Download Cisco.Test-king.300-075.v2017-09-06.by.quinn.102q.ete
Practice Exams:
Verified By Experts
Premium Bundle
$49.00

300-075 Premium ETE File


  • Premium File 393 Questions & Answers
Whats Included:
  • Latest Questions
  • 100% Accurate Answers
  • Fast Exam Updates

Download File

Cisco.Test-king.300-075.v2017-09-06.by.quinn.102q.ete
Exam: 300-075 - Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Size: 6.2 MB
Posted: Wednesday, September 6, 2017
Download: Cisco.Test-king.300-075.v2017-09-06.by.quinn.102q.ete

Rating

90.76%
28

Last Week Results!

190

Customers Passed Cisco 300-075 Exam

88%

Average Score In Real Exam At Testing Centre

83%

Questions came word for word from this dump

Run ETE Files with Vumingo Exam Testing Engine

Comments * The most recent comment are at the top

Alex M.
Mexico
Apr 10, 2018
MBIN, there are questions about VCS that not appear here
MBIN
Saudi Arabia
Apr 06, 2018
Someone attended recently ? Please update. Is this dump still valid ? Or do i have to refer additional dumps apart from this ??
mike jeff
Pakistan
Apr 04, 2018
after passing the exam, i realized how these 300-075 ccnp braindumps are good. it not only prepared me in a very short period. it updated my knowledge very well.
cisco cisco
Romania
Mar 29, 2018
is this 300-075 exam questions valid in india? i have my exam in few weeks and it’s very important
Willian
Brazil
Mar 27, 2018
Please can someone share the 161q dump?
Hermano
Brazil
Mar 22, 2018
11. -
Where in the Cisco Unified Communications Manager Administration GUI must an engineer navigate to
configure Cisco InterCluster Lookup Service authentication in communication manager?
A. Advanced Features> ILS Configuration> Roles
B. Call Routing > Intercluster Directory URI > Intercluster Directory URI configuration
C. Call Routing > Intercluster Directory URI
D. Advanced Features > ILS Configuration
Answer: D
perfection
Philippines
Mar 16, 2018
i prepared for a few days, so i didn't expect for some high result. but i scored amazing marks !!! thank you prepaway for 300-075 ete file!! i met almost all same questions on exam. 80% valid
anonymous
Italy
Mar 11, 2018
you guys saying that ur 300-075 cisco questions are great opportunity to pass the exams quickly without spending a lot of time for preparation. well, i have my exam next week and will give a try to your material ;)
qq
Netherlands
Feb 24, 2018
so amazing 300-075 practice test!!! full description of each topic allowed me to learn a lot of interesting information and update my knowledge.
Rochz
Peru
Feb 22, 2018
Passed today with 909

Some NEW QUESTIONS
1. -
Which two statements regarding the Cisco VCS search and transformation process are true? (Choose two.)
A. The Cisco VCS applies the search rules in priority order (all rules with priority 255 are processed first,
then rules with priority 254, and so on).
B. Transforms do not use priority numbers.
C. Presearch transforms are applied before call policy is configured and before user policy is applied.
D. Presearch transforms are applied after call policy is configured but before user policy is applied.
E. The Cisco VCS applies the search rules in priority order (all rules with priority 1 are processed first, then
priority 2, and so on).
F. You cannot set up search rules according to the protocol SIP or H.323.
Answer: C,E
2. -
Which feature enables users to manage business calls by using one phone number to pick up inprogress
calls on either their desk phone or their mobile phone?
A. desktop call pickup
B. send call to mobile phone
C. mobile connect
D. mobile voice access
E. access list
Answer: A
3. -
Lead to pass your exam quickly and easily. First Test, First Pass! - visit - http://www.exam.com
When configuring a secure SIP trunk, to which Cisco Unified Communications Manager trust store must
you upload the Cisco VCS certificate?
A. CallManager-trust
B. ipsec-trust
C. tomcat-trust
D. TVS-trust
Answer: A
4. -
Which DSCP value and PHB equivalent are the default for audio calls?
A. 34 and AF41
B. 32 and AF41
C. 32 and CS4
D. 46 and EF
Answer: D
5. -
For inbound calls that use SIP gateways to cisco unified communications manager, which two options are
available when configuring the format for the calling number type? (Choose
two)
Lead to pass your exam quickly and easily. First Test, First Pass! - visit - http://www.exam.com
A. Unknown
B. Subscriber
C. Long Distance
D. Default
E. SIP
Answer: B,C
6. -
In which two locations can you verify that a phone has a standby Cisco Unified communications manager?
(Choose two)
A. phone webpage
B. RTMT
C. Cisco Unified Serviceability
D. phone menu
Answer: A,D
7.-
Refer to the exhibit. An engineer is settings up a new deployment and wants to use the Cisco VCS Control
as a gateway for SIP and H.323 endpoints. Which Cisco VCS configuration step must be performed to
allow onset and offset calling?
A. Set the H.323mode On
B. Set the IP mode On.
C. Set the Gatekeeper Auto Discover mode On
D. Set the Interworking mode On.
8. -
Which two items must you configure in Cisco Unified Communications Manager to deploy Cisco SAF?
(Choose two)
A. an MWI
B. voicemail ports
C. a security profile
D. a forwarder
E. a remote destination profile
Lead to pass your exam quickly and easily. First Test, First Pass! - visit - http://www.exam.com
Answer: C,D
9. -
For which two reasons should you mark AF41 as the audio and video channels of a video call?
A. to allow high-definition quality calls over low-speed links
B. to give video calls a higher priority than AF41 in the QoS policy
C. to provide separate classes for audio-only calls and video calls
D. to prioritize video over other high-priority traffic classes
E. to preserve lip synchronization between the audio and video channels
10. -
When troubleshooting high CPU utilization within the Command Line Interface (CLI) of a Cisco Unified
Communications Manager (CUCM). Which command will show the process CPU usage for all processes?
A. show network status
B. show pert query counter Process"% CPU Time"
C. show pert query counter Partition"% Used"
D. show pert query counter Process "Process Status"
Answer: B
11. -
Where in the Cisco Unified Communications Manager Administration GUI must an engineer navigate to
configure Cisco InterCluster Lookup Service authentication in communication manager?
A. Advanced Features> ILS Configuration> Roles
B. Call Routing > Intercluster Directory URI > Intercluster Directory URI configuration
C. Call Routing > Intercluster Directory URI
D. Advanced Features > ILS Configuration
Answer: A
12. -
How is the peer address configured when ExpresswayE has only one NIC enabled and is using static NAT
mode?
A. Expressway-E DHCP
B. Cisco Unified Communications Manager DHCP
C. Expressway-E FQDN
D. Cisco Unified Communications Manager FQDN
Answer: C
Samson
United States
Feb 20, 2018
I took the test today and passed with a score of 888.

@Sarah was correct on the topics to study on to help pass the test.

Below are some of the questions I had on my test.

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.
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
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.

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.
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
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.
• B. 323 gateway
• C. SCCP gateway
• D.
• E. 225 trunk
• F. MGCP gateway
• G. SIP trunk
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)
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
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.
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.
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
When considering Cisco Unified Communications Manager failover, how many backup
servers can be configured in a Cisco Unified Communications Manager Group?
• A. 1
• B. 5
• C. 2
• D. 4
• E. 3
• F. 6
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
The VCS Expressway can be configured with security controls to safeguard external calls
and endpoints. One such option is the control of trusted endpoints via a whitelist.
Where is this option enabled?
• A. on the voice-enabled firewall at the edge of the network
• B. on the VCS under Configuration > registration > configuration
• C. on the TMS server under Registrations > whitelist
• D. on the VCS under System > configuration > Registrations
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.
A local gateway is registered to Cisco TelePresence Video Communication Server with a
prefix of 7. The administrator wants to stop calls from outside the organization being routed
through it. Which CPL configuration accomplishes this goal?
A)


B)


C)


D)


E)


• A. Exhibit A
• B. Exhibit B
• C. Exhibit C
• D. Exhibit D
• E. Exhibit E
What is the default DSCP/PHB for video conferencing packets in Cisco Unified
Communications Manager?
• A. EF/46
• B. CS6/48
• C. AF41/34
• D. CS3/24
In Cisco Unified Communications Manager, where do you configure the +E.164 number
that is advertised globally via ILS?
• A. ILS configuration under Advanced Features
• B. +E.164 alternate number under Directory Number Settings
• C. Device Information under Phone Configuration
• D. Route Pattern under Route/Hunt
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.
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
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
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.
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
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
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.
Sarah
United States
Feb 20, 2018
My advice, study SRNDs and VCS docs thoroughly. There were Q's about tftp options, dns options, sccp phone registrations/sec on fail-over and QOS mappings.
Chicago
South Africa
Feb 20, 2018
Well done @Sarah
can you please share more information on the dumps you used
Rex
Colombia
Feb 17, 2018
giving up to search for 300-075 cisco ccnp pdf and will ask for some help, please
Sarah
United States
Feb 16, 2018
Passed today with 895 :)
Chamba
El Salvador
Feb 12, 2018
Hi!,
Does Anybody has passed the exam recently!? what is the most accurate dump to download?
Rob
South Africa
Jan 27, 2018
Hi Everyone,

Anyone been for the exam of recent? What are valid currently?
Rochz
Peru
Jan 18, 2018
Hi
What is the valid dump now?
CAB
New Zealand
Jan 13, 2018
Thanks all for sharing.
Just wondering how you download or access the 161q dump file.
CAB
New Zealand
Jan 12, 2018
Thanks all for sharing.
Just wondering how you download or access the 160q dump file. The comments provided are all on the Quinn.102q dump file. Any assistance will be great.
Sarah
United States
Jan 11, 2018
could some one pls email me the 161q dump to
welleby@gmail.com.
Sarah
United States
Jan 08, 2018
Hi All,
What is the latest dump available?
fadi
United Arab Emirates
Jan 07, 2018
kindly, anybody send me dumps
@Hari could you share the dump to me ftahhan@hotmail.com
BW_Student
Australia
Jan 04, 2018
Hi

Please can someone share the 161q dump? bwdstudent@gmail.com

Thanks
H
Canada
Jan 02, 2018
Anyone has the latest dump?
Zulu
United States
Dec 12, 2017
The dump 161q is still valid. Passed today.
Andy
Ukraine
Dec 11, 2017
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
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/
greg10
Mexico
Dec 11, 2017
Hi Kazee, look top of this page: 300-075 dump ETE File...and Download
Ymy
Turkey
Dec 11, 2017
Hi Friends;

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

Many Thanks
Syed
United States
Dec 11, 2017
Planning to give CIPv2 exam, can anyone let me know where can I get the 161q dumps?
Shachar
Israel
Dec 11, 2017
Passed yesterday with a score 897. All questions from 161q, refer to the comments in this forum.
thanks for everyone and good luck!
Hari
India
Dec 10, 2017
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.
Bill
United States
Dec 10, 2017
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.
John
India
Dec 10, 2017
@ Sham1629

Congratulations. If you remember any questions, Please share the questions.
Sham1629
United States
Dec 10, 2017
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.
Hari
India
Dec 10, 2017
@ Joseph

It is about 161q dump
Joseph
France
Dec 10, 2017
Please can you share with us the last questions ? Which ETE file are yoy using to check the question number and answers ?

Thanks
sherazkk
Netherlands
Dec 09, 2017
@Hari.

Sorry. Yes I got this question in exam and my answer was A & D.
Sherazkk
Netherlands
Dec 09, 2017
@Hari

No.
TS
Singapore
Dec 09, 2017
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!
MANICSHEEP
South Africa
Dec 09, 2017
Join our COLLABORATION
preet
United States
Dec 09, 2017
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!
sathish778
Singapore
Dec 09, 2017
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.
sham1629
United States
Dec 08, 2017
Guys I am really sorry but unfortunately I dont remember those new questions...
sathish778
India
Dec 08, 2017
@Shem,

Planned to take exam tomorrow morning, please share the new question.
Sathish
India
Dec 08, 2017
@sham1629

If you remember questions please share, planned to take exam tomorrow
nickfutbol
Russian Federation
Dec 08, 2017
??
George
India
Dec 08, 2017
@ 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.
Mariusz
Poland
Dec 08, 2017
@sham1629

What kind of new questions have you seen?
Could you describe them ?
Hari
India
Dec 07, 2017
@ sherazkk

You got Q147 in your exam?
Sherazkk
Pakistan
Dec 07, 2017
@Hari:

According to me the answer is
A&D
sham1629
United States
Dec 07, 2017
Attempted exam but failed yesterday...
Almost Question are from 161Q but 8-10 new questions though.
saintcisco
Turkey
Dec 07, 2017
I passed today with 869 i use 161Q and @Ramesh comment. 161Q is valid but answers not valid. Be careful. Thanks again @Ramesh
Hari
India
Dec 07, 2017
@ 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.
corinth
Slovakia
Dec 07, 2017
@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)
Ferock_mh
Ecuador
Dec 06, 2017
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
Kane
Canada
Dec 06, 2017
Friends,
on which file or document should I start as the baseline? Is this 102q ete 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 ete should I start.

Thanks!
preet
United States
Dec 06, 2017
@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.
Louis
Hong Kong
Dec 06, 2017
Hello i will take the exam next week, any hints which file can I study? All the best. Thank you.
Anco
Netherlands
Dec 06, 2017
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.
preet
United States
Dec 06, 2017
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).
Canada
Canada
Dec 05, 2017
Thanks Everyone and prepaway. I passed my exam Today with 896.
Star
Canada
Dec 05, 2017
How many questions are in the dump file. I've 102 questions in total. Can someone help me.
Juan
Spain
Dec 05, 2017
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?
PASS
United Kingdom
Dec 05, 2017
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
sherazkk
Pakistan
Dec 05, 2017
I passed the exam in first attempt with 906 marks.
Special thanks to @anon12345, @corinth, @Hi-Octane and @suman for their guidance.
JB
United States
Dec 05, 2017
@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.
steve
United States
Dec 04, 2017
corinth,
What was you CAC questions if u remeber?? everyone i see get low marks on CAC
Hi-Octane
Australia
Dec 04, 2017
Just passed the exam with 910 and got lucky 4th time.

Thanks all for your support.
Much appreciated
Ron
Germany
Dec 04, 2017
1.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 am also thinking that answer A and C

Interworking On and Registered only gives ability to interconnect SIP H323

In case when we select H323, what it gives according to SIP? Please correct this or may be some documentation

2. Can anybody explain this question what do you think, A or D?
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

3. In exam which I failed week ago I had qestion

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

But in answers I remember BE6000 was there

and two another answers but also there CME in SRST mode

a litl bit confusing if we will have BE6000 in the branch it will be also mobility and presence

did anybody saw this kind of answers in this question?

4. 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

can any body also share your opinion? many different answers

5. 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?

Everybody says SCCP, but I saw in docs
________________________________________
Restrictions for SCCP Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways

Directed call park and directed call pickup are not supported for Cisco Unified Communications Manager
_______________________________________

MGCP-controlled gateways do not require a media termination point (MTP) to enable supplementary services such as hold, transfer, call pickup, and call park.
________________________________________

any opinion?

6. To how many nodes 'SCCP' phones can connects 1 or 2???

7. I dont remember exact question, If we have on ExpresswayE one NIC what we have to set
FQDN or IP?
Juan
France
Dec 04, 2017
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?
Dave
United States
Dec 04, 2017
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
Dave
United States
Dec 04, 2017
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.
corinth
Slovakia
Dec 03, 2017
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 :-)
CCNP
United Kingdom
Dec 03, 2017
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?
Ramesh
Hong Kong
Dec 03, 2017
Sorry guys forgot to mention the corrections i made were from the prepaway dump 161 dumps where all questions are valid.

Good luck.
suman
Canada
Dec 03, 2017
@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
Ramesh
Hong Kong
Dec 03, 2017
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.
Bill
United States
Dec 03, 2017
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!
calvin
United States
Dec 02, 2017
@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.
CCNP
Japan
Dec 02, 2017
Hello! Please update who recently passed or failed exam in past few days
I failed exam last week only few points
Juan
Spain
Dec 02, 2017
Hi.

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

Thank you very much!
Juan
Spain
Dec 02, 2017
Hi.
Please, could you share the following questions?

q133, q142, q147 and q156

thank you very much
suman
Canada
Dec 02, 2017
passed in second try with 869. Thank u all for your inputs
CCNPCollab
United States
Dec 01, 2017
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
Guillaume
France
Dec 01, 2017
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.
Hi_Octane
Australia
Dec 01, 2017
@Guillaume

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

Thanks!
Ramesh
Hong Kong
Dec 01, 2017
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

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?
Guillaume
France
Dec 01, 2017
@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.
Hi_Octane
Australia
Dec 01, 2017
@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
Adam
United States
Nov 30, 2017
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
Scotty
United Kingdom
Nov 30, 2017
Exam booked for Monday. Here goes. Will be studying all weekend.
Guillaume
United Kingdom
Nov 30, 2017
Hi all,

Just passed this crap with 906.
All questions except the 2 already mentionned are in the 161 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 !
Mariusz
Poland
Nov 30, 2017
@ MERCE

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

Regards.
Adam
United States
Nov 30, 2017
@Larry

Much appreciated
ja
Czech Republic
Nov 30, 2017
Merce

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

pero no estoy seguro de que es la respuesta correcta
ja
Czech Republic
Nov 30, 2017
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.
MERCE
Spain
Nov 29, 2017
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
ja
Czech Republic
Nov 29, 2017
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_Octane
Australia
Nov 29, 2017
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.
larry
United States
Nov 29, 2017
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.
Calvin
United States
Nov 29, 2017
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 (300-075) 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.
Kiki
Greece
Nov 28, 2017
@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.
Orddie
United States
Nov 28, 2017
failed today even though i reviewed most answer below and corrected it in my file.
calvin
United States
Nov 28, 2017
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, 300-075. 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 300-075 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.
plum
United States
Nov 28, 2017
@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
Adam
United States
Nov 28, 2017
@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
Adam
United States
Nov 28, 2017
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.
CCNPCollab
India
Nov 28, 2017
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.
CCNPCollab
India
Nov 27, 2017
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.
CCNPCollab
India
Nov 27, 2017
@Kiki

Thank you.
Mohan
India
Nov 27, 2017
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
Mohan
India
Nov 27, 2017
@Juan 102q is not valid
Please Check 161q Dumps
calvin
United States
Nov 27, 2017
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.
Kiki
Greece
Nov 27, 2017
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
Juan
France
Nov 26, 2017
Could you confirm if dump 102q is valid?
Kiki
Greece
Nov 26, 2017
@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...
KMX
Germany
Nov 26, 2017
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.
Scotty
United Kingdom
Nov 26, 2017
Ignore last comment. Was meant to be posted in 300-210. Sorry
Scotty
United Kingdom
Nov 26, 2017
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.)
CCNPCollab
United States
Nov 25, 2017
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.
D4D
Philippines
Nov 25, 2017
Passed today with 897. Same questions in 161 just be careful on your answers.
CallGurl
United States
Nov 25, 2017
@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
Kiki
Greece
Nov 25, 2017
@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
D4D
Philippines
Nov 25, 2017
Placing my exam tomorrow. Hope the questions in 161 still the same.
Adam
United States
Nov 25, 2017
@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.
Steve
United States
Nov 24, 2017
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.
Nono
Indonesia
Nov 24, 2017
@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.
Scotty
United Kingdom
Nov 24, 2017
@r.zelig

file has all the questions. See what people have posted in here as to what the correct answers should be.
r.zelig
Israel
Nov 24, 2017
eny bady pass the sxem with the primum file ?160q
Hari
India
Nov 24, 2017
@ 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
Adam
United States
Nov 24, 2017
@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
Scotty
United Kingdom
Nov 24, 2017
@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.
Scotty
United Kingdom
Nov 23, 2017
@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.
Ramesh
Hong Kong
Nov 23, 2017
@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
tom
United States
Nov 23, 2017
guys, where can I get the 161 dump? It doesn't seem to be listed
Nono
Indonesia
Nov 23, 2017
@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
D4D
Philippines
Nov 23, 2017
___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!
Hari
India
Nov 23, 2017
@ 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.
biskut
United States
Nov 22, 2017
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?
Chase
United States
Nov 22, 2017
@Hari
Q120: D
Q123: A and E
Q135: D
Q138: A B E
Q160: A
my_time
United States
Nov 22, 2017
I see that folks are discussing questions numbered near 160, but this file says it has only 102 questions . Are all of you discussing the same file? If not, where do I get the file you are discussing?
Hari
India
Nov 22, 2017
@ 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.
Steve
United States
Nov 22, 2017
D'oh! Looks like I'm talking to myself!

I meant to say @Chase!
Hari
India
Nov 21, 2017
@ Scotty and Nono,

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

@ Scotty...Q95 : Thanks for giving the answer on this.
Steve
United States
Nov 21, 2017
@ 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!
Nono
Indonesia
Nov 21, 2017
@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).
Scotty
United Kingdom
Nov 21, 2017
@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.
Scotty
United Kingdom
Nov 21, 2017
@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
Joshua
United States
Nov 21, 2017
I passed the exam after the 3rd attempt. with 888.
Hari
India
Nov 20, 2017
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
Chase
United States
Nov 20, 2017
Just passed with a 906. Was my 3rd attempt. My CCNP expired Saturday, sure hope there was a grace period.
bari
Indonesia
Nov 20, 2017
nono
Which file did you used ? dump ?
Nono
Indonesia
Nov 20, 2017
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
Hari
India
Nov 20, 2017
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.
Scotty
United Kingdom
Nov 20, 2017
@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.
CCNPCollab
United States
Nov 19, 2017
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.
Scotty
United Kingdom
Nov 19, 2017
@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.
Ramesh
Hong Kong
Nov 19, 2017
@ 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 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
CCNPCollab
United States
Nov 19, 2017
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.
Toni Ruiz
Spain
Nov 19, 2017
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?
Kiki
Greece
Nov 19, 2017
@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??
Blue
Canada
Nov 18, 2017
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
Bill
United States
Nov 18, 2017
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.
Anon12345
United States
Nov 18, 2017
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
anon12345
United States
Nov 18, 2017
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/2017/11/16/cipt2-300-075-exam-helpful-information/
Chase
United States
Nov 18, 2017
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
Faquejai
United States
Nov 18, 2017
@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?
Chase
United States
Nov 17, 2017
@JCB
Holy crap I think you are right about 144. Create a block learned pattern.
paul
Bahrain
Nov 17, 2017
@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.
paul
Bahrain
Nov 17, 2017
@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.
JPB
United States
Nov 17, 2017
@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.
Faquejai
United States
Nov 17, 2017
@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.
faquejai
United States
Nov 17, 2017
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
Adam
United States
Nov 16, 2017
@Chase

Best of Luck to you buddy!! Keep us posted!! I am taking the exam for the 3rd time on the 29th.
Joshua
United States
Nov 16, 2017
@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
Kiki
Greece
Nov 16, 2017
Guys you are referring to the questions as if every question is at the same number. I mean every time you open the ETE 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 dump ete 300-075 161q file??
DCY
Brazil
Nov 16, 2017
Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers one-by-one.
Blue
Canada
Nov 16, 2017
@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
paul
Bahrain
Nov 16, 2017
@Sona

I didn't use ete, 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
Ramesh
Hong Kong
Nov 15, 2017
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.
Blue
Canada
Nov 15, 2017
@JPB
160 is B
161 is one answer A
125 IS B,C,E
Chase
United States
Nov 15, 2017
@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
chase
United States
Nov 15, 2017
@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
Sona
Australia
Nov 15, 2017
paul

Bahrain
Nov 13, 2017
Report Comment

Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers What is the exact name of the ETE File for 161Q? Thanks. What is your preferred ETE Player?
JPB
United States
Nov 15, 2017
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
paul
Bahrain
Nov 14, 2017
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
Faquejai
United States
Nov 14, 2017
Mega Sema how you pass the exam?
which pass4 you used?
Chase
United States
Nov 14, 2017
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 :(
test
United States
Nov 14, 2017
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?
Bill
United States
Nov 14, 2017
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
John
India
Nov 14, 2017
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.
Y@Y@
Tunisia
Nov 13, 2017
@Mega Sema: can you share with us your answers
Sam
United States
Nov 13, 2017
HI Mega Sema,
have you used the same dump and its answers
as in dump or researched your own ..answers ... many questions have incorrect answers in 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 .
David
Mexico
Nov 13, 2017
I passed today. 161Q is working but need que check all the answers. In the exam for some questions the options change.
Good Look
David
Mexico
Nov 13, 2017
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
Hi_Octane
Australia
Nov 13, 2017
@ Mega Sema

Can you please confirm about new questions? Were they from 161Q Dumps or out of them?

Thanks
Joshua
United States
Nov 12, 2017
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.
Bill
United States
Nov 12, 2017
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
Rah
United States
Nov 12, 2017
@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.
DCY
Brazil
Nov 12, 2017
@Mega Sema

About questions 115 e 127??
In the two questions I think the right answer is TRANSFORM.

And how about your opinion?
Rah
United States
Nov 12, 2017
@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
DCY
Brazil
Nov 12, 2017
@ Mega Sema

Did you use the original answer of the DUMP or answer by or self? Regards
Mega Sema
Czech Republic
Nov 12, 2017
Just passed today with 916. There are few new questions, and most of the other ones were not so tricky.
Thogadia
India
Nov 11, 2017
Anyone recently passed 300-070?
blue
Canada
Nov 11, 2017
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
Steve
United States
Nov 11, 2017
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
Chase
United States
Nov 11, 2017
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%
DCY
Brazil
Nov 11, 2017
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 ????
DCY
Brazil
Nov 11, 2017
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????
Chase
United States
Nov 10, 2017
Just got an 822 and used all the answers above. Trying to work out which ones are still not correct.
DCY
Brazil
Nov 10, 2017
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?
DCY
Brazil
Nov 10, 2017
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?
Jase
United States
Nov 10, 2017
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.
Jase
United States
Nov 10, 2017
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.


E is definite. However I feel like no other answer applies. I suppose A would be the next best answer but here are my thoughts:

A. In my opinion if the EX60 didn't have a default gateway, then I would think the call would never have been setup in the first place, hence we wouldn't even be talking about one way audio.
B. The port direction they are describing is backwards, otherwise it would be right.
C. I don't see what the TMS has to do with one way audio.
D. If there was no response to SIP INVITE then the call wouldn't be setup, hence no one way audio.
DCY
Brazil
Nov 10, 2017
No, it is automesh!

Explain:
ILS uses automesh functionality to create a full mesh connection between all hub clusters within an ILS network.
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.
DCY
Brazil
Nov 09, 2017
@SP @Everybody
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
I THINK IS A.

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
Sanjay
United States
Nov 09, 2017
Could someone answer the below please?

What are two important considerations when implementing TEHO to reduce long-distance cost?
(Choose two.)
A.on-net calling patterns
B.E911 calling
C.number of route patterns
D.caller ID

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.
SP
United States
Nov 09, 2017
@King
Thank you

@ Akira
Q153
DNS
Server Certificate (Under security certificate)
NTP Server
SIP Domains
So, DNS is first choice

Q127
Search rules - @domain
Transforms - @example.com
I concur transforms as right answer

Q111
B. the destination alias, without the domain portion

Both Q127 & Q111: http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Single-VCS-Control-Deployment-Guide-X8-2.pdf

Still searching for Q160 and Q123
DCY
Brazil
Nov 09, 2017
Which statement about configuring the Cisco VCS Control and Cisco VCS Expressway is true?
A. You do not need to configure search rules for traversal calls.
B. You need to configure the firewall to allow communication from the Cisco VCS Expressway to the Cisco VCS Control.
C. The username on the Cisco VCS Control and Cisco VCS Expressway are local and do not need to match.
D. The Cisco VCS Expressway is the Traversal Server.
D IS RIGHT??
DCY
Brazil
Nov 09, 2017
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.
Is D??
DCY
Brazil
Nov 08, 2017
If you want to delete a SAF-enabled trunk from Cisco Unified Communications Manager Administration, what must you do first?
A. Disassociate the trunk from the CCD advertising service or CCD requesting service.
B. Delete the trunk from the CCD requesting service node.
C. Place the Cisco Unified Communications Manager node in standby mode.
D. Redirect CCD advertising and requesting services to another Cisco Unified Communications Manager.
Correct Answer: A
DCY
Brazil
Nov 08, 2017
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.

Is B??
DCY
Brazil
Nov 08, 2017
When a call is made from a video endpoint to a Cisco TelePresence EX90 that is registered to a Cisco VCS Control, which portion of the destination URI is the is the first match that is attempted?
A. the full URI, including the domain portion
B. the destination alias, without the domain portion
C. the E.164 number that is assigned to the Cisco TelePresence EX90
D. the directory number that is assigned to the Cisco TelePresence EX90

Answer: B is right??
Akira
Philippines
Nov 08, 2017
@King Thank you very much
one last set, @king @fen @Sunday if you can share your thoughts and comments. i will be taking the exam next week.

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

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

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

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

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

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 C


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: D

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
king
Philippines
Nov 08, 2017
@Akira

QUESTION NO: 160
-you choose 4 answers or 4 answers but you need to choose 3?
There will be 4 options only and need to choose 1.

QUESTION NO: 153
-NTP Server, SIP Server, Security Cert and DNS. will NTP be the best answer?
I think the correct answer is DNS Server in Exam

QUESTION NO: 136
-Should be the failover?
Yes Failover

QUESTION NO: 127
-my guess is search rule?
-No I think Transforms

QUESTION NO: 123
-Can you explain your answer to this.
-As I said Can't remember but the answer might be IOS router.
king
Philippines
Nov 08, 2017
@SP
Correct answers of 161 are
The Tomcat certificates do not match.
One cluster is using TLS certificate, and the other is using Password.
The ILS authentication password does not match.
Sathish
India
Nov 08, 2017
Is dump Valid. Is anyone took exam with dump
SP
United States
Nov 07, 2017
@ King and Sunday
What is the correct answer for question number 161?
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.
Dump said C, D, F
Q 160 Which three statements about when user A calls user using SIP are true? (Choose three.) – They ask only one answer
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
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

Ask for only one answer. I choose E but I think A is better choice.
DCY
Brazil
Nov 07, 2017
@John

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.

If only requires 2 answer, I think: FC

Step 1
• Assign Directory URI to Users / Associate Directory URI with Directory Numbers

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
Sanjay
United States
Nov 07, 2017
Could someone answer the below please?

What are two important considerations when implementing TEHO to reduce long-distance cost?
(Choose two.)
A.on-net calling patterns
B.E911 calling
C.number of route patterns
D.caller ID

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.
DCY
Brazil
Nov 07, 2017
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

Summary of Process The configuration process consists of the following tasks.
VCS system configuration:
■ Task 1: Performing Initial Configuration, page 7
■ Task 2: Setting the System Name, page 7
■ Task 3: Configuring DNS, page 8
■ Task 4: Replacing the Default Server Certificate, page 10
■ Task 5: Configuring NTP Servers, page 11
■ Task 6: Configuring SIP Domains, page 11
SP
South Korea
Nov 07, 2017
first test 673 and second test 855. both failed but gettinh better
Adam
United States
Nov 07, 2017
@John

I have answers to a few of your questions that you posted. Perhaps we can collaborate on these questions so we can knock this out with a passing score!


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
Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-series-gateways/product_data_sheet09186a00801d87f6.html

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 hear 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.
my answer: A,E


reasoning:
A: TRUE - the default gateway is not configured.
B: FALSE - The port direction is reversed! if this was accurate then C could hear A&B, not the other way around.
C: FALSE - cannot find evidence to support this answer
D: FALSE - it is responding because the call is being setup
E: TRUE - paths need to be established in both directions

check this document, search for default gateway
http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.html


Can you tell me what you selected for these if these were on your exam? I am sitting for the exam next week.

-----------------------------------------

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.


Correct answer

A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS


Dump answer
B. 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


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



Dump answer is B

Correct answer is E


BFCP allows users to share presentations/desktops within an ongoing video conversation. Desktop sharing video stream will be running as additional one to the actual call which already has audio and video streams


Which code snippet is required for SAF to be initialized?
A.Service Family
B.External-Client
C.router eigrp
D.topology base

C.Router eigrp ??



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.

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.

I DO NOT believe these are the correct answers for this one, as HSRP is a router command.


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 think 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.




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#JABW_TK_S11EF173_00


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


How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2




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.

Dump has D as the answer


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: D,E,F
Explanation:
http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.htm


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


Exam asks for 1 answer




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

Your answer B. The location Hub_None has not been activated
Correct answer A. The DX650's MAC address is incorrect in the Cisco UCM


Reasoning:
In the tab DX650 Configuration it shows the MAC address entered " D0C78914131D "


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. SRST
B. CFUR
C. LRQ
D. AAR

Dump answer: B

Correct answer: D

SRST is for call control survivability

LRQ is a H323 location request message

CFUR is Call Forwarding Un Registered

AAR is Automatic Alternate Routing used for PSTN routing in event of inadequate bandwidth

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b03aar.html


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: A


The correct answer is C. here my 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.


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 voice register global dn
D. Configure telephony service
E. Configure the SIP registrar
F. Configure voice register pool

Dump answer
B. Configure an SRST reference
C. Configure voice register global dn
F. Configure voice register pool

Correct answer
B. Configure an SRST reference
C. Configure voice register global dn
E. Configure the SIP registrar


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

my answer: C
reasoning:
A: FALSE - AAR is to contact a registered device on the CUCM via an alternative method.
B: FALSE - Call Forward unregistered - provides backup dialling to device when unregistered to CUCM
C: TRUE - Local Route Groups - TEHO is to contact a local number at a distant location.
D: FALSE - Survivable Remote Site Telephony - invoked at remote site if the link is lost from the CUCM

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

my answer: C
reasoning:
A: SAF must not be on gatekeeper controlled trunks.
B: SAF is Cisco Proprietary so must be on cisco routers.
C: SAF CCD - With the call control discovery feature, each local Cisco Unified Communications Manager cluster can perform the following tasks:
D: H.225 trunk is a gatekeeper controlled trunk, SAF must not be used on gatekeepers
Chase
United States
Nov 06, 2017
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

Pretty sure this one is D. MGCP. Only MGCP and H323 Gateways. SCCP is an application you put on a H323 or MGCP gateway
Akira
Philippines
Nov 06, 2017
@King

Please hoping for your kind assistance.

QUESTION NO: 160
-you choose 4 answers or 4 answers but you need to choose 3?

QUESTION NO: 153
-NTP Server, SIP Server, Security Cert and DNS. will NTP be the best answer?

QUESTION NO: 136
-Should be the failover?

QUESTION NO: 127
-my guess is search rule?

QUESTION NO: 123
-Can you explain your answer to this.
john
United States
Nov 06, 2017
failed 822/860.

Would someone point me to the right answer for below questions?

QUESTION 10
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.

dump ask for 1 answer only. Which is the answer? i selected B in the exam

QUESTION 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

What is the answeR? is it B. H323?

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

WHat is the answer? I selected SCCP Gateway in the exam, is the answer D. MGCP Gateway?

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.

Dump has 4 answers: AEFH. Exam only requires 2 answers. I selected A,F. What is the correct answer?
King
Philippines
Nov 06, 2017
@Adam,
QUESTION NO: 160
Question and exhibit are same but answers are different. For My case this question had 4 answers which were A,C,D,E .
QUESTION NO: 153
Asks for only one answer and the answers will have only one correct answer.
QUESTION NO: 136
Careful about the answer. Defiantly Q161 answer is wrong.
QUESTION NO: 127
Careful about the answer. Defiantly Q161 answer is wrong.
QUESTION NO: 123
Had different answer than Dumps
I felt like a modified question of QUESTION NO: 43, related to SAF. I can't remember answers are but I gave my answer 'something..Router'.
I took two time this exam and never seen a single questions in between dumps Question Number 57-68.So don't worry about these questions
Adam
United States
Nov 06, 2017
@King

Congrats on passing, Can you lend any insight to questions that have been posted here that you may answered differently than what dumps are providing?
june
Brunei
Nov 05, 2017
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

DUMP Answer is :BCFH

My guess answer = NTP Server, DNS Server, Security Certificate, i cant figure the other one out. Either its SIP server or SIP URI?

http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-7/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-7.pdf
Under Summary Process u can see VCS System configuration:- Task 1 to Task 6.
Task 6 = Sip Domain, so would my 4th answer be SIP Server or SIP URI?

Anyone can help solve this question?
Ham
Australia
Nov 05, 2017
Can someone please answer this?

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.

Dumps say correct answer is D and F.

D is the confirmed. Confused about second option. is it E or F ?
King
Philippines
Nov 05, 2017
Passed today with a decent 900. Q161 is still valid but need to study all questions and find out answers. Thanks everyone.
WA
Saudi Arabia
Nov 05, 2017
@ Sunday

congrats!
Appreciate if you can share the list of questions with their correct answers with us.

thank you!
John
India
Nov 05, 2017
Hi Arsbo,

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 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
Adam
United States
Nov 04, 2017
@Sunday

Can you verify these questions and or can you provide the list of questions you modified for those of us getting ready to sit the exam

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.

Correct answer
A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS

Dump answer
B. 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

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

Exam asks for 1 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. H.264
C. H.224
D H.263
E. BFCP
Dump answer is B
Correct answer is E

BFCP allows users to share presentations/desktops within an ongoing video conversation. Desktop sharing video stream will be running as additional one to the actual call which already has audio and video streams

BFCP Endpoints

BFCP is supported by default on the following endpoints:

Cisco E20, Cisco TelePresence Codec C40, Cisco TelePresence Codec C60, Cisco TelePresence Codec C90, Cisco TelePresence EX60, Cisco TelePresence EX90, Cisco TelePresence Quick Set C20, Cisco TelePresence Profile 42 (C20), Cisco TelePresence Profile 42 (C60), Cisco TelePresence Profile 52 (C40), Cisco TelePresence Profile 52 Dual (C60), Cisco TelePresence Profile 65 (C60), Cisco TelePresence Profile 65 Dual (C90), Cisco TelePresence, Cisco TelePresence 1000, Cisco TelePresence 1100, Cisco TelePresence 1300-47, Cisco TelePresence 1300-65, Cisco TelePresence 1310-65, Cisco TelePresence 3000, Cisco TelePresence 3200, Cisco TelePresence 500-32, Cisco TelePresence 500-37, CSF

Which code snippet is required for SAF to be initialized?
A.Service Family
B.External-Client
C.router eigrp
D.topology base

C.Router eigrp

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.

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.

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 think 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.

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
Tang
Hong Kong
Nov 04, 2017
Finally tamed this beast in a second try. Got 58 questions, 4 were new questions which have already been mentioned on this site, 4 to 5 questions had reworded answers and 4-5 questions had single answer to select vs multiple in the dumps, rest all from the dump but recheck your answers as most of the questions have wrong answers.
Based on the past pattern I think cisco will change this exam in one week to three weeks. best of luck to everyone
arsbo
Brazil
Nov 04, 2017
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: D,E,F
Explanation:
http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html
Adam
United States
Nov 04, 2017
@Chase Collins @Sunday

I have a list of questions, that I believe are incorrect after failing test. Does anyone have the correct answers or has sat for the exam recently to give some info.


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.


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

Exam asks for 1 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. H.264

C. H.224

D H.263

E. BFCP



Which code snippet is required for SAF to be initialized?
A.Service Family
B.External-Client
C.router eigrp
D.topology base


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.





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.












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








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



How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2







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.




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.


Exam asks for 1 answer








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







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






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




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






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


Exam asks for 1 answer
Sunday
Italy
Nov 04, 2017
Hi guys, exam passed. Dump is valid, find tech reference for every question because on the dump I corrected about 30 questions before taking the exam.
Good luck everyone!
Thairo
Thailand
Nov 04, 2017
@Sunday

Thanks for clarification. I will try to exam shortly...
OrangeBoy
Thailand
Nov 03, 2017
Is there anyone took exam(300-075) lately? Still valid 161 questions? I'm plan to take exam on this Friday. Wish me a good luck. Thanks.
King
Philippines
Nov 03, 2017
Hi Guys

What is the correct answer for this
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.

161Q says the Correct answer is C but I think the Correct answer is B. Please let me know your thought.

Thanks
arsbo
Brazil
Nov 03, 2017
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: C
Reference:
http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book/QoSIntro.pdf - Pág, 16
arsbo
Brazil
Nov 03, 2017
QUESTION NO: 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
Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-series-gateways/product_data_sheet09186a00801d87f6.html - Table 1
arsbo
Brazil
Nov 03, 2017
QUESTION NO: 124
Which three messages does a Cisco VCS use to monitor the Presence status of endpoints?
(Choose three.)

A. start-call
B. in-all
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_presence_status_endpoints_kb_186.html
arsbo
Brazil
Nov 03, 2017
QUESTION NO: 107
Which module is the minimum PVDM3 module needed to support video transcoding?
A. PVDM3-32
B. PVDM3-64
C. PVDM3-128
D. PVDM3-192

Answer: C
Reference:http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/15_1/vb_15_1_Book/vb-video-transcoding.html
Chase Collins
United States
Nov 02, 2017
@Sunday by chance are you finished with that guide?
Adam
United States
Nov 02, 2017
@Sunday

Do you have a list of finalized questions? I have a list here, but do not want to keep posting the same info. Please advise..
John
United States
Nov 02, 2017
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.

I think the answer should be ACD. any idea?
binoy
United Arab Emirates
Nov 02, 2017
CIPTV2 (300-075) is still valid ?
IAH
Spain
Nov 02, 2017
Which is the .ete valid?
Sunday
Italy
Nov 02, 2017
@Tairo
I do not agree.
This is the definition of SIP Route Pattern: Cisco Unified Communications Manager uses SIP route patterns to route or block both internal and external calls.

This is the definition of Transform:

The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices. The following transform modifies the destination alias of all call attempts made to destination aliases which do not contain an ‘@’. The old destination alias has @example.com appended to it. This has the effect of standardizing all called destination aliases into a SIP URI format.

I will select "Transform", in case I hit this question.

I will soon try the exam, I'll let you know.

Hope this helps!
Thairo
Thailand
Nov 02, 2017
@Sam, @John and @ Sunday:

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

My instructor told me answer is
"B.SIP route pattern"

Is anyone can confirm with this?
Sunday
Italy
Nov 01, 2017
@Sam and @John:

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

I guess the correct answer is "Transform".
It's like a translation pattern, which manipulates numbers.
A Sip Route Pattern just tells where the call will be sent out from CUCM.
The question is asking for "h323 to sip video capabilities". It could also be between 2 endpoint both registered to VCS. Transform manipulates digits and may strip/add a domain or translate a pattern.

From VCS and CUCM Deployment guide:
Thus, a transform is needed to ensure that the dialed number is transformed into a consistent form, in this case to add the domain (vcs.domain) if required.

Hope this helps!
NemoJP
France
Nov 01, 2017
@John, the answer is
C. The user can log in to only one device at a time.
jimmy
United States
Nov 01, 2017
Default setting is to only allow login to a single device. Need to set it to "logout" which will log a user out of a phone when he / she tries to login to another phone.

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: A

When configure the Extension Mobility, I did not see any where to select phone model. But the exam choose A for the answer, and I’m sure that I can login 2,3 different model…

For B, if the profile is misconfigured then he can’t also login at office also.

For C is wrong for sure, I think user can login 10 devices at a time.

For D I don’t think device pool has anything to do with it but I have no support for this.

So What is the correct answer?
Sunday
Italy
Nov 01, 2017
@argus
I'm arranging a file with correct questions.
I have found technical references for most of the questions, except 4/5 which are inconsistent and ambiguous.
Sunday
Italy
Nov 01, 2017
@JOHN:
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: A

The correct answer is C. here my 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.

I guess C is the correct answer also because the question gives you a specific detail: The User forgot to log out of his phone...

Hope this helps!
Attila
Hungary
Oct 31, 2017
@John Q154

In my opinion, C is the right answer.

Reasoning:
I was assuming, that the Device Mobility Service Parameter is left in its default setting, which is Multiple login not allowed.
If you have a CUCM available, you can check it. You should look for "Multiple Login Behavior".

Thoughts?
argus
United States
Oct 31, 2017
Is the 161q file spoken of here the "dump" file? Here and on other sites, folks say the Q are fine but the A have problems. Has anyone fixed it? I'm so close to passing but keep failing by 50 or 100 points. I can build these things but can't test.
John
United States
Oct 31, 2017
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: A

When configure the Extension Mobility, I did not see any where to select phone model. But the exam choose A for the answer, and I’m sure that I can login 2,3 different model…

For B, if the profile is misconfigured then he can’t also login at office also.

For C is wrong for sure, I think user can login 10 devices at a time.

For D I don’t think device pool has anything to do with it but I have no support for this.

So What is the correct answer?
Attila
Hungary
Oct 31, 2017
Hello Guys,

FYI. The 161q is still valid. I just passed my exam with a score 869. :)
This was my second try, cause I faild my first attempt with 835.

I've got a recommendation for all of you, who are using dump to prepare for the exam. ALWAYS READ ALL ANSWERS CARFULLY! For example. Q136, where A is marked for right answer. Have use seen, that Cisco UNITED CME is written and not UNIFIED!

One more thing. Most of the questions were from the last part of the dump (~60).

I wish you all good luck!
john
United States
Oct 31, 2017
QUESTION NO: 147
Should be A and F.
John
United States
Oct 31, 2017
@Sam
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

Should be Sip Route Pattern. for SIP end point call to h323 via ip address, you implement siptrunk between cucm and create a sip route pattern point to sip trunk.
Adam
United States
Oct 30, 2017
What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM Endpoints

A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
B. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS
C. Media Resource Group List
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS

Your answer

A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS

Correct answer
B. 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


Can someone explain or show me how you can configure a neighbor zone on UCM ??
John
United States
Oct 30, 2017
1. which 2 things do not utlise MTP
a> h.323 fast start  require MTP
b> IPV6 -IPV4 transform not require
c> DTMF inband RTP-NTE (rfc2833) require MTP only 4.0, 5 and late was removed requirement mtp.( CUCM 5.x and later remove the requirement for an MTP when supporting RFC 2833 DTMF)
d> delayed offer h.323  requirement MTP (need to check MTP require)
http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_8241.html

so B and C
Adam
United States
Oct 30, 2017
So to everyone here that is studying and has failed and trying to get the correct answers. My question to everyone is are we looking to put the correct answers down for this test, OR what CISCO has as the correct answer from these dumps we are using... I am very confused.
king
Philippines
Oct 30, 2017
Hi Guys
What is the correct answer of the below question?

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 think the correct answer will be B & D. Reasons of my answers are :
A. h.323 fast start ->Yes - MTP always required for Outbound Fast Start - Voice Calls Only supported (WRONG ANSWER)
B. IPV6 -IPV4 -> IP Converstion Does not need MTP (CORRECT ANSWER)
C. DTMF inband RTP-NTE (rfc2833) -> SIP gateways that support only NTE require MTP resources to be allocated when communicating with endpoints that do not support NTE.
D. delayed offer h.323 -> Dont see any delayed offer for h.323 . H323 has only fast start and slow start and both required MTP.

What do you think guys about my answer and your thoughts.
king
Philippines
Oct 30, 2017
@Sunday

Thanks for the clarifications for the Question 152. I was wrong about the answer of this question but today after viewing your post
I cross checked with my Voice Gateway and I found your are right. show ephone registered and show sip-ua status registrar shoudl be the correct answer.

It is not TRUE that there is no command 'show voice register session-server', in latest IOS has this command.

QUESTION 152
Which two commands verify Cisco IP Phone registration? (Choose two.)
Explanation:
A.show telephony-service ephone-dn -> Show ephone-dn configuration (WRONG)
B.show voice register session-server -> "gateway-1.#show voice register ?
all Show all SIP CME/SRST details
credential Show voice register credential
dial-peers Show dial-peers created dynamically via REGISTERs
dn Show given dn details
global Show voice register global
license Show voice register license
pool Show given pool details
session-server Show registered session servers (WRONG, Because it shows the session server not the IP Phone)
statistics Show voice register statistics"
C.show ephone registered -> Registered ephone status (CORRECT)

D.show ccm-manager hosts -> Hosts Info (WRONG)
E.show sip-ua status registrar -> registrar Display SIP Registrar Clients (CORRECT, For SIP phone in CME or SIP SRST)
Sam
Netherlands
Oct 30, 2017
Thanks for answering my previous questions. I got another one:


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

Thanks.
Adam
United States
Oct 29, 2017
@king

Can you advise to what question #'s you are referring to in your reply back to me?
Sunday
United Kingdom
Oct 29, 2017
Hi Guys,
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

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. used to display all SIP endpoint registered.
So I say 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

Hoping this helps!
King
Philippines
Oct 29, 2017
@Fe

Could you please share your thought on the exam. is 161Q still valid.
king
Philippines
Oct 29, 2017
TO @Adam

As per my last exam experience I got 71% correct and I did not do any study except dumps. After doing little study I think below questions answers are wrong.

161,160,155,153,146,145,144,138,135,132,129,127,142,125,124,123,122,120,118,57,63,68,
And I am confused to below questions.

61,80,130,148,147,

FYI, I most f the questions came to exam were after Question Number 100.

I am dare to take another exam due to cost. I am waiting if someone could give a proper outline or answers.

Please let us know Adam.
King
Philippines
Oct 29, 2017
@Fen,

Thanks for the clarification on Question 123.
When are you planning to take next exam.?

Thanks
king
Philippines
Oct 29, 2017
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.

Is DF are correct answer. I think D and E.
Reason For Choosing E over F is below.
1.When implementing TEHO in a multicluster deployment, configure ICTs
between the clusters. Then you must add a route pattern per TEHO destination in each cluster.
The route pattern refers to the corresponding TEHO trunk as the primary path and uses the local
route group feature for the backup path.
2.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.
For the example in Figure 10-2, in order to implement local failover TEHO routes to Brazil, a site in Paris, France would require a dedicated route
pattern and route list to route the calls to the TEHO gateways in Brazil as a first choice or to the Paris gateways as a second choice. Because the
pattern is linked to a site-specific route list, it cannot be reused at any other site. Likewise, the site in Ottawa, Canada requires its own dedicated
route pattern pointing to an Ottawa-specific route list to allow local failover to a gateway in Ottawa.

Please give your thought
Thanks
king
Philippines
Oct 28, 2017
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 think 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.
What is your thought guys?
Bill
United States
Oct 28, 2017
Q125

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 voice register global dn
D. Configure telephony service
E. Configure the SIP registrar
F. Configure voice register pool

Your answer
B. Configure an SRST reference
C. Configure voice register global dn
F. Configure voice register pool

Correct answer
B. Configure an SRST reference
C. Configure voice register global dn
E. Configure the SIP registrar
Bill
United States
Oct 28, 2017
Q57

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

Your answer B. The location Hub_None has not been activated
Correct answer A. The DX650's MAC address is incorrect in the Cisco UCM


Reasoning:
In the tab DX650 Configuration it shows the MAC address entered " D0C78914131D "
Bill
United States
Oct 28, 2017
Question 68

What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM Endpoints

A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
B. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS
C. Media Resource Group List
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS

Your answer

A. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
E. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS

Correct answer
B. 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
Bill
United States
Oct 28, 2017
Question 86

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. SRST
B. CFUR
C. LRQ
D. AAR

Your answer: B

Correct answer: D

SRST is for call control survivability

LRQ is a H323 location request message

CFUR is Call Forwarding Un Registered

AAR is Automatic Alternate Routing used for PSTN routing in event of inadequate bandwidth

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b03aar.html
Teri
Hungary
Oct 28, 2017
QUESTION 6
You want to avoid unnecessary interworking in Cisco TelePresence Video Communication Server, such as where a call between two H.323 endpoints is made over SIP, or vice versa. Which setting is recommended?
A. H.323 - SIP interworking mode. Reject
B. H.323 - SIP interworking mode. On
C. H.323 - SIP interworking mode. Registered only
D. H.323 - SIP interworking mode. Off
E. H.323 - SIP interworking mode. Variable

My answer is "C"
explanation:
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.

http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/admin_guide/Cisco_VCS_Administrator_Guide_X7-2.pdf
Page 88.
Abishai
United States
Oct 27, 2017
So is the 161Q&A on the files valid or not? Please letteth me know.
Sam
Netherlands
Oct 27, 2017
Hi there,

Can someone give answer to the following two questions?

What are two important considerations when implementing TEHO to reduce long-distance cost?
(Choose two.)
A.on-net calling patterns
B.E911 calling
C.number of route patterns
D.caller ID

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.

Thanks.
Sam
Netherlands
Oct 27, 2017
Hi guys,

Thanks for your answers. I got another one:

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

It says B and C. I can't find any command with "show voice register session-server". There is a command "voice register session-server" but not with "show".

Thanks.
Teri
Hungary
Oct 27, 2017
QUESTION 94
Which code snippet is required for SAF to be initialized?(Choose three.)
A. Service Family
B. External-Client
C. router eigrp
D. topology base

I think the correct answer is "C"
Service Family is written with - :
service-family {ipv4 | ipv6} [vrf vrf-name] autonomous-system autonomous-system-number
John
United States
Oct 27, 2017
Question 122
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#JABW_TK_S11EF173_00
Adam
United States
Oct 27, 2017
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
H. 263
E. BFCP
Ans: E


q161 indicate that answer H. 263 is correct


SCCP phones register to how many nodes?
a. 1
b. 2
c. 3
d. 4


q161 indicates that the answer is B.2


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

q161 indicates that the answer is B


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.


q161 indicates that the answer is BD


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


q161 indicates that the answer is DEF



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.
F. Configure telephony service.

I have seen so many answers on this, does anyone have a definitive 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

my answer: A




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

I believe its ABC





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.

my answer: B

161 indicates D???




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

161 indicates A???





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

161 indicates B??? I have seen D on several other dumps??









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: B,C,D


Exam only asks for one answer?






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: B,C,F,H


Exam only asks for one answer?













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: C,E


Exam only asks for one answer?
Adam
United States
Oct 26, 2017
@Fen and @King

Is there a consolidated list of questions that need answers to? I am reaching out to the source of where I purchased my 161q, and demanding the right answers be given. Please advise
John
United States
Oct 26, 2017
QUESTION NO: 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: AB

Phone C7965 is not a video phone. Cisco Jabber Desktop is. in this case DSCP for Video Calls service parameter should affect on DX-650 and Cisco Jabber even it is not physical phone. What do you guy think?
king
Philippines
Oct 26, 2017
@Fen
QUESTION 94
Answer will be C because of below.
1. enable

2. configure terminal

3. router eigrp virtual-instance-name
Enables an EIGRP virtual instance in global configuration mode.

4. service-family {ipv4 | ipv6} [vrf vrf-name] autonomous-system autonomous-system-number
Enables a Cisco SAF service family for the specified autonomous system on the router.


5. exit-service-family
John
United States
Oct 26, 2017
@Fen question 94
Enabling Cisco SAF
To enable Cisco SAF and create a Cisco SAF service-discovery process, use the following commands:

SUMMARY STEPS
1. enable

2. configure terminal

3. router eigrp virtual-instance-name

4. service-family {ipv4 | ipv6} [vrf vrf-name] autonomous-system autonomous-system-number

5. exit-service-family

So the answer should be C.
Cesar
Spain
Oct 26, 2017
Hello, anyone passes this lately?
Fen
Australia
Oct 26, 2017
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.

my answer: A,E
reasoning:
A: TRUE - 48 phones supported
B: FALSE - some phones are working
C: FALSE - dunno?
D: FALSE - some phones are working
E: TRUE - 48 phones supported

your thoughts?
Fen
Australia
Oct 25, 2017
QUESTION 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

my answer: A
reasoning:
A: TRUE - use zone transforms to modify an alias before the query is sent to a target zone or policy service
B: FALSE - could use but too heavy handed approach
C: FALSE - just no
D: FALSE - call policy specifies an external device for call handling

your thoughts?
Fen
Australia
Oct 25, 2017
QUESTION 94
Which code snippet is required for SAF to be initialized?(Choose three.)
A. Service Family
B. External-Client
C. router eigrp
D. topology base

my answer: A
reasoning:
A: TRUE - how to Configure a Cisco SAF Forwarder service family is the SAF specific command - CONFIRMED
B: FALSE - Configuring a Cisco SAF External Client
C: TRUE - used to enter the SAF configuration on the router
D: FALSE - Configuring a Cisco SAF External Client

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/feature/guide/SAF_FeatureModule.html
SAF and Cisco IOS Service Advertisement Framework
Voice SAF is a subset of Cisco IOS Service Advertisement Framework. Before Voice service SAF is configured, it must first be enabled and configured as a Cisco IOS SAF service family to initiate the SAF service-discovery process. Additionally, interface-specific commands must be configured under service-family for Cisco SAF.

your thoughts?
Sam
Netherlands
Oct 25, 2017
@aungmyozaw: Congratulations on passing the exam! Which answers did you select in exam? You are lucky because passing score is 860 and you got 9 points more :) Any help will be appreciated.

Thanks.
king
Philippines
Oct 25, 2017
@Fen
QUESTION 132
Yes, The answer will Local Route Gorup.
QUESTION 122
Thanks For the clarification
QUESTION 93
I didn't find any evidence for this question but I have a feeling that Answer B is
wrong and the correct answer will be C. Not sure, If this question comes in my next
exam I will go with B.

QUESTION 95
Anser B is correct because look at the below tips on cisco documents
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.

QUESTION 43
Answer B is correct
Reason:
If you have not already done so, configure the Cisco IOS router as the SAF forwarder.
This is first step of this configuration.


QUESTION 129
E is the correct answer. Beacuse of the below tips
As of Cisco Unified Communications Manager version 8.6(2),
you must enable BFCP on the SIP trunk to allow video desktop sharing
capabilities between nodes in a Cisco Unified Communications Manager cluster.
To enable BFCP on the SIP trunk, do the following:

Please let's discuss if you feel any answer is wrong.
Fen
Australia
Oct 25, 2017
@ king
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 hear 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.
my answer: A,E
reasoning:
A: TRUE - the default gateway is not configured.
B: FALSE - The port direction is reversed! if this was accurate then C could hear A&B, not the other way around.
C: FALSE - cannot find evidence to support this answer
D: FALSE - it is responding because the call is being setup
E: TRUE - paths need to be established in both directions

check this document, search for default gateway
http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.html
Fen
Australia
Oct 25, 2017
@Sam
Can someone verify the correct answer?
QUESTION 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

my answer: B
reasoning:
A: if a site is not on the WAN, a dial peer will not work.
B: each sites gateway will need a personalized translation pattern to reach every other site when is SRST mode.
C: not an essential feature
D: MGCP not essential, could use H.323

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 sing

My answer:B
reasoning:
A: there is no publisher node
B: Configures a Cisco SAF External Client with the specified client label and optionally, a basename.
Specifying the basename keyword allows SAF external clients to use a naming convention based on the client-label. The naming convention takes the form of client-label@[1-50] where you can specify a maximum of 50 SAF external clients.
For example, if the external-client command specifies a client label of example, then the basename for a SAF external client would be example@1. Another SAF external client would be example@2, and so on up to a maximum of 50 basenames (@50).
C: it self advertises the nodes?
D: SAF security profile is for CUCM authentication

QUESTION 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

My answer: B
reasoning:
A: FALSE - SAF must not be on gatekeeper controlled trunks.
B: TRUE - SAF forwarders are used for everything related to SAF
C: FALSE - an example SAF service is Call Control Discovery (CCD) for Cisco Unified Communications cluster with an instance ID number
D: FALSE - H.225 trunk is a gatekeeper controlled trunk, SAF must not be used on gatekeepers.

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
H. 263
E. BFCP

My answer: E
reasoning:
A: FALSE - not a codec
B: FALSE - H.264 AVC video codec
C: FALSE - FECC Far End Camera Control
D: FALSE - H.263 Video codec
E: TRUE - Video Presentation sharing (BFCP)
Fen
Australia
Oct 24, 2017
@ king
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
my answer: A,C
reasoning:
A: TRUE - DX650 is CUCM registered Phone device type
B: FALSE - Jabber is virtual CUCM device type
C: TRUE - CP-7965 is CUCM registered Phone device type
D: FALSE - EX60 is CUCM registered TelePresence device
E: FALSE - MX200 is CUCM registered TelePresence device
Fen
Australia
Oct 24, 2017
@ king
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

my answer: C
reasoning:
A: FALSE - AAR is to contact a registered device on the CUCM via an alternative method.
B: FALSE - Call Forward unregistered - provides backup dialling to device when unregistered to CUCM
C: TRUE - Local Route Groups - TEHO is to contact a local number at a distant location.
D: FALSE - Survivable Remote Site Telephony - invoked at remote site if the link is lost from the CUCM
Rier
Malaysia
Oct 24, 2017
161 dump is valid but many answers in the dump 161 are wrong.

Failed today with 723 :(
aungmyozaw
Myanmar
Oct 24, 2017
@Sam,Thank you.:) @Fen , I thought most of your answers was right.
@Sam, Here is my answers of your questions in the exam.

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
Ans: B

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 sing
Ans: B

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
Ans: B

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
H. 263
E. BFCP
Ans: E
king
Philippines
Oct 24, 2017
@Fen
Question Number 6,
I think Correct answer will be H.323 - SIP interworking mode. Registered only, Because this is
Cisco VCS with H.323 endpoints initiating a Multiway conference.
"The Multiway Conference Factory functionality is SIP based. To allow H.323 endpoints to initiate a
Multiway conference:
1. Go to VCS configuration > Protocols > Interworking.
2. Set H.323 <-> SIP interworking mode to Registered only (or On is also acceptable).

QUESTION 123

I think the correct answer is B & E.
Reason-1: TMS is an option deployment in Telepresence VCS-C and VCS-E. So I dont think I would be matter to oneway audio calls.
Reason-1: NAT device need to allow more than RTCP and RTP from Internal to DMZ.
Details you will find below
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Basic_Configuration_Cisco_VCS_Control_with_Cisco_VCS_Expressway_Deployment_Guide_X7-1.pdf

Please let me know if you think this is correct.

QUESTION 160
Correct answer is Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key only. This question is wrong.

details you find
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-1/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-1.pdf

I am agreed with you answer with below questions.
QUESTION 66
QUESTION 63
QUESTION 58
QUESTION 67

Could you please take a took at question 122. I am confused for that. Hope you will reply with your tought
king
Philippines
Oct 24, 2017
@Fen
I think the answer of the below question will be A. Because CM does not allow duplicate registration.

"DuplicateRegistration - Unified CM detected that the device attempted to register to two nodes at the same time.
Unified CM initiated a restart to the phone to force it to re-home to a single node.
No action is necessary; the device will re-register automatically."

SCCP phones register to how many nodes?
a. 1
b. 2
c. 3
d. 4

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/err_msgs/8_x/ccmalarms861.html

Please let me know if you think otherwise
Sam
Netherlands
Oct 23, 2017
Can someone verify the correct answer?

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


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 sing


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


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
H. 263
E. BFCP

Thanks.
Renan Petrosino
Brazil
Oct 23, 2017
@aungmyozaw, Do you have any news?
Fen
Australia
Oct 23, 2017
QUESTION 160
Which three statements about when user A calls user С 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. 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.

my answer: A,B,E
reasoning:
A: TRUE - ports are open
B: TRUE - Apply an Advanced Networking option key on any VCS Expressway that needs static NAT
C: FALSE - only the VCS Expressway supports static NAT
D: FALSE - not essential
E: TRUE - ports are open
F: FALSE - IP addresses are not correct

I don't like my answers to this question, what are your opinions?
this question came up and only wanted ONE answer, not three.
A and E are required (confirmed on live system) so the SINGLE answer would be B?
Fen
Australia
Oct 23, 2017
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.

my answer: C,E
reasoning:
A: FALSE - the internal devices can connect to device C
B: FALSE - Real-time Transport Protocol (RTP)/RTP Control Protocol (RTCP), RTCP provides out-of-band statistics and control information for an RTP session. Used for Internet > DMZ calls (external calling internal)
C: TRUE - cannot find evidence to support this answer
D: FALSE - it is responding because the call is being setup
E: TRUE - paths need to be established in both directions

I don't like my answers to this question, what are your opinions?
Fen
Australia
Oct 23, 2017
QUESTION 101
What happens when a user logs in using the Cisco Extension Mobility Service on a device for which the user has no user device profile?
A. The Extension Mobility log in fails.
B. The device takes on the default device profile for its type.
C. The user can log in but does not have access to any features, soft key templates, or button templates.
D. The device uses the first device profile assigned to the user in Cisco Unified Communications Manager.

my answer: B
reasoning:
??

what are your opinions?
Fen
Australia
Oct 23, 2017
QUESTION 67
The intercluster URI call routing no longer allows calls between sites. What is the reason why this would happen?
A. Wrong SIP domain configured.
B. User is not associated with the device.
C. IP or DNS name resolution issue.
D. No SIP route patterns for cisco.lab exist.

my answer: C,D
reasoning:
A: FALSE - no SIP domain details supplied
B: FALSE - no user details supplied
C: TRUE - DNS has pub and Sub addresses, does not have CIMP address
D: true - but no details provided

what are your opinions?
Fen
Australia
Oct 22, 2017
QUESTION 68
Which three configuration tasks need to be completed on the router to support the registration of Cisco Jabber clients? (Choose three.)
A. The DNS server has the wrong IP address.
B. The internal DNS Service (SRV) records need to be updated on the DNS Server.
C. Flush the DNS Cache on the client.
D. The DNS AOR records are wrong.
E. Add the appropriate DNS SRV for the Internet entries on the DNS Server.

my answer: B,C,E
reasoning:
A: the DNS server has the correct IP address of the CUCM Pub & Sub
B: true
C: true
D: not sure what AOR records are
E: the DNS server only has internal addresses so far

what are your opinions?
Fen
Australia
Oct 22, 2017
QUESTION 63
After adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Device Pool cannot be default.
B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router.
C. The router does not support SRST.
D. The SRST enabled router is not configured correctly.

my answer: D
reasoning:
A: device pool can be default
B: diagram is illegible
C: no evidence to support this statement (no router version supplied)
D: true - none of the IP addresses provided match up

what are your opinions?
aungmyozaw
Myanmar
Oct 22, 2017
@Fen, Thank you so much for your answers and explanation . I just luckily passed the exam with 869 marks.
Fen
Australia
Oct 22, 2017
what are your opinions on this question?
QUESTION 6
You want to avoid unnecessary interworking in Cisco TelePresence Video Communication Server, such as where a call between two H.323 endpoints is made over SIP, or vice versa. Which setting is recommended?
A. H.323 - SIP interworking mode. Reject
B. H.323 - SIP interworking mode. On
C. H.323 - SIP interworking mode. Registered only
D. H.323 - SIP interworking mode. Off
E. H.323 - SIP interworking mode. Variable

my answer: D
reasoning:
A: - not a valid option
B: - VCS will ALWAYS interwork H.323-SIP calls
C: - VCS will interwork ONLY IF one of the endpoints is locally registered
D: - VSC WILL NOT interwork calls.
E: - not a valid option
Fen
Australia
Oct 22, 2017
SCCP phones register to how many nodes?
a. 1
b. 2
c. 3
d. 4
Answer: A or B??
reasoning:
By default, SCCP phones send a keepalive to their primary CUCM server every 30 seconds and to their failover node, which is the second node listed in the phone's Call Manager (CM) Group, every 60 seconds.

Cisco IP phones also send a SCCP keepalive to their secondary node. This is done to maintain and monitor a TCP connection between the phone and the secondary CUCM in order to facilitate a prompt and reliable failover should the need arise. The secondary CUCM, however, does not have a SCCP connection (as the phone has not registered to the secondary node at this point) and will therefore only ACK the TCP connection in response to the SCCP keepalive sent by the phone.

Does this mean it has ONLY registered to the primary (but is aware of the secondary node)
Adam
United States
Oct 22, 2017
Has anyone used the 161q and actually passed the exam? I sat today and failed after getting 95% of the questions from the 161q.

Can anyone please advise on recent exam results?
aungmyozaw
Myanmar
Oct 21, 2017
@Fen, Thank you so much for your answers and explanation. I just passed the exam with 869 marks.
king
Philippines
Oct 21, 2017
@ Fen,

I am agreed with all of your answer except 132. I still think the answer is LRG.

I am confused with Question number 122. what is your answer on that.

Thanks
Fen
Australia
Oct 21, 2017
CORRECTION!!
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

my answer: C
reasoning:
A: SAF must not be on gatekeeper controlled trunks.
B: SAF is Cisco Proprietary so must be on cisco routers.
C: SAF CCD - With the call control discovery feature, each local Cisco Unified Communications Manager cluster can perform the following tasks:
D: H.225 trunk is a gatekeeper controlled trunk, SAF must not be used on gatekeepers.


•Establish an authenticated connection with the SAF network
•Advertise the cluster to the SAF network by providing the IPv4 address or hostname of the server, the signaling protocol and port numbers that the SAF network uses to contact the cluster, and the directory number patterns that are configured in Cisco Unified Communications Manager Administration for the cluster
•Register with the SAF network to listen for requests that are coming from other remote call-control entities that also use the SAF-related network
•Use the information that is learned from the advertisements to dynamically add patterns to its master routing table, which allows Cisco Unified Communications Manager to route and set up calls to these destinations by using the associated IP address and signaling protocol information.
•When connectivity to a remote call-control entity gets lost, the SAF network notifies Cisco Unified Communications Manager to mark the learned information as IP unreachable. The call then goes through the PSTN.
•Provide redundancy to advertise and listen for information, so if a server loses connectivity to its primary SAF forwarder for any reason, another backup SAF router can be selected to advertise and listen for information.
Fen
Australia
Oct 21, 2017
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

My answer: B
Reasoning:
A: SAF must not be on gatekeeper controlled trunks.
B: SAF is Cisco Proprietary so must be on cisco routers.
C: SAF is a network layer service, you don't need a cluster to use it, can be distributed devices
D: H.225 trunk is a gatekeeper controlled trunk, SAF must not be used on gatekeepers.
aungmyozaw
Myanmar
Oct 21, 2017
@king , Thanks for answering . Today i will sit the exam and will post the result.
Fen
Australia
Oct 21, 2017
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.

my answer: D,E
Reason:
A: FALSE - this is the Policy Services
B: FALSE - this is a cluster
C: FALSE - this is the traversal Zone
D: TRUE
E: TRUE
The Local Zone’s subzones are used for bandwidth management and to control registration and authentication policies.
>So D for sure
The Subzones page (Configuration > Local Zone > Subzones) lists all the subzones that have been configured on the VCS, and allows you to create, edit and delete subzones. For each subzone, it shows how many membership rules it has, how many devices are currently registered to it, and the current number of calls and bandwidth in use. Up to 1000 subzones can be configured.

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).

Bandwidth management
The Local Zone’s subzones are used for bandwidth management. After you have set up your subzones you can apply bandwidth limits to:
> individual calls between two endpoints within the subzone
> individual calls between an endpoint within the subzone and another endpoint outside of the subzone
> the total of calls to or from endpoints within the subzone
For full details of how to create and configure subzones, and apply bandwidth limitations to subzones
including the Default Subzone and Traversal Subzone, see the Bandwidth control section.

Registration, authentication and media encryption policies
In addition to bandwidth management, subzones are also used to control the VCS's registration,
authentication and media encryption policies.
Renan Petrosino
Brazil
Oct 20, 2017
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
Answer: B

answer B is correct? I believe to be "C" the correct answer
Fen
Australia
Oct 20, 2017
QUESTION 142
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

My answer: CDF
reasoning:
A: FALSE - VCS are gateways so F is more accurate
B: FALSE - basic endpoint wont consume a licence *is it a traversal-enabled endpoint?
C: TRUE - is a traversal client
D: TRUE - is a traversal client
E: FALSE - provides the link/method of transmission but not an end point
F: TRUE - calls involving either VCS may consume a license

http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html
VCS Traversal Call License Usage
When a call is made and the VCS takes the media as well as the signaling, it is a traversal call and uses a traversal call license on that VCS. Here are some examples of traversal calls that require the VCS to take the media:
For a VCS Control, calls to or from a traversal server (known as Firewall traversal calls).
For a VCS Expressway, calls to or from a traversal client (Firewall traversal calls). Traversal clients include other VCSs, gatekeepers, Border Controllers, or traversal-enabled endpoints.
Calls that are gatewayed (interworked) between H.323 and Session Initiation Protocol (SIP) on the local VCS.
Calls that are gatewayed (interworked) between IPv4 and IPv6 addresses on the local VCS.
For VCSs with Dual Network Interfaces enabled, calls that are inbound from one LAN port and outbound on another.
An SIP-to-SIP call when one of the participants is behind a Network Address Translation (NAT), unless both endpoints use Interactive Connectivity Establishment (ICE) for NAT traversal.
Calls that have a media encryption policy applied.
Encrypted calls to and from the Microsoft Office Communications Server (OCS) Version 2007 or Microsoft Lync Server Version 2010, where the OCS/Lync back-to-back user agent (B2BUA) is not used. If the B2BUA is used, the B2BUA application always takes the media, but the call is not classified as a VCS traversal call and does not consume a traversal call license (it might still consume a non-traversal license if the VCS takes the call signaling).
Fen
Australia
Oct 20, 2017
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.
F. Configure telephony service.
answer BDE or ABC or BCE?

My answer is: B,C,E
reasoning:
A: FALSE - not a required step for SRST configuration
B: TRUE - you need a SRST reference
https://supportforums.cisco.com/discussion/10924876/srst-reference-explanation
C: TRUE - you need an SIP registrar http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080557e81.html
D: FALSE - there is no voice register global dn command
E: TRUE - there is a voice register pool - Enters voice register pool configuration mode.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/command/reference/srstcr/srsa_n_z.html#wp3302578069
F: FALSE - used for CME SRST
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmesrst.html
Fen
Australia
Oct 20, 2017
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.

Answer C or D? I think D, your thoughts?

my answer is: C
A: FALSE - there is no access control list on the VCS-C
B: FALSE - this zone enables outbound calls, a traversal zone on the VCS-E enables inbound calls
C: TRUE - if internal devices have registered to the VCS-C then the local zone needs to have a search rule configured to direct calls.
D: FALSE - this would impact outbound calls to the VCS-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

my answer: A
reasoning:
A: TRUE - Automatic Alternative Routing - Cisco Unified Communications Manager automatically reroutes calls through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth
B: FALSE - Call Forward unregistered - provides backup dialling to device when unregistered to CUCM
C: FALSE - Local Route Groups - used to simplify TEHO call routing configuration
D: FALSE - Survivable Remote Site Telephony - invoked at remote site if the link is lost from the CUCM

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

My answer: A,B,C
reasoning:
A: TRUE - phones should failover to secondary subscriber
B: TRUE - resources failover to secondary CUCM
C: TRUE - SCCP can do SRST mode but only if configured
D: FALSE - router setting not CUCM
E: FALSE - not heard of this?
F: FALSE - SCCP fallback is configured on routers, not CUCM

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.

my answer: B
reasoning:
A: FALSE - on-net call volume exceeds bandwidth prompting AAR use
B: TRUE - WAN bandwidth is maxing out so AAR is routing calls via the PSTN
C: FALSE - working within bandwidth limits
D: FALSE - reporting fine


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

my answer: D
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.
Renan Petrosino
Brazil
Oct 20, 2017
Hello guys,
I failed last week using 161q, but based on documents and research'm forwarding the answers you believe are correct.
Please help me and tell me if my answers are incorrect, I'll do it again my exam tomorrow.

which 2 things do not utlise MTP
a. h.323 fast start
b. IPV6 -IPV4 transform
c. DTMF inband RTP-NTE (rfc2833)
d. delayed offer h.323
Answer: A,B

Hardware MTP requires 2 things:
a. PVDM or DSP resource
b. LTI local transcode resource
c. ref2833
d. one audio codec
e. T1 PRI card
Answer: A,B

SCCP phones register to how many nodes?
a. 1
b. 2
c. 3
d. 4
Answer: B

VCS monitors Presence Status using what:
a. start call
b. registration
c. end call
d. call starting
Answer: B

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: ABF

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

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.
ANSWER: C

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

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

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

What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
A. CS4/32
B. CS6/48
C. EF/46
D. AF41/34
E. CS3/24
Answer: A

Which statement about the function of the "+" symbol in the E.164 format is true?
A. The "+" symbol matches the preceding element one or more times.
B. The "+" symbol matches the preceding element zero or one time.
C. The "+" symbol represents the international country code.
D. The "+" symbol represents the international call prefix.
Answer: D

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

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. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
B. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
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. Media Resource Group List.
Answer: AB

Which two options enable routers to provide basic call handling support for Cisco Unified IP Phones if they lose connection to all

Cisco Unified Communications Manager systems? (Choose two.)
A. SCCP fallback
B. MGCP fallback
C. Cisco Unified Survivable Remote Site Telephony
D. Cisco Unified Communications Manager Express
E. Cisco Unified Communications Manager Express in SRST mode
Answer: BE

Thanks!
king
Philippines
Oct 19, 2017
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

Is A&C are the correct answer? Defiantly D&E are not correct. Just looked at collab 10.x guide
and found that Cisco Jabber also use DSCP AF41 for video call but Cisco Jabber is a software-based desktop clients
Application, thus it means Cisco Jabber is not a device and Answer A&C are correct
king
Philippines
Oct 19, 2017
@aungmyozaw

Answer of the below question will be DEF because VCS calculate license based on VCS Nontraversal and traversal zone license which includes H323 and SIP and VCS itself.

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

If you think it is in different then please explain. I am also doing study and I failed too.
aungmyozaw
Myanmar
Oct 19, 2017
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

Pls. Can anyone answer this question?
King
Philippines
Oct 19, 2017
@aungmyozaw

I think the correct answer is LRG. Because
"When theprimary (TEHO) path is not admitted as a result of reaching the CAC call limit, calls should be
routed through the local gateway."

Please le me know what do you think
aungmyozaw
Myanmar
Oct 19, 2017
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. SRST
C. CFUR
D. LRG

Pls. What is the correct answer?
aungmyozaw
Myanmar
Oct 19, 2017
@Sheraz,Is there any link for the answers?
aungmyozaw
Myanmar
Oct 19, 2017
VCS monitors Presence Status using what:
a>start call
b>registration
c>end call
d>call starting

I think "Answer is B".
http://www.cisco.com/c/en/us/td/docs/telepresence/infrastructure/articles/vcs_monitors_presence_status_endpoints_kb_186.html
Ramesh
United Kingdom
Oct 18, 2017
Hi JUAN

Congrats for passing.

Can you guide me with the following and tell me what answers you think are correct? and also if they came in the exam? I know other people in this forum have answered these but need to get your thoughts as these are confusing and you have passed already.


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.

Answer C or D? I think D, your thoughts?


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

I think Answer is ABC, however some say DEF which doesn't make sence, your thoughts?


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.

I think Answer is BC your thoughts?


Which two options enable routers to provide basic call handling support for Cisco Unified IP
Phones if they lose connection to all Cisco Unified Communications Manager systems? (Choose
two.)
A. SCCP fallback
B. Cisco Unified Survivable Remote Site Telephony
C. Cisco Unified Communications Manager Express
D. MGCP fallback
E. Cisco Unified Communications Manager Express in SRST mode

I think answer is BE your thoughts?


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 ABC or BCE?


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

Answer C or D


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.

I think answer is BE your thoughts?


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

I think answer is ABC your thoughts?


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 A or B, i think A, your thoughts?


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.

I think A and E? if BCD then no phones would work, thoughts?


Which statement about the function of the "+" symbol in the E.164 format is true?
A. The "+" symbol represents the international country code.
B. The "+" symbol represents the international call prefix.
C. The "+" symbol matches the preceding element one or more times.
D. The "+" symbol matches the preceding element zero or one time.

Answer A or B, i think B, your thoughts?

Sorry for the long list but it would be really helpful if you could answer these as it would really help us pass. thanks again mate.
John
United States
Oct 18, 2017
NEW questions:
1. which 2 things do not utlise MTP
a> h.323 fast start  require MTP
b> IPV6 -IPV4 transform not require
c> DTMF inband RTP-NTE (rfc2833) require MTP only 4.0, 5 and late was removed requirement mtp.( CUCM 5.x and later remove the requirement for an MTP when supporting RFC 2833 DTMF)
d> delayed offer h.323  requirement MTP (need to check MTP require)

2. SCCP phones register to how many nodes?
a>1 --> only registered to one subscriber at a time.
b>2
c>3
d>4

3. VCS monitors Presence Status using what:
a>start call
b>registration --> registrattion, call-ended and in-call
c>end call
d>call starting

4. Hardware MTP requires 2 things:
a>PVDM or DSP resource
b>LTI local transcode resource
c>ref2833
d>one audio codec
e>T1 PRI card

a,b
Jase
United States
Oct 18, 2017
Here's another one.. although I'm not 100% on one of the answers. Any feedback?

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 provided: BD
Correct Answer: D and either C or E
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).
Jase
United States
Oct 18, 2017
The below questions from 161q I believe an incorrect answer is provided. I listed what I believe are the correct answers and why.

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 provided: B
Correct answer: A
Reason: CFUR is for call rerouting when phones are unregistered. AAR is used when CAC bandwidth limits (call limits) are reached.

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
Correct Answer: 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

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 provided: DEF
Correct Answer: ABE
Reason: HSRP is not a CUCM feature (it is a router or gatekeeper feature). SCCP fallback is not a CUCM feature (SRST is the correct name). "H.323 redundant connection" is very vague.. I personally have never heard of this, seems incorrect. That leaves ABE for correct answers, which all make sense for redundancy testing.

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
Correct Answer: C
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...
Josh
United States
Oct 18, 2017
Where can I get the q161 version of test? please help.
John
United States
Oct 18, 2017
@Sheraz
SCCP phones register to how many nodes?
a>1
b>2
c>3
d>4

Why 2? I think end point can registered to 1 subscriber at a time.
Answer: B
king
Philippines
Oct 17, 2017
Guys !!

What do you think the right 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

CFUR is the the correct answer, I think the correct answer will be in betweer AAR and LRG.
Fen
Australia
Oct 17, 2017
just failed

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

SCCP phones register to how many nodes?
a>1
b>2
c>3
d>4

VCS monitors Presence Status using what:
a>start call
b>registration
c>end call
d>call starting

Hardware MTP requires 2 things:
a>PVDM or DSP resource
b>LTI local transcode resource
c>ref2833
d>one audio codec
e>T1 PRI card

(diagram with EX60/90 on VCS-E/C)
device A (inside network with VCS-C) calling device C (in DMZ with VCS-E) pick one:
a>VCSE
Fen
Australia
Oct 17, 2017
using dump dump 161, have fact checked all answers, some are blatantly wrong, doing exam today, will post results
Juan
Mexico
Oct 17, 2017
@Sam: I am not 100% about the answers. I analyzed question by question (I investigated about topics unknown for me) and I answered what I thought was fine without considering the answers in this Dump. I didn't make notes but if you have specific questions, post it and I'll tell you my point of view.
king
Philippines
Oct 17, 2017
@Juan,

Hope you have a note of the correct answers, Could you share with us.

Thanks
Sultan Al Arif
Ramesh
Hong Kong
Oct 16, 2017
@Juan
Congrats on passing
will you be able to give us a bit of guidance and paste on here which questions had the wrong answers in the dump pelase?
Thanks
Sam
Netherlands
Oct 16, 2017
@Juan: Did you make notes of correct answers in the dump?
King
Philippines
Oct 16, 2017
What is the correct answer of below question? I think the answer is B.

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.
King
Philippines
Oct 16, 2017
@Marc,

What is your plan for retake?
Bill
United States
Oct 16, 2017
I have seen many references to q161 version of test, but can not find dump. Also, has anyone had luck with dump?
Bill
United States
Oct 16, 2017
@xbr I agree. I was just pointing out that the files answers are incorrect. My answers pasted here are incorrect. @Sunday, you are corr my mistake. AF41 (Dscp 34) is for multimedia conferencing
Marc
Switzerland
Oct 16, 2017
Failed last week. Five or more new questions. Some answers in dump 161 were wrong
king
Philippines
Oct 15, 2017
Failed Today with 710. 95% questions are from 161 but some questions in dumps have multiple answers where as in exam you have to select only one. For example, question no 160 was in the exam but you need to select only one answer. I think 5 questions were not from the dumps. Guys where I could get guaranteed question ans right answer. My certification is going to expire soon. Please let me know the options are if anyone has that
Study material!!!
Colombia
Oct 15, 2017
what study material, book, student guide to get well prepared.

Please.
Ramesh
United Kingdom
Oct 15, 2017
@Sunday: Thanks again for your inputs

@John: Thank you too for your guidance on the corrected answers, have you given the exam yet? Keep us posted if you do and if your answers work

Thanks
Sunday
Italy
Oct 15, 2017
@Bill,
What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
A. CS4/32
B. CS6/48
C. EF/46
D. AF41/34
E. CS3/24
Correct answer is A, I work on Call Manager, both on version 8.6 and 10, and the default parameter is CS4.



About Question:
"Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server.
B. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
C. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
D. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
E. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
F. Verify that the H.323 redundant connection is active.
Answer: CDF
Your answer: CDF
PROBLEM: Just vetting this one. The previous version dump had different answers. "

I don't understand why HSRP should be involved... in my opinion, redundancy could be verified by checking the phones are registered to a second subscriber (they keep a TCP connection open to the secondary subscriber, I hope that's the "meaning" of the question.
so I would say: A, B for sure, then I do not know which answer to pick as third answer... they're not consistent in my opinion.
Luizzza
Canada
Oct 15, 2017
@John

the Traversal zone Search Rule answer is not complete and ambiguous:

It says: "The traversal zone on the VCS Control does not have a search rule configured"

Traversal Zone search rules need to be configured both in VCS-C and VCS-E, if the answer said: "The traversal CLIENT zone on the VCS Control does not have a search rule configured"

or if the answer was: "the traversal zones search rules on VCS-C and VCS-E are not configured" then it would be more clear, but that option is ambiguous therefore I wouldn't choose it.... thoughts?
xbr
Tunisia
Oct 14, 2017
@Bill
Can you explain why C.I think it would be BDF.

There is an HSRP on CUCM ?

Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server.
B. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
C. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
D. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
E. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
F. Verify that the H.323 redundant connection is active.
Answer: CDF
Your answer: CDF
PROBLEM: Just vetting this one. The previous version dump had different answers.
John
United States
Oct 14, 2017
@Luizzza

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?

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.
Sunday
Italy
Oct 14, 2017
Ramesh I think the correct answers are:
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
the reason is that the calling search space is not necessary: hosted DN pattern have to belong to a hosted DN Group, and SAF needs a SAF enabled trunk in order to work.
The CSS is not necessary. The partition has to be set in SAF, but it could be already included in a CSS. So in my opinion, CSS is not a correct answer.
Luizzza
Canada
Oct 14, 2017
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: ABC
Explanation: http://docwiki.cisco.com/wiki/Service_Advertisement_Framework_Support_in_Unified_Communications_-_System_Test_Configuration

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.

ANSWER: C
Explanation:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-1/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-1.pdf-=

Local Zone Search Rule:
To configure the search rules to route calls to the Local Zone (to locally registered endpoint aliases)

Traversal Zone Search Rule:
To create the search rules to route calls via the traversal zone

I know the question is ambiguous BUT is says: "Video
endpoints inside Company X have registered but are unable to receive calls from outside
endpoints" : The registered endpoints don't receive calls, without a LOCAL ZONE SEARCH RULE a registered endpoint WONT get a call
In addition, a Traversal Zone Search Rule needs to be configured both on the VCS-C and the VCS-E to work
and the answer says "traversal zone on the VCS-C doesn't have the search rule configured" it would need it on the VCS-E as well so the option is not
fully correct..... thoughts?
Ramesh
United Kingdom
Oct 14, 2017
Hello, Can someone verify below

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

Some dumps say ABC and some say ABE. I think both answers are right as all 4 services are required, but which one will CISCO accept as correct?
Thanks
Ramesh
United Kingdom
Oct 14, 2017
Thanks John/ Sunday for your responses

King: good luck keep us posted :)
Sunday
Italy
Oct 13, 2017
@king good luck, let us know if you pass it and if dump is reliable!!
Thank you!
Helmy
Saudi Arabia
Oct 13, 2017
@ King: Wish you good luck. please keep us posted. and what dum file you used
king
Philippines
Oct 13, 2017
Booked for 21st May and Now I am afraid. Any last hour tips guys.
John
United States
Oct 13, 2017
to Ramesh

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.

I think it should be D as the traversal zone is how an outside call would come in
John
United States
Oct 13, 2017
Well, I login CUCM myself and already verify, DSCP for video should be AF41(34) and TelePresence Calls should be cs4(32). in this case, I think I have to pick AF41(34) because we don't have another option.
Thanks Guys
Sunday
Italy
Oct 13, 2017
Ramesh, the correct answer is:
D. The traversal zone on the VCS Control does not have a search rule configured.
Mariusz
Poland
Oct 12, 2017
hi John

I think it might be answer B 34 (100010)

Here explanation:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/videodg/vidguide/qos.html
Andy
Ukraine
Oct 12, 2017
2 John:
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
Video -- AF41 (34)
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/videodg/vidguide/qos.html
Ramesh
United Kingdom
Oct 12, 2017
Hello can anyone answer the below, different dumps have different answers, Some say A and some say C, Thanks

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.
John
United States
Oct 12, 2017
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: A
Note: but I think it should be 32 (100000). so what is the correct answer?
aungmyozaw
Myanmar
Oct 12, 2017
Yep, I aslo failed. 161 is valid.
Helmy
Saudi Arabia
Oct 12, 2017
Any new news about the 161q ?
Bill
United States
Oct 11, 2017
Failed today with a 726. ALL NEW QUESTIONS. 239 is no longer valid. 161 is valid, but many of the answers in the dump file are wrong.
Bill
United States
Oct 11, 2017
OK, So the 161 is mostly new questions. Alot of new VCS questions. Also, some of the questions that are duplicates from the 239 now have different answers, some of which are dead wrong. I have the 300-075 scheduled for later today. Fingers crossed.
Martin
Hong Kong
Oct 11, 2017
Failed last week but passed yesterday with thin margin 87x. Most questions are from 161q but don't rely on the dump cause it has some wrong answers. Study using cisco docs and find the right answers. There are some new questions also.
Thiago
Brazil
Oct 11, 2017
hi guys, is dump 161q valid?
Max
Ukraine
Oct 11, 2017
I am failed today
Used dump from exam Q355
Max
Ukraine
Oct 11, 2017
I'm failed today. Use Passlear Q355
Hari
Indonesia
Oct 10, 2017
Any one took the exam recently ? is 161 questions valid ?
Blue
Canada
Oct 10, 2017
Hi guys,

Any body has information about updated Dump 160?
any body try it?
creepichi
Indonesia
Oct 10, 2017
Anyone tried with 161q pdf version?
moko
France
Oct 10, 2017
The dump 161, is valid?
cryptoprotocol
India
Oct 10, 2017
Does anyone tried 161 Questions
ZLEE
Malaysia
Oct 10, 2017
b, e, f
Steve
Philippines
Oct 09, 2017
@aungmyozaw: The answer is AEF.
Does someone tried the new 161Q, is it valid?
aungmyozaw
Myanmar
Oct 09, 2017
Is there anyone can answer this question ?
Which three items must you configure to enable SAF Call Control Discorery? (Choose Three.)
A. a calling serarch space
B. hosted DN patterns
C. translation patterns
D. route patterns
E. the SIP or H.323 turnk
F. hosted DN groups
Eva
Russian Federation
Oct 09, 2017
Asad, how can you explain that answers d: and e: is right? Why not A,B,C?
Exams questions are written in unclear manner and it's impossible to pass!
Ramesh
United Kingdom
Oct 09, 2017
Hello, There is a new dump on the internet with 161 questions seems to be released last week of April. Anyone tried this?
Asad
Canada
Oct 09, 2017
The answer are below:
B. verify that media resources fail over to a secondary subsriber server when the publishers failes
d: verify that HSRP is active on the Cisco Unified Communication manager subscriber servers
e: Verify that H.323 redundant connection is active
Edwin
Germany
Oct 09, 2017
Can any expert clarify and answer this question.
Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server
B. verify that media resources fail over to a secondary subsriber server when the publishers failes
c.Verify that Cisco unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected
d: verify that HSRP is active on the Cisco Unified Communication manager subscriber servers
e: Verify that H.323 redundant connection is active
f: verify that SCCP fallback is configured in Cisco Unified Communication Manager
WA
Unknown country
Oct 08, 2017
Failed today.
Blue
Canada
Oct 08, 2017
Any update about the exam
Jay
United Kingdom
Oct 08, 2017
Failed. 60-70% questions are different to these dumps. Not valid.
max
Poland
Oct 08, 2017
Did anyone passed after 06-Oct-2017
Mark
Lebanon
Oct 08, 2017
any new updated dumps? the current exam is not valid anymore
CiscoCert
United States
Oct 08, 2017
Failed today using 308. Mostly VCS/Expressway.
Alexandre
Portugal
Oct 07, 2017
I didn't pass with the 100 question exam, today.
Adrian
Ireland
Oct 07, 2017
The newest (ver. 16.021) dump from exam with 308Q is NOT valid. I failed. There are too many new questions.
jack
Romania
Oct 07, 2017
people are saying that 308q is not valid either
Thanzy
South Africa
Oct 07, 2017
@Zulus, where did you find the exam, from which site?
zulu56
Germany
Oct 07, 2017
There's a dump 308 Qestions in the web, from 12. April.
Did anyone tryed this one?
Zulu
Germany
Oct 07, 2017
Found a new dump with 308Q from 05.10.2017.
Anyone tryed this one?
Bil
United States
Oct 06, 2017
Test has changed.
Daniels
Nigeria
Oct 06, 2017
Ikkir, Frank, Bill and TED, pls did you use the 239 file?
maga
Germany
Oct 06, 2017
Are you use the 239q and failed???
ikkir
Philippines
Oct 06, 2017
Failed!! 90% of the questions are new and mostly related to video technology. Hoping to have the updated questions soon.
ZLEE
Malaysia
Oct 06, 2017
239q ^valid (read in regex) anymore. Mainly covered VCS and Expressway.
Bil
United States
Oct 06, 2017
Agree with Frank. Test is 90% different. Alot of Video/SAF/CCD/Location type questions.
Ted
Canada
Oct 05, 2017
This is weird , if most of the questions were related to video then in the cisco press book for the 300-075 they have not covered video as much.
Frank
Switzerland
Oct 05, 2017
Exam changed completely. Most of the question were video related
Mw
Ukraine
Oct 05, 2017
Is it the file 239q is valid ?
Bil
United States
Oct 05, 2017
239 valid?
collabcerthelp
United States
Oct 05, 2017
for those that have recently taken the exam. Do you remember the main topics covered? I have a lot of experience with CUCM and CUCM express but not so much on the Video side. Trying to figure out wher I should put my focus on studying
Dcy
Brazil
Oct 05, 2017
Not valid! Exam has changed. more then 70% new questions
Dcy
Brazil
Oct 04, 2017
Exam has changed. more then 70% new questions
Squall
Vietnam
Oct 04, 2017
239q still valid. Pass today with 9xx score.
Michael
Canada
Oct 04, 2017
failed today , exam has changed. more then 70% new questions
R.G.
Canada
Oct 04, 2017
you have to purchase the dump File & also the player ( ETE Exam Simulator )? if so, is it not a scam not having a free player as you have already paid the ete ?
Shishir
India
Oct 04, 2017
239q is still valid.Passed today with score 9XX.some new questions are there as mentioned by other guys.Questions are littlebit tricky.
Ninja
Japan
Oct 04, 2017
239q still valid
Smith
France
Oct 03, 2017
239q still valid i have pass 2 hrs ago.Pass with 9XX score. there are 1 or 2 new Question apart of 5 Q which we already knew. i had answer what peter and other mention.
Bill
United States
Oct 03, 2017
Failed today with an 848. Knew the Q239 well. It's still valid, but the new questions are on there for sure.

new questions
1.a trace RTMT logs (location out of resource)
ANS. Out of bandwidth is the only valid answer.

2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,Annunciator (Need to verify)

3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway (Need to verify)

4.Question about normalization global call routing. (normalization is on the Q239)

5. Scenario: CUCM in HQ, remote site iwth end points, Branch with srst+PSTN+DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-branch calls
(I gotta look into this one, but HQ would definitely have issues if they called the remote site. Branch will work on SRST, but would not be able to call the remote site.)
Peter
United Kingdom
Oct 03, 2017
Hi All

Some research on the new q's and wanted to share, mostly agree with David SS but hopefully the explanations below will help everyone out

1.a trace RTMT logs (about no enough bandwidth)
ANS. location out of resources

Explanation
Two potential counters here: OutOfResources, LocationOutOfResources

2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,SIP TRUNK

Explanation

Call Preservation
The call preservation feature of Cisco Unified Communications Manager ensures that an active call does not get interrupted when a Cisco Unified Communications Manager fails or when communication fails between the device and the Cisco Unified Communications Manager that set up the call.

Cisco Unified Communications Manager supports full call preservation for an extended set of Cisco Unified Communications devices. This support includes call preservation between Cisco Unified IP Phones, Media Gateway Control Protocol (MGCP) gateways that support Foreign Exchange Office (FXO) (non-loop-start trunks) and Foreign Exchange Station (FXS) interfaces, and, to a lesser extent, conference bridge, MTP, and transcoding resource devices.

Enable H.323 call preservation by setting the advanced service parameter, Allow Peer to Preserve H.323 Calls, to True.

The following devices and applications support call preservation. If both parties connect through one of the following devices, Cisco Unified Communications Manager maintains call preservation:

Cisco Unified IP Phones
SIP trunks
Software conference bridge
Software MTP
Hardware conference bridge (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Transcoder (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Non-IOS MGCP gateways (Catalyst 6000 24 Port FXS Analog Interface Module, Cisco DT24 , Cisco DE30 , Cisco VG200)
Cisco IOS H.323 gateways (such as Cisco 2800 series, Cisco 3800 series)
Cisco IOS MGCP Gateways (Cisco VG200, Catalyst 4000 Access Gateway Module, Cisco 2620, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3810)
Cisco VG248 Analog Phone Gateway

The following devices and applications do not support call preservation:
Annunciator
H.323 endpoints such as NetMeeting or third-party H.323 endpoints
CTI applications
TAPI applications
JTAPI applications

3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway

Explanation
First Part
1. Navigate to the trunk device > trunk.
2. Select new or existing trunk
3. Check "SRTP Allowed - When this flag is checked, Encrypted TLS needs to be configured in the network to provide end to end security. Failure to do so will expose keys and other information.
Consider Traffic on This Trunk SecureRequired Field"
4. Select either when using bot sRTP or TLS, when using sRTP

Second Part
1. enable
2. configure terminal
3. voice service voip
4. srtp fallback
5. allow-connections sip to h323
6. allow-connections sip to sip
7. end

4.Question about normalization global call routing. 3 answers
not remember options:
I marked 3 about "gateways" word (I failed here, I think)

Not quote enough info here so maybe reading of this will help, most normalization is done by the gateway but some are also available on trunks

More in-depth detail

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fsgd-851-cm/fscallpn.html

5. Scenario: CUCM in HQ, Branch with srst PSTN DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-calls
d) The service phone does not working.
not remember more...

Thinking B from the selection given is correct as SRST provides remote site redundancy during a WAN failure

Heard something about SAF ports

Question about saf port:
a: tcp port 5050

Explanation

The service family used is IPv4 and TCP port 5050. The keepalive timer for the Cisco SAF external client is optionally set to 360,000 milliseconds (the default is 9600 milliseconds).
service-family external-client listen ipv4 505
external-cliet client-a
username safuser
password safpass456
keepalive 360000
thanju36
United Arab Emirates
Oct 03, 2017
Is 239q still valid?
Samir
Saudi Arabia
Oct 03, 2017
Failed today with 826 score using 239Q dump. About 10 new questions, 5 of which mentioned by David and others which I don't remember. Need to get these new Q&As
Jay-z
Bolivia
Oct 03, 2017
Hi guys!! where can I find the 239q Dump???
Elbert
Canada
Oct 02, 2017
239q is valid. Pass with 930 score
Joselito
Spain
Oct 02, 2017
Dump 239q is valid. Pass with 930 score
AC
United States
Oct 02, 2017
Exam Q239 is valid. I took the test on March 18th and scored 9XX/1000.
Only 4-5 questions were new.
Alfonso
Spain
Oct 02, 2017
Hello,

Dump 239 is still valid, scored 9XX , 5 new questions.
IAH
Spain
Oct 02, 2017
Failed today, 7xx, around 8 new questions related with VCS and DSCP
KS
South Korea
Oct 02, 2017
Q239 is still valid (marth 18)
Rob
United States
Oct 01, 2017
Q239 is still valid
sipXpbx
Bulgaria
Oct 01, 2017
Guys, from where you have this 239 Q&A? From dump Files or?
edisson.realpe.i@hotmail.com
Colombia
Oct 01, 2017
from my point of view I look at what has happened, there are new questions 239Q?, thanks
Act Smith
Kenya
Oct 01, 2017
Hi,

I got a score of 860. Passing score was 860. In the print out from the exam center, they have written that I have passed. When one gets 860/860 is that a pass or can Cisco change the exam result to a Fail?
Tahseen Sheir
Qatar
Oct 01, 2017
The 239 Q is still valid , I passed today 9xx/1000 only 3 new questions out of dump
ercan
Turkey
Oct 01, 2017
I have passed the exam today scored 916 with the 239q. It is still valid.
J00E
Jordan
Sep 30, 2017
Passed now 916 two new questions
Samir
Saudi Arabia
Sep 30, 2017
Hi...can anyone confirm whether the questions are coming from dump? or the latest 119Q at passdaily/ AT ?
ahmad
Jordan
Sep 30, 2017
Hello,

Dump 239 is still valid, scored 944 , new questions as mentioned by David SS
Aliens
Australia
Sep 30, 2017
Hi Guys
239q still valid about 6-7 new questions
Michel
Switzerland
Sep 30, 2017
Hello, 239q is still valid, I pass the exam yesterday. As already the others say, Just few more questions.
Ninja
Japan
Sep 30, 2017
>Failed with score 714, passing score was 860 , most of the question are new

I used 102q
ciptv2
Netherlands
Sep 29, 2017
Hi, 239q is still valid, I pass the exam today. Just few more questions (5/6) as people said (the ones told by Yasir and David)
Sam
Romania
Sep 29, 2017
Ninja,

Can you please confirm which dump u've used ?
ahmad
Romania
Sep 29, 2017
Any update if the 239 is valid or not ??
Rinny
India
Sep 29, 2017
I have passed the exam today (9/27/2017) with score of 9XX.Dump q239 is valid one.
NP
Australia
Sep 29, 2017
Hi Ninja,

Which dumps did study?. Is the different from Q239.

Thanks.
Ninja
Japan
Sep 29, 2017
Failed with score 714, passing score was 860 , most of the question are new
Jake
Canada
Sep 28, 2017
The 239Q file is valid. Passed today 9XX/1000
jose
Mexico
Sep 28, 2017
Yes you can skip the 6 new questions and still pass.
David SS
Spain
Sep 28, 2017
944 Score. Passed the exam 2 hour ago (01/03/16) (54 questions; 860 for pass)
The 239Q is still valid. Only 5 new:

VCS Control: 100%
Collaboration Edge: 100%
Configure CUCM Video Service Parameters: 100%
Describe an Implement Centralizaed Call Processing Redundancy: 70%
Describe and Configure a Milti-Site Dial Plan for CUCM: 100%
Implement Call Control Discovery/ILS: 100%
Implement Video Mobility Features: 83%
Implement Banwidth Management and CAC on CUCM: 100%

new questions
1.a trace RTMT logs (about no enough bandwidth)
ANS. location out of resource

2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,Annunciator

3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway

4.Question about normalization global call routing. 3 answers
not remember options:
I marked 3 about "gateways" word (I failed here, I think)

5. Scenario: CUCM in HQ, Branch with srst+PSTN+DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-calls
d) The service phone does not working.
not remember more...
Yury
Unknown country
Sep 28, 2017
Hi, I have failed the exam with 804 points. Only 10-15 questions from free dumps, all other questions are new.
CSCO_911
Saudi Arabia
Sep 28, 2017
Please let us know if We can still appear for the exam. Yet I can skip 6 new questions..?
jose
Mexico
Sep 28, 2017
239Q valid as of 2/29
Alfonso
Unknown country
Sep 27, 2017
Please, update the new 6 questions
CSCO_911
Unknown country
Sep 27, 2017
Come On..!

How can Cisco not displayed on Practice..?
Burger
Canada
Sep 27, 2017
Hi, Anyone took the exam the last week, can clarity, or at least give a clue about the the now questions. Thank you!
Aspirant
Qatar
Sep 27, 2017
Hi There,

Did anyone passed after 25-Sep-2017?

Please Update. Thanks...
yasir
United States
Sep 27, 2017
239 dump still valid as 02/26/16
new questions
1.a failed call and RTMT logs (no enough band width)
ANS. i believe it is location out of resource
2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone

ANS. Software conference bridge,ip phone,Annunciator

3.call preserve on cisco unified SRST on sip phone
SIP Trunk, i don't remember read about it.
Dav
Unknown country
Sep 27, 2017
Hello all ! Passed the exam 1 hour ago (26/02/16) and the 239Q is still valid... My score 9XX... Only a few new questions (4-5). So if you work the 239Q it will be OK !
Rahul
India
Sep 26, 2017
The Dump 239q is Valid!!!. Took Exam on 26/2, 4-5 new question have been added with respect to SRST,Call routing etc. they are easy to be answered if you know the concept well.

Thanks,
Rahul
llkl
Saudi Arabia
Sep 26, 2017
guys,
Any update if the exam is still valid?
Thanks.
herman
Indonesia
Sep 26, 2017
239Q still valid, 2/23..
Francisco
Chile
Sep 26, 2017
Can you explain the new question please.
karim
Egypt
Sep 26, 2017
congratulations for all who passed 300-075 exam using 239q, i would like to ask you to tell us about new questions you found in the exam and how many?
Rahul
India
Sep 25, 2017
IS 239 dump still valid ?
Gunner
United States
Sep 25, 2017
239q is still valid on 2/23. 6 out of 54 questions are not from dumps. But if you master the 239q, you will pass.
Hanna
Colombia
Sep 25, 2017
239q still valid, passed yesterday with 94X.
moath
Kuwait
Sep 25, 2017
passed already with 902 Score
Joe Mama
United States
Sep 25, 2017
239q still valid 2/19. 5 New questions...easy if you know your shit
Joff
United States
Sep 25, 2017
Hi, did anyone use the last pass4sure dump with 173 questions, from Jan 30, and pass? Just want to know if the file is valid. Thanks.
Passed
Switzerland
Sep 24, 2017
Hi, I passed 9xx the 11th of February. The 239 is valid!
Stan
Canada
Sep 24, 2017
Thank you Alex, and Llkl from SA, i wish you 9xx mark, inshaAllah you will be fine, shahad wa astaghfar before you start answering the test questions.
LPG
Portugal
Sep 24, 2017
@Costa can you give an email to contact you
Belmont73
Switzerland
Sep 24, 2017
Just passed today. 239q is valid
Fireman
Bolivia
Sep 24, 2017
I have passed 13/feb with score 89X
with the 239Q dump as it is valid.
prasanth
United Kingdom
Sep 24, 2017
Passed today. 100% valid
younmahon
Morocco
Sep 24, 2017
239q dumps still valid, passed today with 972
Costa
Brazil
Sep 23, 2017
I got in the last Friday with 958 pts. All questions are there.
Eisenheim
Bolivia
Sep 23, 2017
I pass to day. 88X/1000
dump file q239 steel valid.
PUTAX
France
Sep 23, 2017
Passed this week with 9XX
All questions are in the 239q dump.
Few Q&A are inconsistent, I strongly recommend to comment (there is a button to add a comment in the test)
Some exhibits are not displayed correctly (exactly as shown in the 239q dump : you cannot read the ip addresses on the picture). I recommend to do a request to the local VUE-PROMETRICS representative as a technical incident, and add a comment in the exam.
If you fails, please open a case to Cisco www.cisco.com/go/certsupport and ask for your exam to be reviewed: you may be refunded with a voucher
Collab2016
Brazil
Sep 23, 2017
PASS! 11 Feb Score 930. 239Q dump file is valid!
llkl
Saudi Arabia
Sep 23, 2017
Stan from Canada...I have seen the file has updated questions and would only be able to update you on/after saturday 21-Sep-2017. Wish me Luck.
sipbx
Bulgaria
Sep 22, 2017
Guys, from where you have this 239Q dump?
Alex
United Kingdom
Sep 22, 2017
To STAN
The file 239q is 100% VALID, I passed the exam few days ago, all questions been from dump !
Jake
Canada
Sep 22, 2017
Anyone pass with Actual Tests 239Q dump?
jody
Unknown country
Sep 22, 2017
pass 9xx. 239 still valid. Lot of Stuff in Gatekeeper, Video, SAF, VCS, MOB Device, Extension MOB
Alex
United Kingdom
Sep 22, 2017
Passed today 916 score. the file is 100% valid ! ALL questions was from this file.
PML
Sri Lanka
Sep 22, 2017
Hi All,
CCNP Collab now !
239Q 110% vaild.
Stan
Canada
Sep 21, 2017
Not may people commenting 300-075, either it's a catastrophic failure rate, or some thing else, can anybody shade light on this, because i am we really confused by contradicting testimonials saying the 239Q dump is some how valid, but how much percentage the claimed validity is? you can get in touch with me (sunos55 HM c o m).

My self have already failed this test twice , so please, anyone passed the exam recently that can clarify this ambiguity?
Karim
Australia
Sep 21, 2017
dump file valid passed on 6 feb
Vitaliy
Unknown country
Sep 21, 2017
I have passed today with score 916
with the 239Q dump as it is valid
Vitaliy
Unknown country
Sep 21, 2017
I have passed today with score 916
with the 239Q dump as it is valid
curiosgeorge
United States
Sep 21, 2017
Where is this 239Q that people are using? Is it the file that I don't have access to?
Said
Canada
Sep 21, 2017
Anybody passed the (300-075) exam, and which dump is valid? Thank you in advance.
Collab2016
Brazil
Sep 20, 2017
Guys,

The 239Q dump this really valid?
Alex
United Kingdom
Sep 20, 2017
Is it the file 239q is valid ?
300-075
Unknown country
Sep 20, 2017
I passed but Dump is not valid.
Collab2016
Brazil
Sep 20, 2017
@Costa, please which dump you used to take the exam?

@Everyone, anyone confirm if this new dump ETE File this valid?

Thanks for advance
Ahmed Shouka
Germany
Sep 20, 2017
I have passed today with score 9XX
with the 239Q dump as it is valid
Bill
United States
Sep 20, 2017
PML, where is the 239Q available?
Ahmed Shouka
Germany
Sep 19, 2017
Guys,

there is a new 239Q dump that people says it is valid, two persons advised that they have passed using it
TJ
Jordan
Sep 19, 2017
guys, P4S has this one 173 Questions - Last Update: Sep 17, 2017

anyone tried it ?
Peter
United Kingdom
Sep 19, 2017
The Q277 is also not completely valid.

Some Questions remembered, not on Q277 or any other:

SAF Port number usage : UDP 5050

Fill in the blank on intercluster and intracluster : not sure of the answer

Anyone know others missing?
Prem
India
Sep 19, 2017
I am using 180Q(Actual tests) and 114qfrom certmagic. Do the Questions from these dumps appear in the recent exams? if not please share the valid dumps to my email id. Thanks. I am planning for 2nd week of Feb16.
dimka2000
United States
Sep 19, 2017
new 239q dump is valid. Just passed 300-075 with 958 score :)
most of the exam questions are from ~180-239 range
raptor
Peru
Sep 19, 2017
Costa, what dumps do you used??
iecollab
Brazil
Sep 18, 2017
Costa any mail to ask you to validate a file?
Agrolate
Chile
Sep 18, 2017
I have a 239q dump. someone who has recently given proof could validate whether this is valid?

Pfff
Switzerland
Sep 18, 2017
Guys, I have a good and a bad news to share.
Bad news first: I failed today (28/01/) working with the 109q + 114q + 180q.
Good news now: As far as I can remember a significant part of the missing questions are in the 239q that I got too late. It looks like the dump version has just been updated.
Please share you experience, I need to pass it before mid-February.
Costa
Brazil
Sep 18, 2017
I tried today and didn't passed. I got 815 points.
Eman
Singapore
Sep 18, 2017
Is the 277 Q?
Paul
Australia
Sep 18, 2017
Guys, I barely passed the exam the other day after x number of retakes. Relying on 114q alone will get you around 60 - 70%, then I relied on my memory and researched my way about the tough questions that I could remember.
liolioe
Brazil
Sep 18, 2017
exam does anyone know if that site has valid dump or not? it has an update date from 23 jan someone that take the exam could at least verified the sample they gave?
Aspirant
Russian Federation
Sep 17, 2017
We feel sorry for your unsuccessful attempt. But, could you please let us know if any real exam questions that you had attempted... are from any of these dumps..?
Could you tell us yr experience.
kimberlyc
United States
Sep 17, 2017
Is the file valid?
Ekrem
Turkey
Sep 17, 2017
Hi i take this exam on 15.09.2017 114Q 146q 277Q 180q 96q all is totally invalid .
Aspirant
Romania
Sep 17, 2017
Come On Guys... there's not even a single real expert to be able to PASS this exam..?

This is quite weird/embarrassing to know that out of the Whole WORLD not even a Single Collaboration Expert is a REAL WORTHY COLLAB ENGINEER..? including me (^_^)

Even Dump Experts are helpless...to be able to help us. Anyways Guys give it a try and keep us posted. Thanks...

- Cisco Aspirant.
Aspirant
Luxembourg
Sep 17, 2017
Has anyone able to PASS this exam after 15-Sep-2017..?

Please share your opinions/Experiences/Suggestions with me. Thanks...
- Aspirant.
dimka2000
United States
Sep 16, 2017
Ahmed, what do you mean by 'newest' 180q? Can you provide it, so I could review?
Massimo - Italy
United Kingdom
Sep 16, 2017
Guys no action , all dumps are not valid to pass the exam, there are a lot of new questions.
Ahmed Shouka
United Kingdom
Sep 16, 2017
Guys, we want to know if there is a valid dump for the current exam or not
I need to enter the exam and schedule for a test but i need to make sure if there is a valid dump or not
There was the 114Q and 277Q then a new 180Q then the newest 180Q
Are all of them invalid or what is our next action here ?
PML
Sri Lanka
Sep 16, 2017
HI All,

I have the 180 Q with me.Just wanted to check whether it is accurate.Anyone sitting for the exam before feb 2nd ?
Andi
Indonesia
Sep 16, 2017
Does anybody have a valid dump for this exam?
Bill
United States
Sep 16, 2017
Can you post the 180 Question test. I would like to see if it has the questions I missed on there.
Zahackn
Germany
Sep 15, 2017
I have the 180q question file. I will upload it soon. But there are only two or three video questions.
But I think in combination with the 102q file it will be possible to pass the exam. Mine is on February the 2nd.
dimka2000
United States
Sep 15, 2017
I took a test on 12/18 (looks like only days after they changed the questions), I used ActualTest CIPTV2 v1.0, Version 5.0 (114q) and it had maybe about 15-20 questions matching.
I looked at the 'newer' 180q dump, and even though it says version 6.0, it looks like it is even worse than the older dump I used. It looks like it has a bunch of much older questions from CIPT2 exam.
If you look at the exam topics, 25% of the questions are about VCS-C and VCS-E, and also integration between VCS and CUCM. Your dump does not seem to have those questions.
Also, the several big scenario questions (presenting mixed VCS/CUCM environment) with multiple sub-questions from 114q ActualTest were still on the exam, while 180q dump does not have them.

I'd say this is a bad/outdated dump and will not work for the 300-075
Aspirant
United States
Sep 15, 2017
Can someone confirm that these dumps are enough to PASS and CLEAR the exam...?
dimka2000
United States
Sep 15, 2017
Agrolate or anyone else who has a newer test dump: I took the test recently (after they changed the questions). If you share your the dump you have with me, I'll be able to tell you if this contains new questions.
dimka2000 at yahoo.com
Lika
Brazil
Sep 15, 2017
Agrolate, post your 180 q
Agrolate
Chile
Sep 15, 2017
I have an actualtest of 180q, someone use this? is valid?
Abhijit
India
Sep 14, 2017
I gave 300-075 . I had 52 question in exam.

I failed by 744 score
Logmos
Unknown country
Sep 14, 2017
Hi everyone,

The dumps Ace.102q, Ahmad.91q and Virgil.93q are totally invalid.
I made this test at 09/12/2017 and I didn't pass it, 736 on 860. About 30-35 new questions of 52 questions appear in this test. Only the questions about configurations (DX950, jabber, and so on) are still maintained.
Please, if anyone has the new dump and can send to us as soon as possible, I appreciate a lot.

Thanks.
Aspirant
Saudi Arabia
Sep 14, 2017
Hi,

Can someone please update us with the exam status. Are these dumps still Valid..?

Please Reply. Thanks...
Ahmed Shouka
Germany
Sep 14, 2017
Is there any valid dump after the 114Q as people confirmed it is not valid anymore
Umair
Germany
Sep 14, 2017
No its not valid! i failed
Personal
Unknown country
Sep 14, 2017
Hi.
Please, can you send me the new dump in pdf?
mailpersonal999 at gmail.com

Thank you!
Cisco Aspirant
Saudi Arabia
Sep 13, 2017
Hey Guys!
Can anyone confirm if it still valid? Please update us all. Thanks...
William
United States
Sep 13, 2017
Umair, did you take the test yet with the 114q? Does anyone know if the test changed on 09/11/2017 specifically?
Peter
United Kingdom
Sep 13, 2017
Hi

The 114 and 146 Q&A are not valid many new questions (10-15) present on the exam
Peter Owens
United Kingdom
Sep 13, 2017
The exam 146 is not valid many new questions on exam. Advise anyone to wait for new questions before booking
student
United States
Sep 13, 2017
exam has changed. lots of new questions. :(
Umair
Germany
Sep 12, 2017
Hi,

next week i am going to attempt this exam using 114Q file! can any one confirm is it still valid? did any one pass this exam after 15th Dec?
Massimo
Unknown country
Sep 12, 2017
Yesterday take the exam without success, the question are different and a lot of new questions
Armi
Unknown country
Sep 12, 2017
Do somebody have book Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Foundation Learning Guide (CCNP Collaboration Exam 300-075 CIPTV2), 3rd Edition in pdf?
moath
Unknown country
Sep 12, 2017
this is valid dump ??
please some buddy share with as the 114 Q
muath53@gmail.com
Nixon
Saudi Arabia
Sep 12, 2017
Good day Guys, could somebody send me a pdf of 300-075?: Nixon_Mantua@yahoo.com

Thanks
karim
Egypt
Sep 12, 2017
the dump is NOT valid the exam changed, about 15 questions only i scored 780 do not take the exam now
ed
Unknown country
Sep 12, 2017
this is not valid anymore, some questions are there but most are not.... i failed :(
Rick
Spain
Sep 11, 2017
Hi Bill, Could you share the new questions?
paul
Unknown country
Sep 11, 2017
used 114q but still failed last Dec 12 :-(
paul
Australia
Sep 11, 2017
used 114q but failed the exam yesterday
karim
Egypt
Sep 11, 2017
please guys did anyone take ciptv2 exam recently? to what percent is 114q valid?
what are the new questions if any?
my exam is tomorrow.
Shaheen
United Arab Emirates
Sep 11, 2017
Given 300-075 exam today at Dubai today and failed. More then 15 new questions out of 52q ( 114q dumps) . Scored 770. But 860 needed to pass.
Bill
United States
Sep 11, 2017
That is correct. I did the 114. I see lots of other sites talking about a 144 dump that is out but not able to locate it.
paul
Australia
Sep 10, 2017
Used the 114q but failed the exam yesterday :-(.
Rinny
United States
Sep 10, 2017
Can any one confirm dump is valid or not.I am planning to take exam this week.
John
United States
Sep 10, 2017
Bill, did you study the 114q?
Bill
United States
Sep 10, 2017
Not valid for the exam i took. maybe 15 out of the 60 questions were in the dump.
karim
Egypt
Sep 10, 2017
what are the new questions in the exam that are not in the dump?
Caetthy
India
Sep 09, 2017
I have exam tomorrow, please share recent pdf file on scorpiansaru@gmail.com
Pablo
Unknown country
Sep 09, 2017
114Q Still valid, I passed today 9xx -Argentina-
Juan
Unknown country
Sep 09, 2017
Do you know the passing score of this exam? Thank you!
Bahaa El-deen
Egypt
Sep 09, 2017
Still valied passed yesterday 1-December score 938
John
United States
Sep 09, 2017
I need the PDF, sorry, forgot to mention.

bcchimp@gmail.com
kreemo
Saudi Arabia
Sep 09, 2017
114 Q&A is valid, i was passed in Wednesday
Monsour
Saudi Arabia
Sep 08, 2017
Hi, please who have the valid dump share it with us my email : monsour_888@homtail.com.
Thank you in advance
Pawel Z.
Germany
Sep 08, 2017
Can you please send me latest dump at pawelzaw71@o2.pl
Thanks a lot
amr rashad
Egypt
Sep 08, 2017
Kindly send me the dump
eng_amr_rashad@hotmail.com
Salvatore
Italy
Sep 08, 2017
Italy, Passed today with 917 , still valid. I used Virgil and Ace. Good luck
Mauricio
United States
Sep 08, 2017
can you please send me latest dump at maubolira@gmail.com. thank you
Bisharat Akram
Netherlands
Sep 08, 2017
Passed today with 969 marks.114q file is still valid.god luck guys
gugalde
Mexico
Sep 08, 2017
could some one share last 114 Q&A Dump?. I appreciate a lot.

Regards. and have a greate day.
ch33f
Germany
Sep 07, 2017
Could you please send me lateset dumps at piranhabox@gmail.com
SISCO
India
Sep 07, 2017
hi pls share latest dump.
TingcoMic
Hong Kong
Sep 07, 2017
Hi Dementium / Mayuresh,

Can you please send me the latest dumps also at
tingcomic3@hotmail.com

Many Thanks !!
alshim
United Arab Emirates
Sep 07, 2017
Hi Mayuresh

Please share it to me also alshimmanzil@gmail.com
Dementium
Mexico
Sep 07, 2017
@Mayuresh i sent you the dump, please share it
mayuresh
India
Sep 06, 2017
Hi Dementium,

can you please send me lateset dumps at mayuresh.phade@capgemini.com
my CCNP voice cert is expiring in next week ,need to give CIPTV2 in this week..please help me.
mayuresh
India
Sep 06, 2017
Hi Dementium,
can you pls send me updated dumps at mayuresh.phade@capgemini.com
My expiratory of ccnp voice cert is due in next week I desperately need to give CIPTV2 as soon as possible.Please help me
amr rashad
Egypt
Sep 06, 2017
kindly send me the latest dump
Dementium
Mexico
Sep 06, 2017
114 Q&A is valid, i passed today
Juan
Colombia
Sep 06, 2017
I passed yesterday oct 30, the dump 114 is valid.
Ahmad Kefaya
Saudi Arabia
Sep 06, 2017
dump is valid ??
* Please post your comments about 300-075 Cisco Exam. Don't share your email address asking for Cisco 300-075 dumps or 300-075 pdf files.

Add Comments