Pass Your Cisco 300-075 Exam Easy!

Is this dump Valid. Is anyone took exam with this dump

@ 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.

@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

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.

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

first test 673 and second test 855. both failed but gettinh better

@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

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

@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.

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?

@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

@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?

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?

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 ?

Cisco.ActualTests.300-075.v2015-10-28.by.Ace.102q.vce
is fine to pass the exam?

Passed today with a decent 900. Q161 is still valid but need to study all questions and find out answers. Thanks everyone.

@ Sunday

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

thank you!

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

@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

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

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

@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

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!

@Sunday

Thanks for clarification. I will try to exam shortly...

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.

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

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

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

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

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

@Sunday by chance are you finished with that guide?

@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..

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?

CIPTV2 (300-075) is still valid ?

Which is the .vce valid?

@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!

@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?

@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!

@John, the answer is
C. The user can log in to only one device at a time.

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?

@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.

@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!

@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?

Is the 161q file spoken of here the "premium" 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.

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?

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 this 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!

QUESTION NO: 147
Should be A and F.

@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.

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 ??

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

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.

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.

@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)

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.

@king

Can you advise to what question #'s you are referring to in your reply back to me?

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!

@Fe

Could you please share your thought on the exam. is 161Q still valid.

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.

@Fen,

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

Thanks

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

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?

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

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 "

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

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

So is the 161Q&A on the premium files valid or not? Please letteth me know.

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.

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.

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

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

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?

@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

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?

@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

@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.

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?

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?

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?

@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.

@ 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

@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)

@ 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

@ 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

161 dump is valid but many answers in the premium dump 161 are wrong.

Failed today with 723 :(

@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

@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

@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

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.

Hello, anyone passes this lately?

@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.

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.

@aungmyozaw, Do you have any news?

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?

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?

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?

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?

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?

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?

@Fen, Thank you so much for your answers and explanation . I just luckily passed the exam with 869 marks.

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

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)

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?

@Fen, Thank you so much for your answers and explanation. I just passed the exam with 869 marks.

@ 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

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.

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.

@king , Thanks for answering . Today i will sit the exam and will post the result.

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.

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

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).

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

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.

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!

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

@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.

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?

@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

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?

@Sheraz,Is there any link for the answers?

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

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.

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

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).

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...

Where can I get the q161 version of test? please help.

@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

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.

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

using dump premium 161, have fact checked all answers, some are blatantly wrong, doing exam today, will post results

@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.

@Juan,

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

Thanks
Sultan Al Arif

@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

@Juan: Did you make notes of correct answers in the dump?

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.

@Marc,

What is your plan for retake?

I have seen many references to q161 version of test, but can not find this dump. Also, has anyone had luck with this dump?

@xbr I agree. I was just pointing out that the premium files answers are incorrect. My answers pasted here are incorrect. @Sunday, you are corr my mistake. AF41 (Dscp 34) is for multimedia conferencing

Failed last week. Five or more new questions. Some answers in dump 161 were wrong

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

what study material, book, student guide to get well prepared.

Please.

@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

@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.

@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?

@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.

@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.

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.

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?

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

Thanks John/ Sunday for your responses

King: good luck keep us posted :)

@king good luck, let us know if you pass it and if dump is reliable!!
Thank you!

@ King: Wish you good luck. please keep us posted. and what dum file you used

Booked for 21st May and Now I am afraid. Any last hour tips guys.

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

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

Ramesh, the correct answer is:
D. The traversal zone on the VCS Control does not have a search rule configured.

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

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

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.

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?

Yep, I aslo failed. 161 is valid.

Any new news about the 161q ?

Failed today with a 726. ALL NEW QUESTIONS. 239 is no longer valid. 161 is valid, but many of the answers in the Premium file are wrong.

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.

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.

hi guys, is dump 161q valid?

I am failed today
Used dump from itexamfox Q355

I'm failed today. Use Passlear Q355

Any one took the exam recently ? is 161 questions valid ?

Hi guys,

Any body has information about updated Dump 160?
any body try it?

Anyone tried with 161q pdf version?

The premium dump 161, is valid?

Does anyone tried 161 Questions

b, e, f

@aungmyozaw: The answer is AEF.
Does someone tried the new 161Q, is it valid?

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

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!

Hello, There is a new dump on the internet with 161 questions seems to be released last week of April. Anyone tried this?

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

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

Failed today.

Any update about the exam

Failed. 60-70% questions are different to these dumps. Not valid.

Did anyone passed after 15-Apr-2016

any new updated dumps? the current exam is not valid anymore

Failed today using 308. Mostly VCS/Expressway.

I didn't pass with the 100 question exam, today.

The newest (ver. 16.021) dump from itexamfox with 308Q is NOT valid. I failed. There are too many new questions.

people are saying that 308q is not valid either

@Zulus, where did you find the exam, from which site?

There's a dump 308 Qestions in the web, from 12. April.
Did anyone tryed this one?

Found a new dump with 308Q from 12.04.2016.
Anyone tryed this one?

Test has changed.

Ikkir, Frank, Bill and TED, pls did you use the 239 premium file?

Are you use the 239q and failed???

Failed!! 90% of the questions are new and mostly related to video technology. Hoping to have the updated questions soon.

239q ^valid (read in regex) anymore. Mainly covered VCS and Expressway.

Agree with Frank. Test is 90% different. Alot of Video/SAF/CCD/Location type questions.

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.

Exam changed completely. Most of the question were video related

Is it the premium file 239q is valid ?