
Assign the default directory URI partition to an existing partition that is located in a calling search space by doing the following:

In Cisco Unified CM Administration, choose System > Enterprise Parameters
For the Directory URI Alias Partition enterprise parameter, choose an existing partition that is in an existing calling search space.
Set the URI Dialing Display Preference service parameter for URI dialing as URI for calling display in call park display URI of the calling party. DN is the default setting for the service parameter.

Step 4
Configure the SIP profiles in your network by configuring the following fields in the SIP Profile Configuration window:

Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network.
Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.

Note At this point, intracluster URI dialing is configured. The remaining steps are used to configure intercluster URI dialing.
Step 5
For all the SIP trunks in your network, configure whether the network uses blended addressing by configuring the Calling and Connected Party Info Format drop-down list box in the Trunk Configuration window.
Step 6
Set up ILS on all the clusters in your network.
Step 7
Enable intercluster URI dialing with ILS by checking the Exchange Directory URI Catalogs with Remote Clusters check box in the Intercluster Directory URI Configuration window.
Step 8
In the Intercluster Directory URI Configuration window, create a route string that remote clusters will use to route to this cluster.
Step 9
Configure SIP route patterns that match the route strings for the remote clusters in your ILS network.
Step 10
Associate the SIP route patterns that you created to an outbound SIP trunk or route list.
Step 11
If you are connecting your ILS network to a Cisco TelePresence Video Communications Server, or a third-party call control system, import directory URI catalogs from the other system into Cisco Unified Communications Manager.
Step 12
If your deployment uses digit transformations to transform calling party directory numbers, configure calling party transformation patterns and apply them to the Inbound Call Settings for the phone or device pool. This configuration is used for intercluster calls.
Step 13
If you applied digit transformation patterns in the previous step, configure calling party transformation patterns for the Outbound Call Settings for the phone or device pool. This configuration is used for intracluster calls.

Thank you Scotty.

I suspect A & D to be the answer. Lets see....

QUESTION 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.

@CCNPCollab
The following suggests one or the other but not both.

Within a centralized call processing cluster with N sites, you can implement Tail-End Hop-Off (TEHO) using one of the following methods:

–TEHO with centralized failover

This method involves configuring a set of N route patterns in a global partition, with each pattern pointing to a route list that has the appropriate remote site route group as the first choice and the central site route group as the second choice.

–TEHO with local failover

This method involves configuring N sets of N route patterns in site-specific partitions, with each pattern pointing to a route list that has the appropriate remote site route group as the first choice and the local site route group as the second choice.

@ ANON12345
Thank you so much for this and congrats on passing, its going to be very helpful for this horrible test

can you confirm the 161 dumps you used that had the wrong answers was from Actual tests or was this the Exam collection premium file?
Asking as both these vendors have different answers even to the ones outside of the answers you posted?

Also could you tell me which answers you used for the below questions?

QUESTION 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish
this configuration? (Choose four.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Associate the directory URIs to directory numbers.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.

Answer: As people are saying the exam only asks for two answers, was it C and F?


QUESTION 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.

Correct answer A and F?


QUESTION 148
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network
failure between the remote site and the central office, some of the phones located at the remote site are unable
to make phone calls. Which two options are potential causes of the problem? (Choose two.)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway.
C. Phones at the remote site are assigned to the incorrect device pool.
D. The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.

Answer A and E?


Thanks a lot

Ramesh

Please help me to answer the below

QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.

Question 83:
When considering CUCM failover, how many backup servers can be configured in a CUCM Group?
The answer says 3, and of course you can configure 3 servers on a CUCM group, but only 2 of them will be backup servers.
Am I wrong?

@anon12345
My friend thanks a lot for sharing the right questions!!!!
You mean that all other questions from 161q are correct and only the ones you posted are wrong??

Q83:

Answer: C

A Cisco Unified Communications Manager Group specifies a prioritized list of up to three Cisco Unified
Communications Managers. The first Cisco Unified Communications Manager in the listserves asthe primary
Cisco Unified Communications Manager for that group, and the other members of the group serve assecondary
and tertiary (backup) Cisco Unified Communications Managers

REF:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100/CUCM_BK_C95ABA82_00_admin-guide-100_chapter_0100.pdf
PAGE 1

Passed the test today with a 906. Every question is in the 161q dump, but CHECK THE ANSWERS!!!!! Many of the answers in the dump are wrong.

Good luck.

A couple typos from below:

Question 115: transforms, not transfer

____QUESTION 140____

Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)

A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
ANSWER: A,B,C

I took the test and passed with over 900. Every question was from 161q. There were a LOT of answers that were wrong in the dump. After a lot of research, here were my corrections.

____QUESTION 57____
A new DX650 IP Phone with MAC address D0C7.8914.132D, IP address is 172.18.32.119 has been added to the Cisco Unified Communications Manager, but is not registering properly. What is causing the failure?

A. The DX650's MAC address is incorrect in the Cisco UCM
B. The location Hub_None has not been activated
C. The DX650 is the incorrect calling search space
D. The DX650 Phones does not support SIP
E. Device Pool cannot be default

ANSWER: A

____QUESTION 109____
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
ANSWER: B

____QUESTION 115____
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which component allows for standardized caller addresses between the endpoints?
A. search rules
B. sip route pattern
c. policy service
D. transfer
ANSWER: D

____QUESTION 116____
Which two statements about Cisco Unified Communications Manager Extension Mobility are true?
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. A device can adopt a user profile even when no user is logged in.
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time
ANSWER: C,D

____QUESTION 122____
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200

Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)

A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200

ANSWER: A,B

____QUESTION 123____
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versa.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A. The Cisco EX60 default gateway of user is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E. The router does not have a route back from the DMZ to the internal network.
ANSWER: A,E

____QUESTION 125____
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.
ANSWER: B,C,E

____QUESTION 128____
An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?

A. H.323 gateway
B. SCCP gateway
C. H.225 trunk
D. MGCP gateway
E. SIP trunk
ANSWER: B

____QUESTION 129____
An engineer must enable video desktop sharing between a Cisco Unified Communications Manager registered video endpoint and a Cisco VCS registered video endpoint. Which protocol
must be enabled in SIP profile for VCS SIP trunk on Cisco Unified Communications Manager?
A.RDP
B. H.264
C. H.224
D H.263
E. BFCP
ANSWER: E

____QUESTION 132____
An engineer is performing an international multisite deployment and wants to create an effective backup
method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. CFUR
C. LRG
D. SRST
ANSWER: C

____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D

____QUESTION 134____
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
ANSWER: D,E

____QUESTION 135____
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a
destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway
Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
ANSWER: D

____QUESTION 138____
Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is
disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.
ANSWER: A,B,E

____QUESTION 140____
Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
ANSWER: B,E,F


____QUESTION 142____
A presales engineer is working on a quote for a major customer and must evaluate how many cisco VCS Expressway traversal call licenses for which to plan.
Calls to and from which three routes must the engineer include in the tally?
A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS
ANSWER: C,D,F

____QUESTION 144____
An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes this
task?
A.Create a block route pattern.
B.Create a block learned pattern.
C.Create a block transformation pattern.
D.Create a block translation pattern.
Answer: B

____QUESTION 145____
An administrator is visiting a remote site that has on-net calls with headquarters and one voice
gateway for PSTN calls. When using RTMT to monitor the bandwidth utilization of the remote site,
the administrator notices the OutOfResources counter for the site in LBM has been increasing
slowly in last two weeks, but no call failure reports have been sent from this site. Which description
about this issue is true?

A.The bandwidth settings of the site are fulfilling on-net call volume.
B.AAR is routing some of the calls.
C.The location-based CAC does not work properly.
D.The LBM service is malfunctioning.
ANSWER: B

____QUESTION 152____
Which two commands verify Cisco IP Phone registration? (Choose two.)
A.show telephony-service ephone-dn
B.show voice register session-server
C.show ephone registered
D.show ccm-manager hosts
E.show sip-ua status registrar
ANSWER: C,E

____QUESTION 153____
An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP
address and the system name. After logging into the Cisco VCS Expressway admin page, the
engineer sees this output. Which four options must be configured to complete the Cisco VCS
Expressway system configuration? (Choose four.) ***EXAM ASKS FOR 1

A.NTP server
B.SIP server
C.LDAP server
D.security certificate
E.DHCP server
F.DNS server
G.SIP URI
H.Cisco Unified Communications Manager IP address
ANSWER: F

____QUESTION 158____
Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit
numbers. Which option should be configured if you want outgoing calls from 4-digit internal directory
number to be presented as a 10-digit number?
A.calling party transformation pattern
B.AAR group
C.translation pattern
D.route pattern
ANSWER: A

____QUESTION 160____
Which three statements about when user A calls user using SIP are true? (Choose three.) **EXAM ASKS FOR 1
A.SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B.Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking
option key.
C.Cisco VCS Control and Cisco VCS Expressway support static NAT.
D.Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
E.RTP and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F.The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
ANSWER: B


____QUESTION 161____
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password
ANSWER: B,D,F

Good luck guys. Check this blog out for the answer justifications: https://italchemy.wordpress.com/2016/06/20/cipt2-300-075-exam-helpful-information/

All, I got 100% on my CUCM video service parameters and I chose this

¨¨¨¨QUESTION 122¨¨¨¨¨¨
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)

A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
My Answer: A,B

@paul

Thank for your help.

Can you please also take some time answering this other questions:


¨¨¨¨¨¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints

inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.

Dumb select option: "...The access control list on the VCS Control must..."
I saw other users selecting option D.
Traversal zone is for external call, that mean outside network. Local zone is for local network. in this case the question is for "Outside Call". I think D is correct answer.

I selected "The traversal zone on the VCS Control does not have a search rule configured" but I failed the exam.
Not sure which is the correct one.


¨¨¨¨Question 123¨¨¨¨¨¨

An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user B and vice versa.
User A can hear user C, however user C cannot hear user A.
User B can hear user C, however user C cannot hear user B.

Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user C is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user C is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E.The router does not have a route back from the DMZ to the internal network.


My Answer : C and E but I failed the exam.



¨¨¨¨¨¨QUESTION 125¨¨¨¨¨¨
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.

Seeing a lot of conflicting answers from everyone.
Dump answer: B / D / E
I selected options: BCE.
Just refer to the SRST admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf
page 190

But I failed the exam, not sure which are the corrects, any thoughts?

****QUESTION 128******

An administrator is setting up analog phones that connect to a Cisco VG310. Which type of gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the administrator set up to

allow the phones to have the call pickup feature?
A. H.323 gateway
B. SCCP gateway
C. H.225 trunk
D. MGCP gateway
E. SIP trunk

Dump: H323 GW
But I selected option: SCCP Gateway
Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-series-gateways/product_data_sheet09186a00801d87f6.html - Table 1

But I failed the exam, not sure which is the correct, any thoughts?

****QUESTION 134******
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.

Dumb answer: B D
My final answer: C and D
Reason: Per Cisco's VCS Administrator guide, "Subzones are used to control the bandwidth used by various parts of your network, and to control the VCS's registration, authentication, and media encryption

policies." So D for sure. E maybe, however it seems tricky because I don't think you can manage bandwidth limits based on the type of endpoint (like standard definition). Bandwidth is managed based on

Within subzone, to/from other subzones, and total bandwidth of all calls. I like C better - the traversal subzone is utilized when calls need to traverse a firewall (when VCS Expressway is deployed).

But I failed the exam, not sure which is the correct, any thoughts?

****QUESTION 135******

Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a
destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway
Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
Answer provided: B
I selected: D
Reason: Per Cisco's MRA Deployment guide, 443 is opened from internet to DMZ only for administrative access to VCS Expressway (which is strongly discouraged). See firewall port reference on the following

guide: http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-5/Mobile-Remote-Access-via-VCS-Deployment-Guide-X8-5-2.pdf

But I failed the exam, not sure which is the correct, any thoughts?

****QUESTION 140******
Which three items must you configure to enable SAF Call Control Discovery ( choose three)

A. Calling Search Space
B. Hosted DN Groups
C. Translation Patterns
D. The SIP or H323 trunk
E. Hosted DN Patterns
F. Route Patterns

Dump answer: A / D / E
I selected below answerS:
B. Hosted DN Groups
D. The SIP or H323 trunk
E. Hosted DN Patterns

But I failed the exam, not sure which is the correctS, any thoughts?


****QUESTION 142******
A presales engineer is working on a quote for a major customer and must evaluate how many Cisco VCS Expressway traversal call licenses for which to plan. Calls to and from which three routes must the

engineer include in the tally? (Choose three.)

A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS

Dump: C E F
I selected answer : C, D and F

For a VCS Expressway, calls to or from a traversal client (Firewall traversal calls). Traversal clients include ""other VCSs(F), gatekeepers(D), Border Controllers(C), or traversal-enabled endpoints.
Reference : http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html

But I failed the exam, not sure which is the correctS, any thoughts?


****QUESTION 144******
An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes this
task?
A.Create a block route pattern.
B.Create a block learned pattern.
C.Create a block transformation pattern.
D.Create a block translation pattern.
Dump answer: C
I selected option: D Create a block translation pattern.

But I failed the exam, not sure which is the correct, any thoughts?

****QUESTION 148******
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phones
located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two.)

A.The site has exceeded the number of SRST endpoints supported by the voice gateway.
B.The ccm-manager fallback command is configured incorrectly on the voice gateway.
C.Phones at the remote site are assigned to the incorrect device pool.
D.The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.

What will be the correct answer of this.
I selected A and E.
Reason:
1. Cisco 2821 voice bundle with PVDM2-32, SRST featuring 48-phone license,
2. If BCD are correct then some phones will not work.


But I failed the exam, not sure which is the corrects, any thoughts?


****QUESTION 149******
Which function can be implemented without MTP resources? SELECT TWO
A.DTMF relay conversion
B. terminating a media stream that uses the same codec
C. Multicast music on hold
D.SIP early offer
E.IPV4 to IPV6 conversation

Dump answer: B terminating a media stream....
I selected option C "Multicast music on hold" and E "IPV4 to IPV6 conversation"


But I failed the exam, not sure which is the corrects, any thoughts?


****QUESTION 152******
Which two commands verify Cisco IP Phone registration? (Choose two.)

A.show telephony-service ephone-dn
B.show voice register session-server
C.show ephone registered
D.show ccm-manager hosts
E.show sip-ua status registrar

Dump answer: B C
But I believe that is incorrect:
A: false, this command is used to see the phone confiuration, not registration.
B: false, command do not exists.
C. Correct, for SCCP phones.
D. False, the command is used for MGCP gateway
E. Correct, used to display all SIP endpoint registered.

So I selected: C and E:

for reference:
Step 3 show sip-ua status registrar

Use this command to display all the SIP endpoints currently registered with the contact address.
From http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080557e81.html

But I failed the exam, not sure which is the corrects, any thoughts?



****QUESTION 153******
An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP address and the system name. After logging into the Cisco VCS Expressway admin page, the engineer sees this

output.
Which four options must be configured to complete the Cisco VCS Expressway system configuration? (Choose four.)
A. NTP server
B. SIP server
C. LDAP server
D. security certificate
E. DHCP server
F. DNS server
G. SIP URI
H. Cisco Unified Communications Manager IP address

If have to mark one answer, I think the right is: D - DNS Server
this questions wasnt in my exam.


****QUESTION 154******
After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the

problem?

A.The phone at the remote location is a different model than the phone in the user’s main office.
B.The user’s Extension Mobility profile is misconfigured.
C. The user can log in to only one device at a time.
D.The device pool is misconfigured.

Dump answer: A
I selected option: C.
reasons:
A: does not matter, if the phone is a different model, it will take the default profile for its type (I remember there is also another question in the dump regarding this).
B: Don't think so, because he was able to log in to another phone.
C: Yes, true, because of the "Multiple login allowed" parameter, in the extension mobility parameters
D: Inconsistent question. Device pool does not impact Extension Mobility.

For reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsgd-861-cm/fsem.html
•You can set the service parameter to allow for multiple logins. If you set multiple login not allowed, Cisco Extension Mobility supports only one login at a time for a user. Subsequent logins on other

devices will fail until the user logs out on the first device


****QUESTION 155******
Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?
A. SRST without MGCP fallback
B. SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C. SRST with MGCP fallback
D. Cisco Unified Communications Manager Express in SRST mode

DUMP ANSWER: C
I selected option: D CME in SRTS mode.
reasoning:
http://www.ciscopress.com/articles/article.asp?p=1744068&seqNum=4
CUCME in SRST Mode Usage
Examples of features that are provided only by CUCM Express in SRST mode are Call Park, Presence, Cisco Extension Mobility, and access to Cisco Unity Voice Messaging services using SCCP.

Correct Answer: None!
Reason: This one is tricky.. the closest answer is D since presence and extension mobility are both CUCME features, however while in SRST these enhanced features are not supported. I will pick D if I get

this question, but hopefully this is one of the "not graded" questions...

I selected D but I failed the exam, not sure if that is the correct one. Any thoughts?

@JCB
Holy crap I think you are right about 144. Create a block learned pattern.

@faquejai

¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.
My Answer: C

The Search rules page (Configuration > Dial plan > Search rules) is used to configure how the
Expressway routes incoming search requests to the appropriate target zones (including the Local Zone) or policy services.

@faquejai

I didn't have Q68 and Q80 during my exam


******QUESTION 98******
Which functionality does ILS use to link all hub clusters in an ILS network?
A. Fullmesh
B. Automesh
C. ILS updates
D. multicast
My Ans: A

******QUESTION 109******
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
My Answer: C

¨¨¨¨¨QUESTION 122¨¨¨¨¨¨
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)

A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
My Answer: A,C
You might need to recheck my answer on this one.


QUESTION NO: 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to
accomplish this configuration? (Choose two.)
C. Associate the directory URIs to directory numbers.
F. Assign directory URIs to users.
G. Configure the SIP profile.


QUESTION NO: 160
Which three statements about when user A calls user C using SIP are true? (Choose three.)
My Answer:
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.

But I think the correct answer is: A, if it's not vice verza.
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.

QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
B. The Tomcat certificates do not match.
D. The ILS authentication password does not match.
F. One cluster is using TLS certificate, and the other is using Password.

@Chase, good luck for your certification today I cross fingers for you!
Q144 I think is Create a block learned pattern. based on that:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011101.pdf
If you want to prevent a local Cisco Unified Communications Manager cluster from routing calls to a learned alternate number
or learned alternate number pattern, you can configure a local blocking rule on that cluster.

@Joshua:
Q115 and Q127
I think the answer for both is TRANSFORM based on that:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Basic_Configuration_Control_with_Expressway_Deployment_Guide_X7-2.pdf
The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices.
For example, if the called address is an H.323 E.164 alias “01234” the VCS will automatically append the configured domain name (in this case example.com)
to the called address (that is, 01234@example.com making it into a URI), before attempting to set up the call.

Q139: definitively is A because it is best QoS compare to C. C being the limit max from Cisco.

@faquejai:
Q98: Answer is B because "ILS uses automesh functionality to create a full mesh connection between all hub clusters within an ILS network"
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmfeat/CUCM_BK_CEF0C471_00_cucm-features-services-guide-90/CUCM_BK_CEF0C471_00_cucm-features-and-services-guide_chapter_011111.pdf

@Kiki, Sona use pdf 161q format

@all
These are the questions that made us in trouble so we have to focus on them:
Q93: Could be answer A.
Q116: Most of you answer B,C. But I don't understand how an Autogenerated devices profile can be loaded at teh same time as a user profile.
Q123: Knowing that on each test the response list change and number of response requested also.
Q125: because I am not 100% sure, and because of my % test result.
Q135:
Q136: because I am not 100% sure, and because of my % test result.
Q137
Q138: because I am not 100% sure, and because of my % test result.
Q146: Knowing that on each test the response list change and knowing that the steps to configure URI calling are:
1-Assign directory URIs to users
2-Associate the directory URIs to directory numbers.
3-Configure the SIP profile.
4-Configure SIP trunk.
5-Configure SIP route patterns
Q148:
Q160: Knowing that on each test the responses list change and number of responses requested also.

@paul
I noticed that you passed the exam yesterday with 897.
Can you please see my latest post with several questions and let us know how you answer those questions in your exam?
You score 897 which means most of those question you answered properly so we really appreciate you can tell us how to answer those questions.

Please PLEASE Please help help.

I just took the exam and failed 748
I used q161, questions are correct but answer are incorrect. They selected the questions that we have doubts, they are posting the difficult questions :-S
I tried to do a research with the official cisco documents and trying to find the correct answers and I used those answers today but I still failed the exam.
We need to work together to pass this exam.
We need assistance from people that passed the exam recently to please provide the answers they used.

PLEASE HELP in below questions:

*****QUESTION 68****
What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM endpoints? (Choose two.)
A Media Resource Group List.
B.Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
C.Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS.
D.Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM.
E.Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
Dump answer
C. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified C

I used below answers in the exam (not sure if it was good):
B. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS
E. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM

****QUESTION 80******
What is the correct value to use for the "DSCP for TelePresence Calls" Cisco CallManager service parameter?
A. 28 (011100)
B. 34 (100010)
C. 41 (101001)
D. 46 (101110)
Answer: B

Q161 indicates A
I used B 34 but not sure if it was correct.

******QUESTION 98******
Which functionality does ILS use to link all hub clusters in an ILS network?
A. Fullmesh
B. Automesh
C. ILS updates
D. multicast

DUMP ANSWER: B
What is the correct one?
EXPLAIN:
When a new hub cluster registers to another hub cluster in an existing ILS network, ILS automatically creates a full mesh connection between the new hub cluster and all the existing hub clusters in the ILS network.

Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/configure___intercluster_lookup_service.pdf
question no 160 was in the exam but you need to select only one answer.

******QUESTION 109******
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
Answer: B

161Q says the Correct answer is C but I think the Correct answer is B
I selected B but I failed the exam.

¨¨¨¨¨¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.

Dumb select option: "...The access control list on the VCS Control must..."
I saw other users selecting option D.
Traversal zone is for external call, that mean outside network. Local zone is for local network. in this case the question is for "Outside Call". I think D is correct answer.

I selected "The traversal zone on the VCS Control does not have a search rule configured" but I failed the exam.
Not sure which is the correct one.

¨¨¨¨¨¨QUESTION 122¨¨¨¨¨¨
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200

Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)

A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200

read "Set DSCP Values". so answer should be A,B.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_6/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide_appendix_01111.html

@Chase

Best of Luck to you buddy!! Keep us posted!! I am taking the exam for the 3rd time on the 29th.

@JPB made a good point on Question 139.

QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?

A. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning

Answer should be A.
Read this:

It says NO MORE. So best answer is A.
Loss should be no more than 1 percent.
One-way latency should be no more than 150 ms.

http://www.ciscopress.com/articles/article.asp?p=357102&seqNum=2

QUESTION NO: 115
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?
A.search rules
B.SIP route pattern
C.policy service
D.transform
ANSWER = A, search rules.

Q. 127 is B. transforms

Guys you are referring to the questions as if every question is at the same number. I mean every time you open the VCE player questions are mixed up. So what is the point of saying Q5 or Q89 as we don't understand to which question you are talking about.
First of all you are mentioning about the premium vce 300-075 161q file??

Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers one-by-one.

@JBG
Here is the difference between my answers and your answers

115 B
116 B, C
121 B, E
125 B,C,E
127 B
135 B
136 C
139 C
146 F, C
151 B,C,D
153 F
160 B
161 A

@Sona

I didn't use vce, used pdf instead.

@Jpb and chase
These are my answers and rest the same.
Q116: My Ans: B,C
Q122: My Ans: A,C
Q123: My Ans: B,E
Q127: My Ans: B
Q134: My Ans: D,E
Q137: My Ans: H.323 trunk , SIP trunk
Q139: My ANs: C

This seems to be a horrible test. I Gave this 3 weeks ago and failed with 810, and even today there just seems to be way too much uncertainty about certain questions and too many contradicting answers which makes it difficult to pass.

Guys i know this might be asking too much, but can one of the guys who have passed (congrats by the way) take a bit of time to helping us all who are trying to pass?

If you could post all the question numbers here with the correct answers? Or if not then at least the ones you have answered differently from the 161 dump answers or the questions that you think the answers should be different?

your help would be highly appreciated. Thanks guys.

@JPB
160 is B
161 is one answer A
125 IS B,C,E

@JPB
These are the answers I chose that differed from yours
Q115- Transform
Q116- B, C
Q122- DX650 and Jabber. SRND guide does say Jabber does pull DSCP info from CUCM
Q123- A, E
Q124- in-all, call-ended, registration
Q125- voice register pool, voice register global dn, SIP registrar
Q127- I chose Transform but I feel like I should have chosen Search Rules
Q128- SCCP
Q129- BFCP
Q132- AAR just because it asks “Call Limit”
Q135- it only asks for 1 and I chose A
Q136- C Failover
Q137- SIP and H323. Gatekeeper control uses H.323. If tests asks for H.225 I will chose that
Trunk Types in Cisco Unified Communications Manager Administration
Your choices for configuring trunks in Cisco Unified Communications Manager depend on whether the IP WAN uses gatekeepers to handle call routing. Also, the types of call-control protocols that are used in the call-processing environment determine trunk configuration options.
You can configure the trunk types in Cisco Unified Communications Manager Administration listed in this section.
• H.225 Trunk (Gatekeeper Controlled)
• Intercluster Trunk (Gatekeeper Controlled)
• Intercluster Trunk (Non-Gatekeeper Controlled)
• SIP Trunk

Q138- A, B, E
Q139- C
Q140- B, E, F
Q144- C – block transformation pattern
Q145- B AAR is routing some of the calls
Q146. I believe this one is tricky but it does say within the same cluster. I chose Sip route patterns and assign URI’s to users
Q148- A and E
Q152 – C and E
Q153- DNS
Q154 – C
Q158- A Calling party transformation pattern
Q160- I chose E (anyone that passed please correct)
Q161 – B, D, F

@JPB
I'm taking the test for the 3rd time tomorrow hopefully. These were the last scores I got
VCS Control - 63
VCS Expressway - 50
CUCM Video - 100
Centralized Call - 60
Multi-Site - 88
ILS - 67
Mobility - 83
CAC - 57

Lets try to work together and get past this horrible test

paul

Bahrain
Jun 23, 2016
Report Comment

Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers What is the exact name of the VCE File for 161Q? Thanks. What is your preferred VCE Player?

Hi,

I failed for the second time the test yesterday, with a 847 score.
I obtain the same % score for question categories 1, 3, 4 and 8, so definitively I am wrong on some answers.
Please help.

1- VCS Control: 50%
2- Collaboration Edge(VCS Expressway): 63%
3- Configure CUCM Video Service parameters: 86%
4- Describe and Implement Centralized Call Processing Redundancy: 60%
5- Describe and configure a Multi-Site Dial Plan for CUCM: 80%
6- Implement Call Control Discovery/ILS: 100%
7- Implement Video Mobility Features: 100%
8- Implement Bandwidth Management and Call Control on CUCM: 57%

Q1: A
Q7: A,C,E
Q9: A,D
Q13: D
Q14: A,C
Q15: B,D
Q16: D
Q17: C
Q22: A,C
Q34: A,D
Q35: A,C
Q48: D
Q75: B
Q84: A,C,D
Q91: A,B
Q93: B
Q95: B
Q96: B
Q103 B
Q104: C
Q115: Transform (both search rules and transform were proposed)
Q116: C,D
Q117: A,B,F
Q118: D (How many CUCM nodes can a skinny phone establish an SCCP connection to at the same time?)
Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio codec; E. T1 PRI card)
Q122: A,B
Q123: A,E (Someone could give the right answer?????)
Q124: D (A. start-call; B. end-call; C. call-started; D. registration)
Q125: A,B,C
Q126: B,C,E
Q127: search rules (both search rules and transform were proposed)
Q128: B
Q129: E
Q130: A
Q131: B
Q132: A
Q134: B,D
Q135: D
Q136: C
Q137: C,D (A was H246 Video Trunk, I should chose A because Trunk H323 is not really supported it is Trunk H225)
Q138: A,B,E (Someone could give the right answer?????)
Q139: A (appears to be best compare to C)
Q140: B,E,F
Q141: D,E
Q143: A,B,D
Q144: B
Q145: B
Q146: D,E (A. Configure SIP route patterns; B. Activate the URI service in Cisco Unified Serviceability; C. Configure SIP trunk; D. Assign directory URIs to users; E. Configure the SIP profile)
Q148: A,E
Q149: B,C (A. h.323 fast start; B. IPV6 -IPV4 transform; C. DTMF inband RTP-NTE (rfc2833); D. delayed offer h.323)
Q150: B
Q152: C,E
Q153: B (A. LDAP server; B. DNS server; C. Cisco Unified Communications Manager IP address; D. DHCP server)
Q154: C
Q155: D
Q157: C
Q158: A
Q159: B,C,D
Q160: C (Someone could give the right answer?????)
Q161: B,D,F

Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers.


QUESTION NO: 160
Which statement about when user A calls user C using SIP is true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Cisco VCS Control and Cisco VCS Expressway support static NAT.
C. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
D. RTP and RTCP ports must be opened at the firewall from internal to DMZ and vice versa

Mega Sema how you pass the exam?
which pass4 you used?

I agree with Blue. Anyone have these correct answers. I'm taking the exam tomorrow for a third time. Last score I had was 822. So close :(

How many nodes can a phone establish a connection to at the same time?
A. 4
B. 3
C. 1
D. 2
Correct Answer: D

Can someone verify this answer or explain? A phone will only register to 1 CUCM node correct?

An engineer has configured a Cisco EX60 to register with a Cisco VCS-C, but the device is not showing up as registered. During troubleshooting, which component will the engineer likely find missing in the configuration.

A. default gateway
B. MCU
C. TMS
D. DNS
E. Gatekeeper

My guess on this is TMS, Telepresence Management Server

Hi Sanjay,

Answer is A, C and E.

Reference : http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_8_9_and_X7-2.pdf

Which three statements about configuring an encrypted trunk between Cisco TelePresence Video Communication Server and Cisco Unified Communications Manager are true? (Choose three.)

A. The root CA of the VCS server certificate must be loaded in Cisco Unified Communications Manager.
B.A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP.
C. The Cisco Unified Communications Manager trunk configuration must have the destination port set to 5061.
D. A SIP trunk security profile must be configured with Device Security Mode set to TLS.
E.A SIP trunk security profile must be configured with the X.509 Subject Name from the VCS certificate.
F. The Cisco Unified Communications Manager zone configured in VCS must have SIP authentication trust mode set to On.
G. The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set to Off.

@Mega Sema: can you share with us your answers

HI Mega Sema,
have you used the same dump and its answers
as in this dump or researched your own ..answers ... many questions have incorrect answers in this dump .. but i am wondering if in exams these questions are being evaluated on incorrect answers as well .. so i want to make sure as many people complaining the same thing if you research and use correct answers you will fail and if you simply follow dump you will pass ... what you think .

I passed today. 161Q is working but need que check all the answers. In the exam for some questions the options change.
Good Look

I passed today... for some questions the answers are different, and like a lot people say the guide has a lot of answer wrong. 95% Q was from 161Q

@ Mega Sema

Can you please confirm about new questions? Were they from 161Q Dumps or out of them?

Thanks

I just failed with 847/860 I was so close. This is my 2nd Attempt, I first got 785/860.

2 new questions were:
2 Types of Trunks in CUCM
A. H.246 Trunks
B. SIP Trunks
C. H.343 Trunks
D. MGCP Trunks
E. CO Trunks
F. POTS Trunks.

I Chose A. H.246 Trunks and B. SIP Trunks but I think Answer should be SIP Trunks and CO Trunks. Any ideas?

QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password

I chose B,D,F, not sure if this was right.


QUESTION 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish this configuration? (Choose four.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.

I Chose E and F, option C was not an option in the test.


Which 2 things do not utlise MTP
A. H.323 fast start
B. IPV6 -IPV4
C. DTMF inband RTP-NTE (rfc2833)
D. delayed offer h.323
I chose A and B. I think A is wrong.

Can someone share the right answers for this?
I will later add all my answers like Steve.

Q14

Which three items must you configure to enable SAF Call Control Discovery ( choose three)

A. Calling Search Space
B. Hosted DN Groups
C. Translation Patterns
D. The SIP or H323 trunk
E. Hosted DN Patterns
F. Route Patterns

Verified Answer
B,D,E

@Mega, Congrats! Any chance you can provide the correct answers for the questions you had to correct for the latest dump? Thanks for the help.

@Mega Sema

About questions 115 e 127??
In the two questions I think the right answer is TRANSFORM.

And how about your opinion?

@Mega. Congrats!. Can you post answers to the questions you recall from the test? Specially the ones that are wrong in the recent dump? Or is you have the answers to the questions that are wrong in the dump? Thanks for the help

@ Mega Sema

Did you use the original answer of the DUMP or answer by or self? Regards

Just passed today with 916. There are few new questions, and most of the other ones were not so tricky.

Anyone recently passed 300-070?

Hi All,

I am still looking for the correct answers of the below questions, I am planning to place the exam next week..

QUESTION NO: 43
Which component is needed to set up SAF CCD?
A.SAF-enabled H.323 intercluster (gatekeeper controlled) trunk
B.SAF forwarders on Cisco routers
C.Cisco Unified Communications cluster
D.SAF-enabled H.225 trunk
------
QUESTION NO: 109
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A.G.711 requires 128K of bandwidth per call.
B.G.729 requires 24K of bandwidth per call.
C.The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D.To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and
only use G.711 between regions.
-------
QUESTION NO: 115
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?
A.search rules
B.SIP route pattern
C.policy service
D.transform
-------
QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.
-------
QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password

Please you help

Okay, took the test this morning, and the good news is that every question save on was on the 161q dump.

The bad news is that I failed with a 797!

I did not go verbatim on what the dump said as there seemed to be many wrong or questionable. Those I had researched and I answered what I found to be "correct" from the research. Looks like I should have followed the dump.

The results showed this:
VCS Control 50%
Collaboration Edge: 63%
Configure CUCM Video Parameters: 100%
Central Call Processing Redundancy: 60%
Multi-site Dial Plan: 63%
CCD/ILS: 67%
Video Mobility Features: 83%
Bandwidth Mgmt and CAC: 57%

Here's the breakdown of which questions I got and what I answered, anyone care to shed some light on which ones were the ones I got wrong?

1:A
9:A,D
13: D
14:A,C
15:B,D
16: D
17: G
22: A,C
27: A,B,E
34: A,D
35: A,C
79: B
84: A,C,D
88: A,B,C
91: A,B
93: B
95: B
96: B
103: B
104: C
109: C
114: E
115: B
116: B,C
117: A,B,F
118: D. Question actually said :How many CUCM Nodes can a skinny phone establish an SCCP connection to at the same time"
119: B
120: B
121: D,E. D was Listed as a specific type of transcoding on the IOS, A was just One Audio Codec, no B
122: A,B
123:C,E
124: F. Pick 1, had End-Call, Call-Started, Registration, one more
125: A,B,C
126: B,C,E
127: D
128: D
129: E
130: A
131: B
132: A
134: B,D
135: A
136: A
137: C,D
138: A,B,E
139: C
140: A,E,F
141: D,E
143: A,B,D
144: B
145: B
146: C,D. Choose 2, no F or H
148: A,E
149: Can't recall, Pick Two, had DTMF Relay, MOH, SIP Early Offer, h323 early offer, one more
150: B
152: B,C
153: F. Pick 1 out of 4, had NTP, LDAP, DNS and CUCM IP
154: C
155: D
156: C
157: C
158: D
159: B,C,D
160: B. Pick 1 out of 4, had A,B,C and F
161: B,D,F

What questions do you think are part of Implementing Bandwidth Management and CAC on CUCM? First time I took it I got 86% and this time I got a 57%

Which two statements about configuring mobile and remote access on Cisco TelePresence Video Communication Server Expressway are true? (Choose two.)
A. The traversal server zone on Expressway-C must have a TLS verify subject name configured.
B. The traversal client zone and the traversal server zone Media encryption mode must be set to Force encrypted.
C. The traversal client zone and the traversal server zone Media encryption mode must be set to Auto.
D. The traversal client zone on Expressway-C Media encryption mode must be set to Auto.
E. The traversal client zone and the traversal server zone must be set to SIP TLS with TLS verify mode set to On.
Correct Answer: BE ????

A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway in the DMZ to facilitate VPN-less SIP calls with users outside of the network. However, the users report that calls via the VCS are erratic and not very consistent.
What must the administrator configure on the firewall to stabilize this deployment?
A. The VCS Control should not be on the LAN, but it must be located in the DMZ with the Expressway.
B. The firewall at Company X must have all SIP ALG functions disabled.
C. The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the same network that the VCS Control is on.
D. A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway and the VCS Control servers.
Correct Answer: B????

Just got an 822 and used all the answers above. Trying to work out which ones are still not correct.

Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with vídeo endpoints registered on both systems. Users find that video endpoints registered on Call manager can call each other and likewise for the endpoints registered on the VCS server. The administrator for Company X realizes he needs a SIP trunk between the two systems for any video endpoint to call any other vídeo endpoint. Which two steps must the administrator take to add the SIP trunk? (Choose two.)
A. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server.
B. Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
C. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone.
D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.
E. Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate.
AC is right?

Which two statements about Cisco Unified Communications Manager Extension Mobility are true?
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. A device can adopt a user profile even when no user is logged in.
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time.
Correct Answer: BC
It is correct?

QUESTION 125
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.

Seeing a lot of conflicting answers from everyone. Here's what I will choose: BCE. Just refer to the SRST admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf

QUESTION 132
An engineer is performing an international multisite deployment and wants to create an effective backup
method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. CFUR
C. LRG
D. SRST

I originally thought A, but now I agree with others that C is best answer. Within context of TEHO, you should make sure LRG (Local Route Groups) are utilized as backup when WAN is down/CAC limit is reached.