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Exam: | 300-075 - Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) |
Size: | 6.2 MB |
Posted: | Thursday, September 7, 2017 |
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Sorry. Yes I got this question in exam and my answer was A & D.
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
ANSWER: C and E. EXPLANATION: You cannot have A and C on at the same time, and E has to be on for H323 to work at all.
By default, the VCS acts as a SIP-H.323 and H.323-SIP gateway but only if at least one of the endpoints that are involved in the call is locally registered. You can change this setting so that the VCS will act as a SIP-H.323 gateway regardless of whether the endpoints involved are locally registered. You also have the option to disable interworking completely.
The options for the H.323 <-> SIP interworking mode are:
Off: the VCS will not act as a SIP-H.323 gateway.
Registered only: the VCS will act as a SIP-H.323 gateway but only if at least one of the endpoints is locally registered.
On: the VCS will act as a SIP-H.323 gateway regardless of whether the endpoints are locally registered.
You are recommended to leave this setting as Registered only (where calls are interworked only if at least one of the endpoints is locally registered). Unless your network is correctly configured, setting it to On (where all calls can be interworked) may result in unnecessary interworking, for example where a call between two H.323 endpoints is made over SIP, or vice versa.
Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What
value should be entered into the gatekeeper to support this bandwidth?
Cisco 300-075 Exam
20
A. 768 kbps
B. 384 kbps
C. 512 kbps
D. 192 kbps
Answer: B
Explanation:
Incorrect answer: A, C, D
A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This
sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video
rate increases to maintain a total bandwidth of 384 kb/s.
Link:
http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.ht
ml#wp1059726
===I think the corect is Answer A
http://flylib.com/books/en/2.110.1.217/1/
@Hari could you share the dump to me [email protected]
According to me the answer is
A&D
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
I am also thinking that answer A and C
Interworking On and Registered only gives ability to interconnect SIP H323
In case when we select H323, what it gives according to SIP? Please correct this or may be some documentation
2. Can anybody explain this question what do you think, A or D?
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured
3. In exam which I failed week ago I had qestion
Which solution is needed to enable presence and extension monility to branch office phones during a WAN failure?
A. SRST without MGCP fallback
B. SRST with VOIP dial peers to CME
C. SRST with MGCP fallback
D. CUCM Express in SRST mode
But in answers I remember BE6000 was there
and two another answers but also there CME in SRST mode
a litl bit confusing if we will have BE6000 in the branch it will be also mobility and presence
did anybody saw this kind of answers in this question?
4. Which statement about when user A calls user С using SIP are true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa
can any body also share your opinion? many different answers
5. An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?
Everybody says SCCP, but I saw in docs
________________________________________
Restrictions for SCCP Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways
Directed call park and directed call pickup are not supported for Cisco Unified Communications Manager
_______________________________________
MGCP-controlled gateways do not require a media termination point (MTP) to enable supplementary services such as hold, transfer, call pickup, and call park.
________________________________________
any opinion?
6. To how many nodes 'SCCP' phones can connects 1 or 2???
7. I dont remember exact question, If we have on ExpresswayE one NIC what we have to set
FQDN or IP?
Some NEW QUESTIONS
1. -
Which two statements regarding the Cisco VCS search and transformation process are true? (Choose two.)
A. The Cisco VCS applies the search rules in priority order (all rules with priority 255 are processed first,
then rules with priority 254, and so on).
B. Transforms do not use priority numbers.
C. Presearch transforms are applied before call policy is configured and before user policy is applied.
D. Presearch transforms are applied after call policy is configured but before user policy is applied.
E. The Cisco VCS applies the search rules in priority order (all rules with priority 1 are processed first, then
priority 2, and so on).
F. You cannot set up search rules according to the protocol SIP or H.323.
Answer: C,E
2. -
Which feature enables users to manage business calls by using one phone number to pick up inprogress
calls on either their desk phone or their mobile phone?
A. desktop call pickup
B. send call to mobile phone
C. mobile connect
D. mobile voice access
E. access list
Answer: A
3. -
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When configuring a secure SIP trunk, to which Cisco Unified Communications Manager trust store must
you upload the Cisco VCS certificate?
A. CallManager-trust
B. ipsec-trust
C. tomcat-trust
D. TVS-trust
Answer: A
4. -
Which DSCP value and PHB equivalent are the default for audio calls?
A. 34 and AF41
B. 32 and AF41
C. 32 and CS4
D. 46 and EF
Answer: D
5. -
For inbound calls that use SIP gateways to cisco unified communications manager, which two options are
available when configuring the format for the calling number type? (Choose
two)
Lead to pass your exam quickly and easily. First Test, First Pass! - visit - http://www.certleader.com
A. Unknown
B. Subscriber
C. Long Distance
D. Default
E. SIP
Answer: B,C
6. -
In which two locations can you verify that a phone has a standby Cisco Unified communications manager?
(Choose two)
A. phone webpage
B. RTMT
C. Cisco Unified Serviceability
D. phone menu
Answer: A,D
7.-
Refer to the exhibit. An engineer is settings up a new deployment and wants to use the Cisco VCS Control
as a gateway for SIP and H.323 endpoints. Which Cisco VCS configuration step must be performed to
allow onset and offset calling?
A. Set the H.323mode On
B. Set the IP mode On.
C. Set the Gatekeeper Auto Discover mode On
D. Set the Interworking mode On.
8. -
Which two items must you configure in Cisco Unified Communications Manager to deploy Cisco SAF?
(Choose two)
A. an MWI
B. voicemail ports
C. a security profile
D. a forwarder
E. a remote destination profile
Lead to pass your exam quickly and easily. First Test, First Pass! - visit - http://www.certleader.com
Answer: C,D
9. -
For which two reasons should you mark AF41 as the audio and video channels of a video call?
A. to allow high-definition quality calls over low-speed links
B. to give video calls a higher priority than AF41 in the QoS policy
C. to provide separate classes for audio-only calls and video calls
D. to prioritize video over other high-priority traffic classes
E. to preserve lip synchronization between the audio and video channels
10. -
When troubleshooting high CPU utilization within the Command Line Interface (CLI) of a Cisco Unified
Communications Manager (CUCM). Which command will show the process CPU usage for all processes?
A. show network status
B. show pert query counter Process"% CPU Time"
C. show pert query counter Partition"% Used"
D. show pert query counter Process "Process Status"
Answer: B
11. -
Where in the Cisco Unified Communications Manager Administration GUI must an engineer navigate to
configure Cisco InterCluster Lookup Service authentication in communication manager?
A. Advanced Features> ILS Configuration> Roles
B. Call Routing > Intercluster Directory URI > Intercluster Directory URI configuration
C. Call Routing > Intercluster Directory URI
D. Advanced Features > ILS Configuration
Answer: A
12. -
How is the peer address configured when ExpresswayE has only one NIC enabled and is using static NAT
mode?
A. Expressway-E DHCP
B. Cisco Unified Communications Manager DHCP
C. Expressway-E FQDN
D. Cisco Unified Communications Manager FQDN
Answer: C
Expressway-C, and Expressway-E, an engineer sees this output in the Expressway-E logs.
Event=”Authentication Failed” Service=”SIP” Src-ip=”10.50.2.1”
Src-port=”25723” Detail=”Incorrect authentication credential for user”
Protocol “TLS” Method=”OPTIONS” Level=”1”
What is the cause of this issue?
A.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Zone username/password.
B.
The Expressway-C Traversal Client username/password do not match the Expressway-E
Traversal Server username/password.
C.
The Expressway-C Traversal Client Zone username/password do not match the Expressway-E
Traversal Zone username/password.
D.
The Expressway-C Traversal Zone username/password do not match the Expressway-E
TraversalClient username/password.
E.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Client username/password.
Answer is B (BE CAREFUL AS THEY MADE MORE ANSWERS SIMILAR TO THIS like Client zone and server zone, DONT SELECT THIS)
I don’t understand because you configure the u/p in zones
I am going to have to say C
What kind of new questions have you seen?
Could you describe them ?
You got Q147 in your exam?
What was you CAC questions if u remeber?? everyone i see get low marks on CAC
Could someone explain why this has to go with Option A
QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco UNITED CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco Unified CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.
Answer: A
Why people think the answer is A or C? any specifics
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
Is it B or C? If so why? any particular reason...
Thanks.
[email protected].
Hello Everyone,
I wanted to reach out to the users who have passed. I failed again today with a 845, and have questions about the below. There were 2 new questions that I put at the bottom of my list. I would GREATLY APPRECIATE ANY INSIGHT ON THESE QUESTIONS AS I AM GOING TO TAKE THE EXAM AGAIN NEXT WEEK....
Q.109
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A.G.711 requires 128K of bandwidth per call.
B.G.729 requires 24K of bandwidth per call.
C.The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D.To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
Answer I selected: B
QUESTION NO: 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video
endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A.The traversal zone on the VCS Control does not have a search rule configured.
B.The access control list on the VCS Control must be updated with the IP for the external users.
C.When a traversal zone is set up on VCS Control only outbound calls are possible.
D.The local zone on the VCS Control does not have a search rule configured.
Answer I selected: D
I have seen some users select A
Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio codec; E. T1 PRI
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder -LTI Transcoder Resource
E.DSP resources -DSP PVDM Resource
F.a T1 card
Answer: D,E
QUESTION NO: 123
Refer to the exhibit.
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versa.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.
Answers I selected: A,E
QUESTION NO: 132
An engineer is performing an international multisite deployment and wants to create an effective backup method to access TEHO destinations in case the call limit triggers. Which technology
provides this ability?
A.AAR
B.CFUR
C.LRG
D.SRST
Answers I selected: C.LRG
I have seen several users answer A.AAR
____QUESTION 134____
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
ANSWER: D,E
Q.135
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?
A.when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B.when you require encrypted calls to endpoints on your corporate LAN
C.when you want to enable calls to web applications by using HTTP
D.when you require administrative access to the Cisco Expressway Edge from the Internet
Answer:D
I have seen "A" as answer by many on this community
QUESTION NO: 146 in Exam options are reduced to choose 2
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish this configuration? (Choose four.)
A.Configure SIP route patterns.
B.Configure the directory URI partition and calling search space.
C.Associate the directory URIs to directory numbers.
D.Activate the URI service in Cisco Unified Serviceability.
E.Configure SIP trunk.
F.Assign directory URIs to users.
G.Configure the SIP profile.
H.Configure the URI service parameters.
Answers I chose: F,G
Should it be the SIP Trunk or SIP Profile??
URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html
Congrats on passing this tough exam.
Can you please help for the following questions answers?
109,116,120,123,132,136,146,147,149,153,156,160
Although some people has given answers for above but would appreciate if you can share what you have chosen. It will be great.
Many Thanks!
Thanks
file has all the questions. See what people have posted in here as to what the correct answers should be.
@Sarah was correct on the topics to study on to help pass the test.
Below are some of the questions I had on my test.
When you configure a globalized dial plan, in which three ways can you enable ingress
gateways to process calls? (Choose three.)
• A. Configure the called-party transformation settings for incoming calls on H.323 gateways.
• B. Configure translation patterns in the partitions used by the gateway calling search space.
• C. Configure SIP trunks between Cisco Unified Communications Manager clusters.
• D. Configure a remote site device pool.
• E. Configure a hunt group.
• F. Configure the gateway with prefix digits to add necessary country and region codes.
Which three configuration settings are included in a default region configuration? (Choose
three.)
• A. Immersive Bandwidth
• B. Video Call Bandwidth
• C. Audio Codec
• D. Link Loss Type
• E. Real Time Protocol
• F. Location Description
Which statement about the SAF Client Control is correct?
• A. The SAF Client Control is a configurable inherent component of Cisco Unified Communications Manager.
• B. The SAF Client Control is a non-configurable inherent component of Cisco Unified Communications Manager.
• C. The SAF Client Control is a non-configurable inherent component of the Cisco IOS Routers.
• D. The SAF Client Control is a configurable inherent component of the Cisco IOS Routers.
Which three tests can you perform to verify redundancy in the customer environment?
(Choose three.)
• A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
• B. Verify that all phones are registered to a second subscriber server.
• C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
• D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
• E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
• F. Verify that the H.323 redundant connection is active.
In Cisco Unified Communications Manager, where do you configure the default bit rate for
audio and video devices?
• A. Enterprise Parameters
• B. Region under Region Information
• C. Cisco CallManager service under Service Parameter Configuration
• D. Enterprise Phone Configuration
An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?
• A.
• B. 323 gateway
• C. SCCP gateway
• D.
• E. 225 trunk
• F. MGCP gateway
• G. SIP trunk
What is the standard Layer 3 DSCP media packet value that should be set for Cisco
TelePresence endpoints?
• A. CS3 (24)
• B. EF (46)
• C. AF41 (34)
• D. CS4 (32)
Where can you change the clusterwide DSCP setting for Cisco Unified Communications
Manager?
• A. enterprise parameters
• B. service parameters
• C. enterprise phone configuration
• D. Ethernet configuration
Company X has a Cisco Unified Communications Manager cluster and a VCS Control
server with video endpoints registered on both systems. Users find that video endpoints
registered on Call manager can call each other and likewise for the endpoints registered on
the VCS server. The administrator for Company X realizes he needs a SIP trunk between
the two systems for any video endpoint to call any other video endpoint. Which two steps
must the administrator take to add the SIP trunk? (Choose two.)
• A. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server.
• B. Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
• C. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone.
• D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.
• E. Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate.
Which two statements regarding you configuring a traversal server and traversal client
relationship are true? (Choose two.)
• A. VCS supports only the H.460.18/19 protocol for H.323 traversal calls.
• B. VCS supports either the Assent or the H.460.18/19 protocol for H.323 traversal calls.
• C. VCS supports either the Assent or the H.460.18/19 protocol for SIP traversal calls.
• D. If the Assent protocol is configured, a TCP/TLS connection is established from the traversal client to the traversal server for SIP signaling.
• E. A VCS Expressway located in the public network or DMZ acts as the firewall traversal client.
Which situation requires TCP port 443 to be open for packets that are sourced from the
Internet with a destination in the corporate DMZ?
• A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
• B. when you require encrypted calls to endpoints on your corporate LAN
• C. when you want to enable calls to web applications by using HTTP
• D. when you require administrative access to the Cisco Expressway Edge from the Internet
When considering Cisco Unified Communications Manager failover, how many backup
servers can be configured in a Cisco Unified Communications Manager Group?
• A. 1
• B. 5
• C. 2
• D. 4
• E. 3
• F. 6
When configuring Cisco Unified Mobility, which parameter defines the access control for a
call that reaches out to a remote destination?
• A. Calling Party Transformation Calling Search Space under Remote Destination Profile Information
• B. User Local under Remote Destination Profile Information
• C. Rerouting Calling Search Space under Remote Destination Profile Information
• D. Rerouting Calling Search Space under Remote Destination information
• E. Calling Search Space under Phone Configuration
The VCS Expressway can be configured with security controls to safeguard external calls
and endpoints. One such option is the control of trusted endpoints via a whitelist.
Where is this option enabled?
• A. on the voice-enabled firewall at the edge of the network
• B. on the VCS under Configuration > registration > configuration
• C. on the TMS server under Registrations > whitelist
• D. on the VCS under System > configuration > Registrations
Which two actions ensure that the call load from Cisco TelePresence Video
Communication Server to a Cisco Unified Communications Manager cluster is shared
across Unified CM nodes? (Choose two.)
• A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.
• B. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses.
• C. Create one neighbor zone in VCS for each Unified CM node.
• D. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.
• E. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node.
A local gateway is registered to Cisco TelePresence Video Communication Server with a
prefix of 7. The administrator wants to stop calls from outside the organization being routed
through it. Which CPL configuration accomplishes this goal?
A)
B)
C)
D)
E)
• A. Exhibit A
• B. Exhibit B
• C. Exhibit C
• D. Exhibit D
• E. Exhibit E
What is the default DSCP/PHB for video conferencing packets in Cisco Unified
Communications Manager?
• A. EF/46
• B. CS6/48
• C. AF41/34
• D. CS3/24
In Cisco Unified Communications Manager, where do you configure the +E.164 number
that is advertised globally via ILS?
• A. ILS configuration under Advanced Features
• B. +E.164 alternate number under Directory Number Settings
• C. Device Information under Phone Configuration
• D. Route Pattern under Route/Hunt
Refer to the exhibit.
Which option describes the effect of this configuration?
• A. It implements Cisco United CME redundancy.
• B. It configures a standby Cisco Unified E.
• C. It configures failover.
• D. It implements Cisco IOS redundancy.
• E. It creates dial peers.
• F. It implements HSRP.
Which three items must you configure to enable SAF Call Control Discovery? (Choose
three.)
• A. a calling search space
• B. hosted DN patterns
• C. translation patterns
• D. route patterns
• E. the SIP or H.323 trunk
• F. hosted DN groups
Which two commands verify Cisco IP Phone registration? (Choose two.)
• A. show telephony-service ephone-dn
• B. show voice register session-server
• C. show ephone registered
• D. show ccm-manager hosts
• E. show sip-ua status registrar
What two tasks must be completed in order to support calls between the VCS controlled
endpoints and the Cisco Unified CM endpoints? (Choose two.)
• A. Media Resource Group List.
• B. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
• C. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS.
• D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM.
• E. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
An engineer must enable video desktop sharing between a Cisco Unified Communications
Manager registered video endpoint and a Cisco VCS registered video endpoint. Which
protocol must be enabled in SIP profile for VCS SIP trunk on Cisco Unified
Communications Manager?
• A. RDP
• B. H.264
• C. H.224
• D. H.263
• E. BFCP
The Cisco Unified Communications system of a company has five types of devices:
Cisco Jabber Desktop
CP-7965
DX-650
EX-60
MX-200
Which two types of devices are affected when an engineer changes the DSCP for Video
Calls service parameter? (Choose two.)
• A. DX-650
• B. Cisco Jabber Desktop
• C. CP-7965
• D. EX-60
• E. MX-200
Which two statements about the use of the Intercluster Lookup Service in a multicluster
environment are true? (Choose two.)
• A. Cisco Unified Communications Manager uses the ILS to support intercluster URI dialing.
• B. ILS contains an optional directory URI replication feature that allows the clusters in an ILS network to replicate their directory URIs to the other clusters in the ILS network.
• C. Directory URI replication does not need to be enabled individually for each cluster.
• D. To enable URI replication in a cluster, check the Exchange Directory URIs with Remote Clusters check box that appears in the SIP trunk configuration menu.
• E. If the ILS and directory URI replication feature is disabled on a cluster, this cluster still accepts ILS advertisements and directory URIs from other neighbor clusters; it just does not advertise its local directory URIs.
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
wouldn't it be de, because both are default off and you would have to turn both on for internetworked calls?
Thanks, jose
Planned to take exam tomorrow morning, please share the new question.
What is the valid dump now?
@ Corinths@ Anco
@ Canada
@ Pass
@ sherazkk
Any of you got below question in your exam?..If Yes, What is the correct answer
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.
Holy crap I think you are right about 144. Create a block learned pattern.
Today I passed the exam with 916
I hope it's helpful for you
thanks everybody in this forum for sharing the questions and answers
For those who are here just to grab the dumps, memorize and pass, good luck.
I can verify that the question on the 161 dump, cover most of what is on the test.
I do want to pay it forward, though, with the help of those who have succeeded and reported, in hopes that it helps others.
First of all. @anon12345, @Steve and @Mohan. Thank you thank you thank you for your input! Bringing attention to the meat of the questions we all saw, helped me.
I got over 880 on my score!
My method for prepping:
I went through and marked all the questions that I got. I then verified the sources sited from the dumps. Or I looked it up. There were about 10 questions that were so vague, it was really hard to find a definitive answer. BUT! Many questions, I was able to find the verbage, almost word-for-word.
Anon12345 is a great place to start.
The first part of the test dump mostly pulls questions from the old CCNP Voice test, 300-075. There is an online version that confirms most of the questions from the dumps. There were a few questions that had a different answer, so I went with the 300-075 answers.
Once I had my answers pretty well confirmed, or changed from my research, I then compared what I had, to Anon12345, Steve and Mohan's answers. What they posted was pretty helpful. But as Anon12345 said, don't take their answers for 100%. Verify!
That said, here are a couple of specifics.
Mohan and Steve provided confirmation and details of new questions. This was a HUGE help!
Question 123 has been a huge pain, because only 1 of the 5 choices made sence!
Refer to the exhibit.
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
•User A can hear user В and vice versa.
•User A can hear user C, however user С cannot hear user A.
•User В can heat user C, however user С cannot hear user В.
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user С is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.
Regardless of what dumps say, the fixed option.
E was always correct.
B, however now reads...
"The NAT device is allowing only RTP/RTCP ports from the DMZ to the internal network." Which follows with the fact that audio is not getting to the internal network.
My answer, is B and E from the list above.
Pay attention to the changes that Steve and Mohan have mentioned. I can confirm that all of the new changes replaced the old questions.
Today I passed with 906 marks in second attempt.
Really really appreciate this community!
And I really recommend to read through all comment here before exam.
No new questions in my exam.
but I got all questions which corinth mentioned.
I am not sure correct answer, but you may want to reviews answers well.
good luck and many thanks!
QUESTION NO: 118
How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2
Has the above question changed to how many SCCP phones can register at one time?
No this question did not changed and it's not clear to me. I answered "2".
QUESTION NO: 130
An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources
did this question read as "video call" is the answer still A?
--> I answered "lack of video bandwidth
QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Did you answer A or C?
--> C
QUESTION NO: 155
Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?
A.SRST without MGCP fallback
B.SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C.SRST with MGCP fallback
D.Cisco Unified Communications Manager Express in SRST mode
Did you answer C or D?
--> D. You can find this in the cisco documentation.
QUESTION NO: 133 (did not get in the exam)
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which
result is achieved by enabling this option?
A.Resolve FQDN using DNS type SRV record.
B.Resolve FQDN using DNS type A record.
C.Ensure FQDN is used in SIP Identity header.
D.Ensure FQDN is used in SIP Request header.
Answer: D
QUESTION NO: 142
A presales engineer is working on a quote for a major customer and must evaluate how many Cisco VCS Expressway traversal call licenses for which to plan. Calls to and from which three
routes must the engineer include in the tally? (Choose three.)
A.gateway
B.Cisco 9971 Endpoint
C.border controllers
D.gatekeeper
E.SIP trunk
F.VCS
Answer: C,D,F
QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.
Answer: D,E
QUESTION NO: 156
An engineer has configured a Cisco EX60 to register with a Cisco VCS-C, but the device is not showing up as registered. During troubleshooting, which component will the engineer likely find
missing in the configuration?
A.gatekeeper
B.MCU
C.default gateway
D.TMS
E.DNS
Answer: A
It is about 161q dump
Here's the breakdown of the sections:
VCS Control: 75%
Collab Edge: 50%
CUCM Video Service: 100%
Centralized Call Processing Redundancy: 80%
Multi-site Dial Plan: 700%
CCD/ILS: 100%
Video Mobility: 83%
Bandwidth and CAC: 29% (proof that this test is jacked, this is part of my job and has been for years!)
Big heads up, they've changed a few questions again!
Here's my notes on what they've changed and what I answered on the debated questions that I had today:
6.) No longer asks about avoinind unneccessary intrworking, instead asks what steps are needed to set up h.323 to SIP and vise versa, pick two: (these are as best as I can remember)
a.) Protocol>Sip>Sip on
b.) H.323-SIP internetworking mode On
c.) H.323-SIP internetworking mode Off
d.) H.323-SIP internetworking mode Registered Only
e.) Protocol>H.323>On
f.) Protocol>Sip>configuration>Sip On
I chose D and F.
109.) I answered B
116.)Options are slightly different:
a.) After an Autogenerated device profile is created you can associate it with one or more users.
b.) An autogenerated device profile can be loaded on a device at the same time as a user profile
c.) When Logged off the device can be set to use an autogenerated profile or a user defined profile.
d.) A device profile has most of the same attributes as a physical device
e.) Devices can be configured to allw more than one user to be logger in at the same time.
I answered C and D
118.) How many CUCM nodes cana Skinny phone establisha SCCP connection to at the same time?
I answered 1
120.) I answered D, Local Zone search
121.) I answered Hardware Transcoder and PVDM/DSP
122.) I answered Jabber and DX-650
123.) I answered A and E
124.) Choose 1, did not have B or D. I answered F
125.) I mistakely answered BCE when I should have answered ACE
128.) Answered B, SCCP
129.) Answered E, BFCP
130.) BIG CHANGE!!!! It now reads "An engineer must resolve a VIDEO call failure...." With that in mind I answered B, Lack of Video Bandwidth
131.) Answered B. Be careful, they changed one answer to read the exact same as B but with "Client Zone" and "Server Zone"
132.) Answered C, LRG
134.) Answered D, Apply registration, auth and policies, and E, Manage Bandwidth to restrict....
135.) Answered D
136.) Answered C
138.) Answered ABE
139.) Answered C
140.) Answered BEF
144.) Answered B, Block Learned Pattern
145.) Answered B
146.) Choose Two, got the following choices:
a) Configure SIP route patterns
b) Configure SIP trunk
c) Assign directory URIs to users
d) Configure the SIP profile
I answered C and D
148.) Answer C has been changed to "Some phones at the remote site are assigned to a device pool which does not have an SRST Reference assigned.
With that in mind I picked A and C
149.) The answers I was given:
a) DTMF Relay conversion
b)H.323 Outbound Fast Start
c) SIP early offer
d) IPv4 to IPv6 conversion
e) Multicast MoH
I answered A and D
152.) I answered C and E
153.) Pick one from C E F and H, I chose F, DNS
154.) I answered A
155.) I answered D
158.) A now reads "calling party transformation pattern filed in Route Pattern" and D now reads "Route Pattern field in Route Pattern".
I chose A
160.) Answered A
161.) Answered BDF
New Question!:
Don't remember the wording, but basically "An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com", what happens to the call.
a) Route fails
b) Sent to Cisco.com
c) sent as CCNPCOLAB
d) Sent as [email protected]
I chose D
I Hope this helps everyone, good luck!
If anyone has any questions about any other questions let me know and I'll see what I can help with.
Thanks indeed for the answers really appreciate. I will be attempting again hopefully next week.
Thanks!
What is the latest dump available?
____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D
I think the answer is C, not D. Referring to the Cisco doc:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html
What do you think?
If you remember questions please share, planned to take exam tomorrow
I think it should be:
Config-protocols-Interworking-On
Config-protocols-H323-H323Mode-On
Regards.
thanks for everyone and good luck!
These were the CAC questions I can remember, its possible there has been more of them in the test. Honestly, i have had just 57% score report on CAC, so some of the answers I have provided must have been wrong..
An engineer must resolve a [[VIDEO]] call failure issue. When using RTMT, the engineer notices that the Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option is the cause of the call failure?
B.lack of video bandwidth
In Cisco Unified Communications Manager, where do you configure the default bit rate for audio and video devices?
C. Cisco CallManager service under Service Parameter Configuration
Which statement is true when considering a Cisco VoIP environment for regional configuration?
B. G.729 requires 24K of bandwidth per call.
When you use the Query wizard to configure the trace and log central feature to collect install logs, if you have servers in a cluster in a different time zone, which time is used?
A. TLC adjusts the time change appropriately.
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
D. a hardware transcoder -LTI Transcoder Resource = Local Transcoding Interface
E. DSP resources -DSP PVDM Resource
Which function can be implemented without MTP resources?
e) Multicast MoH
(sorry, cannot remember all the answers, just the one i have selected)
No.
Does Anybody has passed the exam recently!? what is the most accurate dump to download?
Cisco CallManager Extension Mobility supports only one login at a time on a device. Subsequent logins will fail.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a0080153e60.html
Almost Question are from 161Q but 8-10 new questions though.
Please, could you share the following questions?
q133, q142, q147 and q156
thank you very much
I was able to pass this beast of an exam on my 4th attempt today, scored 888...
Here are the answers I chose for the 2 new questions. Please feel free to reach out and I will help anyone that is testing. Thanks to all of those who helped during this journey
2 new questions....
H.323-SIP internetworking mode Registered Only
Protocol>Sip>Sip on
2nd new question i selected "call is sent as ccnpcollab" because transform is in disabled state and is just ignored
Congratulations for passing this difficult exam. Can you help to double check this again.
An engineer is configuring URI calling within the same cluster. Which two actions must be taken to accomplish this configuration? (Choose two.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Activate the URI service in Cisco Unified Serviceability.
D. Configure SIP trunk.
E. Assign directory URIs to users.
F. Configure the SIP profile.
G. Configure the URI service parameters.
Tricky part in the question is "within the same cluster". SIP Profile is needed for Inter Cluster URI Dialing.
So my choice is E and B
Step 1: Assign Directory URI to Users (Option E)
Step 2: Associate Directory URI with Directory Numbers (This option not available)
Step 3:Assign the default directory URI partition to an existing partition that is located in a calling search space(Option B)
Reference : http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/configure_uri_dialing.pdf
Thanks all for your support.
Much appreciated
Congratulations. If you remember any questions, Please share the questions.
Q146. Thanks for the clarification. Yes it is E and F.
@ Scotty...Q95 : Thanks for giving the answer on this.
For q133:
____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D
I think the answer is C, not D. Referring to the Cisco doc:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html
What do you think?
To answer your question, the dump and the test have the same typo.
QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.It implements Cisco UNITED CME redundancy.
B.It configures a standby Cisco Unified E.
C.It configures failover.
D.It implements Cisco IOS redundancy.
E.It creates dial peers.
F.It implements HSRP.
Answer: A
So A is incorrect. The test writer put that mispelling there on purpose. It's not an accident.
So, go with Failover.
Congrats on passing, could you tell me what you answered for the two below please? if you got them in your exam? or your thoughts on the correct answers that "CISCO would recognize as correct"
QUESTION NO: 118
How many nodes can a phone establish a connection to at the same time?
A.4
B.3
C.1
D.2
Has the above question changed to how many SCCP phones can register at one time?
Is the answer still C?
QUESTION NO: 130
An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
Cisco 300-075 Exam
A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources
did this question read as "video call" is the answer still A?
QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Did you answer A or C?
QUESTION NO: 155
Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?
A.SRST without MGCP fallback
B.SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C.SRST with MGCP fallback
D.Cisco Unified Communications Manager Express in SRST mode
Did you answer C or D?
New question (mentioned earlier)
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string :
Pattern Behaviour :add suffix
Replace string : @cisco.com
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
My answer: C (snapshot from transform with state: disabled)
Thank you.
can you please share more information on the dumps you used
Just wondering how you download or access the 161q dump file.
Mi respuesta de segund pregunta fue Internetworking-on y Internetworking-registered-only
pero no estoy seguro de que es la respuesta correcta
Please can someone share the 161q dump? [email protected]
Thanks
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/admin_guide/Cisco_VCS_Administrator_Guide_X7-2.pdf page 170 says:
State Indicates if the transform is enabled or not. Use this setting when making or testing
configuration changes, or to temporarily
enable or disable certain rules. Any
disabled rules still appear in the rules list
but are ignored.
Re q120. I don't see what the local zone has to do with receiving outside calls as this is used only for internal calls so don't think the answer could be D. I believe it is A as from a VCS control perspective all calls are outgoing, even incoming calls. The outside call hits the VCS expressway which in turn notifies the VCS Control which makes the necessary settings to make the call appear as an outgoing call. Otherwise the firewall would stop all incoming calls as they were not initiated from the internal network.
Explanation:
The traversal client (VCS Control) constantly maintains a connection via the firewall to a designated port on the
traversal server (VCS Server). This connection is kept alive by the client sending packets at regular intervals
to the server. When the traversal server receives an incoming call for the traversal client, it uses
this existing connection to send an incoming call request to the client. The client then initiates the
necessary outbound connections required for the call media and/or signaling.
This process ensures that from the firewall’s point of view, all connections are initiated from the
traversal client inside the firewall out to the traversal server.
I just passed with score of 925. Here are the questions I have reviewed myself and are different from what has already been published in this forum.
Which two commands verify Cisco IP Phone registration? (Choose two.)
Changed question:
What are the two commands, one of which can be used to verify SIP phone registration and one can be used to verify SCCP phone registration in CME?
A. show telephony-service ephone-dn
B. show voice register session-server
C. Show ephone registered
D. show ccm-manager hosts
E. show sip-ua status registrar
Answers remains the same: C, E
Refer to the exhibit (screenshot of Transform where some field are popluted).
Pattern type : Regex
Pattern string : ccnpcolab
Pattern Behaviour : Add Suffix
Replace string : @cisco.com
State : Disabled
A call is initiated from endpoint to address "ccnpcolab" from a VCS where the above configuration is applied. Which of the below statements is true?
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
Answer: C (note the status of transform is disabled, which means the dialed URI is NOT BEING transformed and is sent in its original form)
Which two statements about Cisco Unified Communications Manager Extension Mobility are true? (Choose two.)
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. When Logged off the device can be set to use an autogenerated profile or a user defined profile
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time.
Answer: C, D (100% score report on this topic)
QUESTION NO: 118 This is no more valid question, it now says:
How many nodes can a phone establish a SCCP connection to at the same time?
C. 1
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.
Answer I selected: D
-but due to my 75% score report on VCS I would say that better is to go with option A
Which option describes the effect of this configuration? (two identical configurations of CME router)
A. It implements Cisco United CME redundancy. (the word UNITED is there by purpose, meaning this is false answer)
B. It configures a standby Cisco Unified E.
C. It configures failover.
D. It implements Cisco IOS redundancy.
E. It creates dial peers.
F. It implements HSRP.
Answer: C (100% score report on this topic)
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phones located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two.)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway.
C. Phones at the remote site are assigned to the incorrect device pool.
D. The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
Answer: A,C
Which function can be implemented without MTP resources?
-regarding this question - it comes in multiple combinations, but keep in mind
-MTP required:
DTMF relay conversion
DTMF inband RTP-NTE (rfc2833)
terminating a media stream that uses the same codec
H.323 Outbound Fast Start
SIP early offer
IPv4 to IPv6 conversion
-MTP NOT required:
multicast music on hold
delayed offer h.323
SIP delay offer
Which statement about when user A calls user С using SIP are true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
Answer: B !!!!
-note: i have selected A in the exam, aparently i was wrong due to lower score report on this topic
Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit numbers. Which option should be configured if you want outgoing calls from 4-digit internal directory number to be presented as a 10-digit number?
A.calling party transformation pattern filed in Route Pattern
B.AAR group
C.translation pattern
D.route pattern
Answer: A (has changed and makes more sense now)
What are the tasks required to route calls between H323 and SIP ENDPOINTS and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
My answer AE, but apparently this combination is wrong (due to lower VCS score report)
Hope this helps :-)
Why do you think that in the new question ( C. sent as CCNPCOLAB) is the correct?
I friend told me a other new question too:
What are the tasks required to route calls from H323 to SIP and viceversa?
Config-protocols-Interworking-On
Config-protocols-Interworking-Off
Config-protocols-Interworking-registered only
Config-protocols-Sip-Config-Mode-On
Config-protocols-H323-H323Mode-On
¿do you know what are the two answers?
thanks
Based on the past pattern I think cisco will change this exam in one week to three weeks. best of luck to everyone
on which file or document should I start as the baseline? Is this 102q ete file still valid? I understand that many thanksful people posted right answers and explanation on this thread, but really need to know which dump or ete should I start.
Thanks!
I failed exam last week only few points
Can you collect-gather all the correct answers for all the post you have replied and you believe that are wrong?? It would be much easier if you posted all together nice and clean. Thank you in advance
Have you any update for dump. Who entered the exam in today?
Many Thanks
Anyone been for the exam of recent? What are valid currently?
Now it have 8-10 new questions?..Till 24th July everyone was speaking about 2 new questions.
If you can recollect, Please share what are the additional questions other than 161q + 2 discussed new questions.
PASSED: 910
I PASSED TODAY IN UK (18/7/16) WITH A HEALTHY 910, yes scary huh. Man what a ride this beast has been. My 2nd attempt.
First i would like to thank all the guys who took time out to help us specifically Guillaume and Anon12345 who laid the platform and others like Adam, Larry, Steve, Calvin and whoever i missed. This forum has been very helpful.
OK i have spent a bit of time going through every correction for you and the below should be enough for all of you to pass. I have only made corrections to the answers i felt that are wrong based on other comments. the other answer feel correct to me so i have left those. Please analyze your own answers but below are my Personal corrections.
Below are my corrections which should get you guys through this HORRIBLE exam.
57: E
-> The DX650's MAC address is incorrect in the Cisco UCM.
68: BE
-> Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
-> Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
80: B
-> 34(100010)
83: c
-> 2 servers
109: B
-> G.729 requires 24K of bandwidth per call.
115: D
-> Transform
116: CD
-> device can adopt a user profile even when no user is logged in.
-> device profile has most of the same attributes as a physical device.
118: D
-> 2 nodes
120: A
-> The traversal zone on the VCS Control does not have a search rule configured.
121: Qestion has been changed to below
What are requirements for hardware MTP on Cisco IOS routers?
a. PVDM or DSP resource
b. LTI local transcode resource
c. ref2833
d. one audio codec
e. T1 PRI card
Answer AB
122: AB
-> DX-650
-> Cisco Jabber Desktop
123: BE
Answers changed. Finally CISCO released their mistake.
-> Answer B now reads The NAT device is allowing only RTP/RTCP ports from DMZ to internal network. so select this.
-> The router does not have a route back from the DMZ to the internal network.
124: F
-> registration. Question only asks for one answer
125: ACE
-> Configure voice register pool.
-> Configure an SRST reference.
-> Configure the SIP registrar.
127: B
-> Transform
128: B
-> SCCP Gateway
129: E
-> BFCP
130: Careful question has been changed to Engineer must resolve a "VIDEO" call failure issue.
-> I selected B: Lack of video BW
131: Answer is B (BE CAREFUL AS THEY MADE MORE ANSWERS SIMILAR TO THIS like Client zone and server zone, DONT SELECT THIS)
132: C
-> LRG
133: D
-> Ensure FQDN is used in SIP Request header.
134: DE
-> Apply registration, authentication, and media encryption policies.
-> Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of BW
135: D
-> when you require administrative access to the Cisco Expressway Edge from the Internet
136: A
-> It implements Cisco Unified CME redundancy.
138: ABE
-> Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
-> Verify that all phones are registered to a second subscriber server.
-> Verify that media resources fail over to a secondary subscriber server when the publisher fails.
139: Should be A based on the question "best QOS" but it think CISCO got this wrong
-> I selected C: 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
140: BEF
-> hosted DN patterns
-> the SIP or H.323 trunk
-> hosted DN groups
142: CDF
-> border controllers
-> gatekeeper
-> VCS
144: B
-> Create a block learned pattern.
145: B
-> AAR is routing some of the calls.
146: FG (This question only asks for two answers)
-> Assign directory URIs to users.
-> Configure the SIP profile.
147: I wasn't asked this one but i think CISCO got it wrong but if i was asked i would select AF (your risk)
-> Implement local failover
-> Implement centralized failover
148: AC
Answer has changed
-> The site has exceeded the number of SRST endpoints supported by the voice gateway.
-> Phones at the remote site are assigned to the incorrect device pool.
C now reads "Some Phones at the remote site are assigned a device pool without SRST reference"
so select this one. CISCO finally realized their mistake.
149: DE
Answer have been changed to the below.
A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833)
D. Sip delay offer
E. Multicast MOH
-> I selected DE
151: BCE
-> TEHO
-> CER
-> AAR
152: CE
-> show ephone registered
-> show sip-ua status registrar
153: F (Question only asks for one answer: DNS sever)
154: C
-> The user can log in to only one device at a time.)
155: D
-> Cisco Unified Communications Manager Express in SRST mode
158: A
-> calling party transformation pattern
160: I selected A
-> SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
161: BDF
-> The Tomcat certificates do not match.
-> The ILS authentication password does not match.
-> One cluster is using TLS certificate, and the other is using Password.
-------------------------------------------------------
2 EXTRA questions.
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are
three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
Answer is C
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
Answer AE
Good luck guys, i really hope this post of mine should be enough for all of you to pass this crap and we no longer need to keep asking one question at at a time to get the correct answers
Thanks.
Guys i know this might be asking too much, but can one of the guys who have passed (congrats by the way) take a bit of time to helping us all who are trying to pass?
If you could post all the question numbers here with the correct answers? Or if not then at least the ones you have answered differently from the 161 dump answers or the questions that you think the answers should be different?
your help would be highly appreciated. Thanks guys.
sorry I misunderstood ur answer before
@Scotty:
to configure search rule, you need to define source and target.
the problem, call from "outside" to "inside"
source: traversal zone
target : local zone
when you done, the search rule will belong to target.
Please, could you put the questions q133, q142, q147 and q156?
Thank you very much!
Good luck.
63%
75%
100%
90%
100%
100%
100%
43% this one always poor, may be we need also correct some answers from DUMP or here in disscution
All Questions are from dump and Comments, please read all comment here, I think it is enought to pass!
Some questions are not 100 known, please read for them in docs
for exmaple
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
I thought to ansewr AC
But after reading what @JB said anwered CE
It stays that you have to use two ansewrs
but REGISTERED ONLY or ON cant be set at same time, read docs
Also one more question about
Mobility and Presence when WAN goes down
now there are new option BE6000 in fallback
Of cource BE6000 will provide Mobility Presence
I chose CME in SRST in voice gateway
because it asked in BRANCH,
@Faquejai on page 2 was writing about it
Please read all commnts here, and read docs for some questions which are not having exact ansewrs
thank to all who shared exam, and goodluck to all who will take exam
here's some modified Question i got in my exam:
An engineer is configuring URI calling within the same cluster. Which two actions must be taken to accomplish this configuration? (Choose two.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Activate the URI service in Cisco Unified Serviceability.
D. Configure SIP trunk.
E. Assign directory URIs to users.
F. Configure the SIP profile.
G. Configure the URI service parameters.
My choice at that time (E,F)
*i decided to follow configuration guide order
Which statements about when user A calls user С using SIP are true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Cisco VCS Control and Cisco VCS Expressway support static NAT.
C. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
D. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
My choice at that time (B), but after cool down..
I think the correct answer is (A)
afaik, VCSC don't have static NAT and dual interface option
VCSE's indeed use static NAT (if you want deploy it inside NAT)
VCSE's use Advanced Networking/dual interface (if you want deploy it behind NAT) but it's not a MUST (if your NAT device support it)
RTР and RTCP ports must be opened (if it's blocked you simply don't receive any video / audio)
SIP TCP/TLS ports, this one is requirement for SIP traversal calls and the question asking for SIP.
*I hope it'll help you a bit and sorry for my bad english :D
Can you shed some light on these 2 questions, I take my exam on July 5th and feel like I am getting very close to being ready.
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder
E.DSP resources
F.a T1 card
Answer: D,E
Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer
Answer: B
Hey guys, you seem to be pretty good, can you help me with the following as there seems to be a lot of uncertainty
QUESTION NO: 149
Which function can be implemented without MTP resources?
A. DTMF relay conversion
B. terminating a media stream that uses the same codec
C. music on hold
D. SIP early offer
Is the correct answer B or C?
QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A.0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Is the correct answer A as the question implies "best"?
Also the below seems to be an extra question outside the dump?
which 2 things do not use MTP
a> h.323 fast start
b> IPV6 -IPV4
c> DTMF inband RTP-NTE (rfc2833)
d> delayed offer h.323
Is the correct answer B and C?
Thanks
Ramesh
Question 115: transforms, not transfer
____QUESTION 140____
Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
ANSWER: A,B,C
3rd time failed with 845 Marks feeling so depressed.
Anyways can anyone please confirm for the following Questions answers?
146, 147,149,156,160, 132
no idea why still not able to achieve anything above 35% in Collaboration Edge (VCS-E) section and 40% in Implementing bandwidth Mgt and CAC on CUCM section.
@Anon12345 and all who has passed please share the right answers.
For new question i chose option
There is a new question that someone earlier had mentioned.
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
Option A
Please help us to pass this tough exam.
In question 146 I would choose:
C.Associate the directory URIs to directory numbers.
F.Assign directory URIs to users.
Because it says URI calling within the "same cluster"
If someone else has a different option I would like to hear it.
Better do your own research and more on VCS.
A is wrong because A hears C, so C has correct default gateway.
C is wrong because TMS is needed for conferences.
D - problem is not related to call setup (INVITE)
131. Choice between very similar answers
The Expressway-C Traversal Client username/password do not match the Expressway-E Traversal Server username/password
The Expressway-C Traversal client zone username/password do not match the Expressway-E Traversal server zone username/password
I choose second. Citation from VCS admin guide: “The traversal server zone for the VCS client must be configured with the client's authentication”. So the password is configured for the zone, not for the client or server.
160 (select ONE, answers were slightly changed)
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa.
Please Check 161q Dumps
Which file did you used ? dump ?
____QUESTION 57____
A new DX650 IP Phone with MAC address D0C7.8914.132D, IP address is 172.18.32.119 has been added to the Cisco Unified Communications Manager, but is not registering properly. What is causing the failure?
A. The DX650's MAC address is incorrect in the Cisco UCM
B. The location Hub_None has not been activated
C. The DX650 is the incorrect calling search space
D. The DX650 Phones does not support SIP
E. Device Pool cannot be default
ANSWER: A
____QUESTION 109____
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
ANSWER: B
____QUESTION 115____
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which component allows for standardized caller addresses between the endpoints?
A. search rules
B. sip route pattern
c. policy service
D. transfer
ANSWER: D
____QUESTION 116____
Which two statements about Cisco Unified Communications Manager Extension Mobility are true?
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. A device can adopt a user profile even when no user is logged in.
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time
ANSWER: C,D
____QUESTION 122____
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
ANSWER: A,B
____QUESTION 123____
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versa.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A. The Cisco EX60 default gateway of user is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E. The router does not have a route back from the DMZ to the internal network.
ANSWER: A,E
____QUESTION 125____
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.
ANSWER: B,C,E
____QUESTION 128____
An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?
A. H.323 gateway
B. SCCP gateway
C. H.225 trunk
D. MGCP gateway
E. SIP trunk
ANSWER: B
____QUESTION 129____
An engineer must enable video desktop sharing between a Cisco Unified Communications Manager registered video endpoint and a Cisco VCS registered video endpoint. Which protocol
must be enabled in SIP profile for VCS SIP trunk on Cisco Unified Communications Manager?
A.RDP
B. H.264
C. H.224
D H.263
E. BFCP
ANSWER: E
____QUESTION 132____
An engineer is performing an international multisite deployment and wants to create an effective backup
method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. CFUR
C. LRG
D. SRST
ANSWER: C
____QUESTION 133____
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is achieved by enabling this option?
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
ANSWER: D
____QUESTION 134____
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
ANSWER: D,E
____QUESTION 135____
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a
destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway
Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
ANSWER: D
____QUESTION 138____
Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is
disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.
ANSWER: A,B,E
____QUESTION 140____
Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
ANSWER: B,E,F
____QUESTION 142____
A presales engineer is working on a quote for a major customer and must evaluate how many cisco VCS Expressway traversal call licenses for which to plan.
Calls to and from which three routes must the engineer include in the tally?
A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS
ANSWER: C,D,F
____QUESTION 144____
An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes this
task?
A.Create a block route pattern.
B.Create a block learned pattern.
C.Create a block transformation pattern.
D.Create a block translation pattern.
Answer: B
____QUESTION 145____
An administrator is visiting a remote site that has on-net calls with headquarters and one voice
gateway for PSTN calls. When using RTMT to monitor the bandwidth utilization of the remote site,
the administrator notices the OutOfResources counter for the site in LBM has been increasing
slowly in last two weeks, but no call failure reports have been sent from this site. Which description
about this issue is true?
A.The bandwidth settings of the site are fulfilling on-net call volume.
B.AAR is routing some of the calls.
C.The location-based CAC does not work properly.
D.The LBM service is malfunctioning.
ANSWER: B
____QUESTION 152____
Which two commands verify Cisco IP Phone registration? (Choose two.)
A.show telephony-service ephone-dn
B.show voice register session-server
C.show ephone registered
D.show ccm-manager hosts
E.show sip-ua status registrar
ANSWER: C,E
____QUESTION 153____
An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP
address and the system name. After logging into the Cisco VCS Expressway admin page, the
engineer sees this output. Which four options must be configured to complete the Cisco VCS
Expressway system configuration? (Choose four.) ***EXAM ASKS FOR 1
A.NTP server
B.SIP server
C.LDAP server
D.security certificate
E.DHCP server
F.DNS server
G.SIP URI
H.Cisco Unified Communications Manager IP address
ANSWER: F
____QUESTION 158____
Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit
numbers. Which option should be configured if you want outgoing calls from 4-digit internal directory
number to be presented as a 10-digit number?
A.calling party transformation pattern
B.AAR group
C.translation pattern
D.route pattern
ANSWER: A
____QUESTION 160____
Which three statements about when user A calls user using SIP are true? (Choose three.) **EXAM ASKS FOR 1
A.SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B.Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking
option key.
C.Cisco VCS Control and Cisco VCS Expressway support static NAT.
D.Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
E.RTP and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F.The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
ANSWER: B
____QUESTION 161____
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password
ANSWER: B,D,F
Good luck guys. Check this blog out for the answer justifications: https://italchemy.wordpress.com/2017/11/16/cipt2-300-075-exam-helpful-information/
Much appreciated
Where in the Cisco Unified Communications Manager Administration GUI must an engineer navigate to
configure Cisco InterCluster Lookup Service authentication in communication manager?
A. Advanced Features> ILS Configuration> Roles
B. Call Routing > Intercluster Directory URI > Intercluster Directory URI configuration
C. Call Routing > Intercluster Directory URI
D. Advanced Features > ILS Configuration
Answer: D
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
QUESTION NO: 139
Which option indicates the best QoS parameters for interactive video?
A. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Answer should be A.
Read this:
It says NO MORE. So best answer is A.
Loss should be no more than 1 percent.
One-way latency should be no more than 150 ms.
http://www.ciscopress.com/articles/article.asp?p=357102&seqNum=2
QUESTION NO: 115
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?
A.search rules
B.SIP route pattern
C.policy service
D.transform
ANSWER = A, search rules.
Q. 127 is B. transforms
I just passed the exam. I will post the questions and answers from 356 dump that I think will help you to night. I don't understand why those individuals never responded to your questions.
QUESTION 75
A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway in the DMZ to facilitate VPN-less SIP calls with users outside of the
network. However, the users report that calls via the VCS are erratic and not very consistent.What must the administrator configure on the firewall to stabilize this deployment?
A. The VCS Control should not be on the LAN, but it must be located in the DMZ with the Expressway.
B. The firewall at Company X must have all SIP ALG functions disabled.
C. The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the same network that the VCS Control is on.
D. A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway and the VCS Control servers.
QUESTION 95
When using SAF, how do you prevent multiple nodes in a cluster from showing up in the Show Advance section of the SAF Forwarder configuration?
A. Configure the publisher node only in the SAF Forwarder configuration page.
B. Append an @ symbol at the end of the client label value in the SAF Forwarder configuration page.
C. Configure the correct node in the EIGRP configuration of the gateway router that is associated with the Cisco Unified Communications Manager node.
D. Configure the SAF Security Profile Configuration to support only a single node.
QUESTION 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.
QUESTION 123
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user В and vice versa.
User A can hear user C, however user С cannot hear user A.
User В can heat user C, however user С cannot hear user В.
Which two properties are the most likely reasons for this issue? (Choose two.)
A. The Cisco EX60 default gateway of user С is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E. The router does not have a route back from the DMZ to the internal network.
QUESTION 135
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
QUESTION 138
Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.
QUESTION 149
Which function can be implemented without MTP resources?
A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833);
D. delayed offer h.323
QUESTION 160
Which three statements about when user A calls user С using SIP are true? (Exam Asks only One)
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
You should really make a study of the course materials and still then it's hard to find the right answers.
Just wondering how you download or access the 160q dump file. The comments provided are all on the Quinn.102q dump file. Any assistance will be great.
¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.
My Answer: C
The Search rules page (Configuration > Dial plan > Search rules) is used to configure how the
Expressway routes incoming search requests to the appropriate target zones (including the Local Zone) or policy services.
Question N0: 109
As far as I know CUCM calculates bandwidth with headers (g.729: 8k+16k headers = 24k total bandwidth) in the Location section for sure. I do not know if it calculates bandwidth similarly in Region too...
@KMX
Thank you for breaking down the different options. My question though, is that B is saying that RTP/RTCP traffic is only allowed to flow towards user C. The problem states that User C is the one who cannot hear. Shouldn't it be the other way around?
It really seems like E is the only valid option.
Can you please explain question
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
why answer AE?
Q120: D
Q123: A and E
Q135: D
Q138: A B E
Q160: A
I just passed, with an 880 on my exam. This forum has been so helpful, that I wanted to also add some more helpful information.
- First off, the 161 dump is valid, as far as questions. The answers need to be checked!
- For anyone just looking to find the dumps, memorize them, answers and take the test, good luck! I failed my first test, even though I had checked some of the answers on my own. After I failed the test, I went back through and tried to look up as many questions on my own to verify. I learned ALOT!
- @anon12345, @Steve and @Mohan, thank you for the wealth of information you shared!
- I recommend starting with what @anon2345 posted.
Now for some more specific information, to help those who have been debating some of the more difficult questions to verify.
Steve and Mohan both reported back with some new information. There are some test questions that have been updated. From what they have reported, it looks like maybe the original version of those questions either didn't have a right answer, or the right answer was almost as vague as the other wrong answers. Check what they say about those question. I think they will help you, but still check their answers. I had already gone through and verified ALL of the questions I remembered seeing on the test, then verifed the ones they also mentioned. Which was a difference of about 20 questions. Most of my answers matched theirs. But not all.
Question 123 according to the dumps is pretty much impossible. I want to report what I saw, because it gives you two possible right answers. It will help all those who have been debating the question since I've been studying for this exam.
*********************************
QUESTION 123
Refer to the exhibit.
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
•User A can hear user В and vice versa.
•User A can hear user C, however user С cannot hear user A.
•User В can heat user C, however user С cannot hear user В.
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user С is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E.The router does not have a route back from the DMZ to the internal network.
*********************************
Everyone can agree that E is correct. But none of the others match the direction of traffic, where audio is NOT moving from the internal network to the DMZ.
On the test, they changed B.
It now reads
B.The NAT device is allowing only RTP/RTCP ports from the DMZ to the internal network.
I chose B and E.
The first time I took the test, I thought I had the questions memorized. Then, after failing, taking the time to learn the questions, and also reading what Steve and Mohan added after I failed, helped me not only verify the old answers, but learn why some of the questions I thought I knew, just didn't seem to have the right answers. They modified them!
Mohan pointed out a new one, with a picture, and being asked about how the traffic will be routed.
*************************************
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
************************************
I chosed same as Mohan. A.
Because state was disabled.
One last bit, I hope this helps save some time.
* The last 53 questions of the dump, are were I saw about 2/3 of the exam, and they are discussed at great length here. Check out what people say, but still verify!
* In looking up the questions, and the correct answers online, I found the correct answers WORD-FOR-WORD on Cisco online documentation. Most of them are in tables, or setup guides. Most of there were "Deployment Guides" or "Administration Guides."
* Someone else mentioned that Questions 57 - 68 aren't on the exam. I can comfirm that I didn't see ANY of those questions either time I took the exam.
* Any questions from the first part of the test, that don't involve VCS topics, look like they were pulled from an old CCNP CIPT2 (300-075) test. I even had to correct a couple of questions from THIS dump, based on the old dump.
* Look at the question list that people posted they saw, and add that to your list of questions to verify.
Can you give me your advice on this question, this another grey area..
After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the problem?
A. The phone at the remote location is a different model than the phone in the user's main office.
B. The user's Extension Mobility profile is misconfigured
C. The user can log in to only one device at a time.
D. The device pool is misconfigured.
Note: Dump answer is A but other said that the answer is C.
Re Q146 Hope this helps.
Step 1
Assign directory URIs to the users in your network.
Step 2
Associate the directory URIs to directory numbers by assigning both a primary extension and phone to the users in your network.
Step 3
Assign the default directory URI partition to an existing partition that is located in a calling search space by doing the following:
In Cisco Unified CM Administration, choose System > Enterprise Parameters
For the Directory URI Alias Partition enterprise parameter, choose an existing partition that is in an existing calling search space.
Set the URI Dialing Display Preference service parameter for URI dialing as URI for calling display in call park display URI of the calling party. DN is the default setting for the service parameter.
Step 4
Configure the SIP profiles in your network by configuring the following fields in the SIP Profile Configuration window:
Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network.
Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.
Note At this point, intracluster URI dialing is configured. The remaining steps are used to configure intercluster URI dialing.
Step 5
For all the SIP trunks in your network, configure whether the network uses blended addressing by configuring the Calling and Connected Party Info Format drop-down list box in the Trunk Configuration window.
Step 6
Set up ILS on all the clusters in your network.
Step 7
Enable intercluster URI dialing with ILS by checking the Exchange Directory URI Catalogs with Remote Clusters check box in the Intercluster Directory URI Configuration window.
Step 8
In the Intercluster Directory URI Configuration window, create a route string that remote clusters will use to route to this cluster.
Step 9
Configure SIP route patterns that match the route strings for the remote clusters in your ILS network.
Step 10
Associate the SIP route patterns that you created to an outbound SIP trunk or route list.
Step 11
If you are connecting your ILS network to a Cisco TelePresence Video Communications Server, or a third-party call control system, import directory URI catalogs from the other system into Cisco Unified Communications Manager.
Step 12
If your deployment uses digit transformations to transform calling party directory numbers, configure calling party transformation patterns and apply them to the Inbound Call Settings for the phone or device pool. This configuration is used for intercluster calls.
Step 13
If you applied digit transformation patterns in the previous step, configure calling party transformation patterns for the Outbound Call Settings for the phone or device pool. This configuration is used for intracluster calls.
I suspect A & D to be the answer. Lets see....
QUESTION 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.
I didn't have Q68 and Q80 during my exam
******QUESTION 98******
Which functionality does ILS use to link all hub clusters in an ILS network?
A. Fullmesh
B. Automesh
C. ILS updates
D. multicast
My Ans: A
******QUESTION 109******
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
My Answer: C
¨¨¨¨¨QUESTION 122¨¨¨¨¨¨
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
My Answer: A,C
You might need to recheck my answer on this one.
QUESTION NO: 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to
accomplish this configuration? (Choose two.)
C. Associate the directory URIs to directory numbers.
F. Assign directory URIs to users.
G. Configure the SIP profile.
QUESTION NO: 160
Which three statements about when user A calls user C using SIP are true? (Choose three.)
My Answer:
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
But I think the correct answer is: A, if it's not vice verza.
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
B. The Tomcat certificates do not match.
D. The ILS authentication password does not match.
F. One cluster is using TLS certificate, and the other is using Password.
__Question 148___
You have deployed a Cisco 2821 ISR to poerform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phone located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway
C. Phone at the remote site are assigned to the incorrect device pool
D. The ccm-manager fallback-mgcp cimmand is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
Note: Dump answer are B & D but other said that the answer are A & E.
----
I've deployed SRST dozens and dozens of times.
C is correct:
C. Phone at the remote site are assigned to the incorrect device pool
The device pool contains SRST reference. If phone doesn't belong to the srst device pool, that phone won't be in SRST if connection to CUCM is lost.
"A" is 2nd choice
A. 4
B. 3
C. 1
D. 2
Correct Answer: D
Can someone verify this answer or explain? A phone will only register to 1 CUCM node correct?
Here are my answers for the questions you asked:
109: answer has changed, i selected "The site has exceeded the number of SRST endpoints supported by the voice GW" and "Some Phones at the remote site are assigned a device pool without SRST reference"
116: CD
120: I selected A (traversal zone has no search rule configured) because issue is for external calls.
123: Answer has changed so i selected "The nat device is allowing only RTP/RTCP ports from DMZ to internal network" and "The router does not have a route back from the DMZ to the internal network"
132: C (LRG) answer has changed and it's more clear now, it do not mention "call limit" anymore so AAR is not relevant anymore.
136: A (CME redudancy)
146: you have to select only two answers and it's more easy. Answers are "configure SIP profile" and "assign directory URI to users"
147: i did not hit that one but even in the dump question is not clear at all so i'm sorry but i can't help on this one.
149: answer has changed and it's more clear, it asks for 2 answers and i selected "MOH" and "sip delayed offer"
153: answers has changed, only 4 choices now (ldap server, dhcp, cucm ip, and DNS) so it's clearly DNS.
156: I selected A (gatekeeper) because gatekeeper VCS is the component in charge for registration.
160: only one answer needed and i selected "SIP TLS" because they ask for "SIP call". I added a comment during the exam because it was not clear to me. RTP/RTCP ports need to be open too if you want to get video and voice media.
QUESTION 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.
A. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server.
B. Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
C. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone.
D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.
E. Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate.
AC is right?
Congrats on passing, let us know if there was a grace period!
Would it be possible for you to go through your list from the 23rd and verify which answers were correct?
Thanks!
Just passed this crap with 906.
All questions except the 2 already mentionned are in the 161 file.
Some questions were rewrited and it's more clear.
First new question
I selected :
Config-protocols-Interworking-On
Config-protocols-H323-H323Mode-On
2nd new question i selected "call is sent as ccnpcollab" because transform is in disabled state and is just ignored.
During exam do not hesitate to comment when the questions are not clear.
Feel free to ask for other questions.
Good luck !
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
Step 3 refers to assigning but the possible answer refers to configuring so I don't think it is this. I still think it is E and F as SIP profile is required as part of the Intra Cluster setup as per the referenced document. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.pdf
Could you please explain question
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
why the answer should be AE?
I suspect it to be A and C
Just a confirmation of ne of the D&D's
Risk Rating Calculation
Risk rating is a quantitative measure of your network's threat level before IPS mitigation. For each event fired by IPS signatures, Cisco IPS Sensor Software calculates a risk rating number. The factors used to calculate risk rating are:
• Signature fidelity rating: This IPS-generated variable indicates the degree of attack certainty.
• Attack severity rating: This IPS-generated variable indicates the amount of damage an attack can cause.
• Target value rating: This user-defined variable indicates the criticality of the attack target. This is the only factor in risk rating that is routinely maintained by the user. You can assign a target value rating per IP address in Cisco IPS Device Manager or Cisco Security Manager. The target value rating can raise or lower the overall risk rating for a network device. You can assign the following target values:
– 75: Low asset value
– 100: Medium asset value
– 200: Mission-critical asset value
• Attack relevancy rating: This IPS-generated value indicates the vulnerability of the attack target.
• Promiscuous delta: The risk rating of an IPS deployed in promiscuous mode is reduced by the promiscuous delta. This is because promiscuous sensing is less accurate than inline sensing. The promiscuous delta can be configured on a per-signature basis, with a value range of 0 to 30. (The promiscuous delta was introduced in Cisco IPS Sensor Software Version 6.0.)
• Watch list rating: This IPS-generated value is based on data found in the Cisco Security Agent watch list. The Cisco Security Agent watch list contains IP addresses of devices involved in network scans or possibly contaminated by viruses or worms. If an attacker is found on the watch list, the watch list rating for that attacker is added to the risk rating. The value for this factor is between 0 and 35. (The watch list rating was introduced in Cisco IPS Sensor Software Version 6.0.)
Thanks
A. Disassociate the trunk from the CCD advertising service or CCD requesting service.
B. Delete the trunk from the CCD requesting service node.
C. Place the Cisco Unified Communications Manager node in standby mode.
D. Redirect CCD advertising and requesting services to another Cisco Unified Communications Manager.
Correct Answer: A
Which functionality does ILS use to link all hub clusters in an ILS network?
A. Fullmesh
B. Automesh
C. ILS updates
D. multicast
DUMP ANSWER: B
I THINK IS A.
EXPLAIN:
When a new hub cluster registers to another hub cluster in an existing ILS network, ILS automatically creates a full mesh connection between the new hub cluster and all the existing hub clusters in the ILS network.
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/configure___intercluster_lookup_service.pdf
Q144 I think is Create a block learned pattern. based on that:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011101.pdf
If you want to prevent a local Cisco Unified Communications Manager cluster from routing calls to a learned alternate number
or learned alternate number pattern, you can configure a local blocking rule on that cluster.
@Joshua:
Q115 and Q127
I think the answer for both is TRANSFORM based on that:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Basic_Configuration_Control_with_Expressway_Deployment_Guide_X7-2.pdf
The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices.
For example, if the called address is an H.323 E.164 alias “01234” the VCS will automatically append the configured domain name (in this case example.com)
to the called address (that is, [email protected] making it into a URI), before attempting to set up the call.
Q139: definitively is A because it is best QoS compare to C. C being the limit max from Cisco.
@faquejai:
Q98: Answer is B because "ILS uses automesh functionality to create a full mesh connection between all hub clusters within an ILS network"
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmfeat/CUCM_BK_CEF0C471_00_cucm-features-services-guide-90/CUCM_BK_CEF0C471_00_cucm-features-and-services-guide_chapter_011111.pdf
@Kiki, Sona use pdf 161q format
@all
These are the questions that made us in trouble so we have to focus on them:
Q93: Could be answer A.
Q116: Most of you answer B,C. But I don't understand how an Autogenerated devices profile can be loaded at teh same time as a user profile.
Q123: Knowing that on each test the response list change and number of response requested also.
Q125: because I am not 100% sure, and because of my % test result.
Q135:
Q136: because I am not 100% sure, and because of my % test result.
Q137
Q138: because I am not 100% sure, and because of my % test result.
Q146: Knowing that on each test the response list change and knowing that the steps to configure URI calling are:
1-Assign directory URIs to users
2-Associate the directory URIs to directory numbers.
3-Configure the SIP profile.
4-Configure SIP trunk.
5-Configure SIP route patterns
Q148:
Q160: Knowing that on each test the responses list change and number of responses requested also.
Can you please confirm about new questions? Were they from 161Q Dumps or out of them?
Thanks
Bahrain
Nov 13, 2017
Report Comment
Passed yesterday with 897. All Qs from 161Q dump but need to recheck your answers What is the exact name of the ETE File for 161Q? Thanks. What is your preferred ETE Player?
I meant to say @Chase!
Here are my scores
VCS Control: 88%
Collab Edge: 50%
CUCM Video Service: 57%
Centralized Call Processing Redundancy: 60%
Multi-site Dial Plan: 100%
CCD/ILS: 100%
Video Mobility: 100%
Bandwidth and CAC: 29%
There is a new question that someone earlier had mentioned.
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
I selected A since the state is disabled as another user on the board had commented.
Any advise from anyone who has passed can give me to bring up these scores. I was only a question or 2 away from taming this beast.
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?
b. Configure the directory uri partition and calling search space
c. Associate the directory uris to directory numbers
f. Assign directory uris to users
g. Configure the sip profile
page 2 steps 1 - 4
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.pdf
What you guys think?
have you used the same dump and its answers
as in dump or researched your own ..answers ... many questions have incorrect answers in dump .. but i am wondering if in exams these questions are being evaluated on incorrect answers as well .. so i want to make sure as many people complaining the same thing if you research and use correct answers you will fail and if you simply follow dump you will pass ... what you think .
A. After an autogenerated device profile is created, you can associate it with one or more users.
B. An autogenerated device profiles can be loaded on a device at the same time as a user profile.
C. A device can adopt a user profile even when no user is logged in.
D. A device profile has most of the same attributes as a physical device.
E. Devices can be configured to allow more than one user to be logged in at the same time.
Correct Answer: BC
It is correct?
Special thanks to @anon12345, @corinth, @Hi-Octane and @suman for their guidance.
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?
The following suggests one or the other but not both.
Within a centralized call processing cluster with N sites, you can implement Tail-End Hop-Off (TEHO) using one of the following methods:
–TEHO with centralized failover
This method involves configuring a set of N route patterns in a global partition, with each pattern pointing to a route list that has the appropriate remote site route group as the first choice and the central site route group as the second choice.
–TEHO with local failover
This method involves configuring N sets of N route patterns in site-specific partitions, with each pattern pointing to a route list that has the appropriate remote site route group as the first choice and the local site route group as the second choice.
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?
Please Confirm Answers and cross verify with documents.
1- VCS Control: 88%
2- Collaboration Edge(VCS Expressway): 75%
3- Configure CUCM Video Service parameters: 100%
4- Describe and Implement Centralized Call Processing Redundancy: 80%
5- Describe and configure a Multi-Site Dial Plan for CUCM: 88%
6- Implement Call Control Discovery/ILS: 83%
7- Implement Video Mobility Features: 83%
8- Implement Bandwidth Management and Call Admission Control on CUCM: 29%
New Question : I dont remember Exactly question they asked in Exam
An engineer is troubleshooting a dial path etc" and it gives a config snapshot. There are
three fields populated with "CCNPCOLAB", "Add Suffix" and "@Cisco.com"
Mode : .....
Pattern type : Regex
Pattern string : @cisco.com
Pattern Behaviour :.......
Replace string : ccnpcollab
Target : .....
State : Disabled
A. Can not route call
B.Sent to Cisco.com
C. sent as CCNPCOLAB
D. Sent as [email protected]
I selected A since the state is disabled.
Not sure whether is right or wrong please check guides.
QUESTION NO: 1
Which parameter should be set to prevent H.323 endpoints from registering to Cisco
TelePresence Video Communication Server automatically?
A.
On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Discover to off.
B.
On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Registration to off.
C.
On the VCS, navigate to Configuration, Registration, Allow List, and set Auto Registration to
off.
D.
On the VCS, navigate to Configuration, Registration, Configuration, and set Auto Registration
to off.
Answer: A
QUESTION NO: 7
Which three statements about configuring an encrypted trunk between Cisco TelePresence Video
Communication Server and Cisco Unified Communications Manager are true? (Choose three.)
A.
The root CA of the VCS server certificate must be loaded in Cisco Unified Communications
Manager.
B.
A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP.
C.
The Cisco Unified Communications Manager trunk configuration must have the destination port
set to 5061.
D.
A SIP trunk security profile must be configured with Device Security Mode set to TLS.
E.
A SIP trunk security profile must be configured with the X.509 Subject Name from the VCS
certificate.
F.
The Cisco Unified Communications Manager zone configured in VCS must have SIP
authentication trust mode set to On.
G.
The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set
to Off.
Answer: A,C,E
QUESTION NO: 9
Which two actions ensure that the call load from Cisco TelePresence Video Communication
Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes?
(Choose two.)
A.
Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.
B.
Create a single traversal client zone in VCS with the Unified CM nodes listed as location
peer addresses.
C.
Create one neighbor zone in VCS for each Unified CM node.
D.
Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.
E.
In VCS set Unified Communications mode to Mobile and remote access and configure each
Unified CM node.
Answer: A,D
QUESTION NO: 13
If delegated credentials checking has been enabled and remote workers can register to the VCS
Expressway, which statement is true?
A.
H.323 message credential checks are delegated.
B.
SIP registration proxy mode is set to On in the VCS Expressway.
C.
A secure neighbor zone has been configured between the VCS Expressway and the VCS Control.
D.
SIP registration proxy mode is set to Off in the VCS Expressway.
Answer: D
QUESTION NO: 14
Which two options should be used to create a secure traversal zone between the Expressway-C
and Expressway-E? (Choose two.)
A.
Expressway-C and Expressway-E must trust each other's server certificate.
B.
One Cisco Unified Communications traversal zone for H.323 and SIP connections.
C.
A separate pair of traversal zones must be configured if an H.323 connection is required and
Interworking is disabled.
D.
Enable username and password authentication verification on Expressway-E.
E.
Create a set of username and password on each of the Expressway-C and Expressway-E to
authenticate the neighboring peer.
Answer: A,C
Explanation:
QUESTION NO: 15
Which two statements regarding you configuring a traversal server and traversal client
relationship are true? (Choose two.)
A.
VCS supports only the H.460.18/19 protocol for H.323 traversal calls.
B.
VCS supports either the Assent or the H.460.18/19 protocol for H.323 traversal calls.
C.
VCS supports either the Assent or the H.460.18/19 protocol for SIP traversal calls.
D. If the Assent protocol is configured, a TCP/TLS connection is established from the
traversal client to the traversal server for SIP signaling.
E.
A VCS Expressway located in the public network or DMZ acts as the firewall traversal client.
Answer: B,D
QUESTION NO: 16
What is the standard Layer 3 DSCP media packet value that should be set for Cisco
TelePresence endpoints?
A.
CS3 (24)
B.
EF (46)
C.
AF41 (34)
D.
CS4 (32)
Answer: D
QUESTION NO: 17
When you configure QoS on VCS, which settings do you apply if traffic through the VCS should
be tagged with DSCP AF41?
A.
Set QoS mode to DiffServ and tag value 32.
B.
Set QoS mode to IntServ and tag value to 34.
C.
Set QoS mode to DiffServ and tag value 34.
D.
Set QoS mode to IntServ and tag value to 32.
E.
Set QoS mode to ToS and tag value to 32.
Answer: C
QUESTION NO: 22
Which two options are valid service parameter settings that are used to set up proper video
QoS
behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)
A.
DSCP for Video Calls when RSVP Fails
B.
Default Intraregion Min Video Call Bit Rate (Includes Audio)
C.
Default Interregion Max Video Call Bit Rate (Includes Audio)
D.
DSCP for Video Signaling
E.
DSCP for Video Signaling when RSVP Fails
Answer: A,C
QUESTION NO: 34
Which two options should be selected in the SIP trunk security profile that affect the SIP
trunk pointing to the VCS? (Choose two.)
A.
Accept Unsolicited Notification
B.
Enable Application Level Authorization
C.
Accept Out-of-Dialog REFER
D.
Accept Replaces Header
E.
Accept Presence Subscription
Answer: A,D
QUESTION NO: 35
Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with
video endpoints registered on both systems. Users find that video endpoints registered on
Call
manager can call each other and likewise for the endpoints registered on the VCS server. The
administrator for Company X realizes he needs a SIP trunk between the two systems for any
video
endpoint to call any other video endpoint. Which two steps must the administrator take to add
the
SIP trunk? (Choose two.)
A.
Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the
VCS server.
B.
Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
C.
Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option
VCS Configuration > Zones > zone.
D.
Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and
Transport set to TCP.
E.
Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco
UCM to communicate.
Answer: A,C
QUESTION NO: 48
Which statement about setting up FindMe in Cisco TelePresence Video Communication Server is
true?
A.
Users are allowed to delete or change the address of their principal devices.
B.
Endpoints should register with an alias that is the same as an existing FindMe ID.
C.
If VCS is using Cisco TMS provisioning, users manage their FindMe accounts via VCS.
D.
A VCS cluster name must be configured.
Answer: D
QUESTION NO: 74
Company X currently uses a Cisco Unified Communications Manager, which has been configured
for IP desk phones and Jabber soft phones. Users report however that whenever they are out of
the office, a VPN must be set up before their Jabber client can be used. The administrator
for
Company X has deployed a Collaboration Expressway server at the edge of the network in an
attempt to remove the need for VPN when doing voice. However, devices outside cannot
register.
Which two additional steps are needed to complete this deployment? (Choose two.)
A.
A SIP trunk has to be set up between the Expressway-C and Cisco UCM.
B.
An additional interface must be enabled on the Cisco UCM and placed in the same subnet at the
Expressway.
C.
The customer firewall must be configured with any rule for the IP address of the external
Jabber
client.
D.
The Expressway server needs a neighbor zone created that points to Cisco UCM.
E.
Jabber cannot connect to Cisco UCM unless it is on the same network or a VPN is set up from
outside.
Answer: A,D
QUESTION NO: 75
A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway
in the DMZ to facilitate VPN-less SIP calls with users outside of the network. However, the
users
report that calls via the VCS are erratic and not very consistent.
What must the administrator configure on the firewall to stabilize this deployment?
A.
The VCS Control should not be on the LAN, but it must be located in the DMZ with the
Expressway.
B.
The firewall at Company X must have all SIP ALG functions disabled.
C.
The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the
same
network that the VCS Control is on.
D.
A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway
and
the VCS Control servers.
Answer: B
QUESTION NO: 78
Company X wants to implement RSVP-based Call Admission Control and move away from the current
location-based configuration.
Where does the administrator go to create a default profile?
A.
System > Call Manager >Clusterwide> Service Parameters > RSVP
B.
System > Service Parameters > RSVP
C.
System > Service Parameters > Call Manager >Clusterwide parameters > RSVP
D.
on each MGCP gateway at all remote locations
Answer: C
QUESTION NO: 79
Where can you change the clusterwide DSCP setting for Cisco Unified Communications
Manager?
A.
enterprise parameters
B.
service parameters
C.
enterprise phone configuration
D.
Ethernet configuration
Answer: B
QUESTION NO: 81
Which two statements about remote survivability are true? (Choose two.)
A.
SRST supports more Cisco IP Phones than Cisco Unified Communications Manager Express in SRST
mode.
B.
Cisco Unified Communications Manager Express in SRST mode supports more Cisco IP Phones
than SRST.
C.
MGCP fallback is required for ISDN call preservation.
D.
MGCP fallback functions with SRST.
Answer: A,D
QUESTION NO: 84
Which three CLI commands are used when configuring H.323 call survivability for all calls?
(Choose three.)
A.
voice service voip
B.
telephony-service
C.
h323
D.
call preserve
E.
call-router h323-annexg
F.
transfer-system
Answer: A,C,D
QUESTION NO: 91
Which two statements about the use of the Intercluster Lookup Service in a multicluster
environment are true? (Choose two.)
A.
Cisco Unified Communications Manager uses the ILS to support intercluster URI dialing.
B.
ILS contains an optional directory URI replication feature that allows the clusters in an ILS
network to replicate their directory URIs to the other clusters in the ILS network.
C.
Directory URI replication does not need to be enabled individually for each cluster.
D.
To enable URI replication in a cluster, check the Exchange Directory URIs with Remote
Clusters
check box that appears in the SIP trunk configuration menu.
E.
If the ILS and directory URI replication feature is disabled on a cluster, this cluster still
accepts ILS
advertisements and directory URIs from other neighbor clusters; it just does not advertise
its local
directory URIs.
Answer: A,B
Explanation:
QUESTION NO: 93
When implementing a dial plan for multisite deployments, what must be present for SRST to
work
successfully?
A.
dial peers that address all sites in the multisite cluster
B.
translation patterns that apply to the local PSTN for each gateway
C.
incoming and outgoing COR lists
D.
configuration of the gateway as an MGCP gateway
Answer: B
QUESTION NO: 95
When using SAF, how do you prevent multiple nodes in a cluster from showing up in the Show
Advance section of the SAF Forwarder configuration?
A.
Configure the publisher node only in the SAF Forwarder configuration page.
B.
Append an @ symbol at the end of the client label value in the SAF Forwarder configuration
page.
C.
Configure the correct node in the EIGRP configuration of the gateway router that is
associated
with the Cisco Unified Communications Manager node.
D.
Configure the SAF Security Profile Configuration to support only a single node.
Answer: B
QUESTION NO: 96
Which statement about the SAF Client Control is correct?
A.
The SAF Client Control is a configurable inherent component of Cisco Unified Communications
Manager.
B.
The SAF Client Control is a non-configurable inherent component of Cisco Unified
Communications Manager.
C.
The SAF Client Control is a non-configurable inherent component of the Cisco IOS Routers.
D.
The SAF Client Control is a configurable inherent component of the Cisco IOS Routers.
Answer: B
QUESTION NO: 103
How many Cisco Unified Mobility destinations can be configured per user?
A.
1
B.
10
C.
4
D.
6
Answer: B
QUESTION NO: 104
When configuring Cisco Unified Mobility, which parameter defines the access control for a
call that
reaches out to a remote destination?
A.
Calling Party Transformation Calling Search Space under Remote Destination Profile
Information
B.
User Local under Remote Destination Profile Information
C.
Rerouting Calling Search Space under Remote Destination Profile Information
D.
Rerouting Calling Search Space under Remote Destination information
E.
Calling Search Space under Phone Configuration
Answer: C
QUESTION NO: 106
In Cisco Unified Communications Manager, where do you configure the default bit rate for
audio
and video devices?
A.
Enterprise Parameters
B.
Region under Region Information
C.
Cisco CallManager service under Service Parameter Configuration
D.
Enterprise Phone Configuration
Answer: C
QUESTION NO: 109
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A.
G.711 requires 128K of bandwidth per call.
B.
G.729 requires 24K of bandwidth per call.
C.
The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D.
To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and
only use G.711 between regions.
Answer: B
QUESTION NO: 114
What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified
Communications Manager?
A.
EF/46
B.
CS6/48
C.
AF41/34
D.
CS3/24
E.
CS4/32
Answer: E
QUESTION NO: 115 As per Dumps Answer B is wrong Correct answer is D
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?
A.
search rules
B.
SIP route pattern
C.
policy service
D.
transform
Answer: D
QUESTION NO: 116 As Per Dumps B&C In Exam Option C was not present but still the option was
to choose 2.
I Selected B&D
Which two statements about Cisco Unified Communications Manager Extension Mobility are true?
(Choose two.)
A.
After an autogenerated device profile is created, you can associate it with one or more
users.
B.
An autogenerated device profiles can be loaded on a device at the same time as a user
profile.
C.
A device can adopt a user profile even when no user is logged in.
D.
A device profile has most of the same attributes as a physical device.
E.
Devices can be configured to allow more than one user to be logged in at the same time.
Answer: B,D
QUESTION NO: 117
When you configure a globalized dial plan, in which three ways can you enable ingress
gateways
to process calls? (Choose three.)
A.
Configure the called-party transformation settings for incoming calls on H.323 gateways.
B.
Configure translation patterns in the partitions used by the gateway calling search space.
C.
Configure SIP trunks between Cisco Unified Communications Manager clusters.
D.
Configure a remote site device pool.
E.
Configure a hunt group.
F.
Configure the gateway with prefix digits to add necessary country and region codes.
Answer: A,B,F
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/dialplan.html#wp115316
6
QUESTION NO: 118
How many nodes can a phone establish a connection to at the same time?
A.
4
B.
3
C.
1
D.
2
Answer: D
QUESTION NO: 119
Company X has a primary and a backup Cisco Unified Communications Manager instance. The
administrator had to do maintenance on the primary node and did a shutdown, which resulted in
a
failover to the backup node. What happens when the primary node comes back online?
A.
The primary node becomes the backup node.
B.
Endpoints detect that the primary is back and reregisters automatically.
C.
The backup node must be shut down first to allow the endpoints to realize that the primary
node is
online again.
D.
Nothing, the endpoints only failover when the node they lose connection to their registered
node.
Answer: B
QUESTION NO: 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To
facilitate
external calls, VCS Expressway is deployed and traversal server zone is set up there. Video
endpoints inside Company X have registered, but are unable to receive calls from outside
endpoints. Which option could be the cause of this issue?
A.
The traversal zone on the VCS Control does not have a search rule configured.
B.
The access control list on the VCS Control must be updated with the IP for the external
users.
C.
When a traversal zone is set up on VCS Control only outbound calls are possible.
D.
The local zone on the VCS Control does not have a search rule configured.
Answer: D
QUESTION NO: 121 Options are Changed
Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio
codec; E. T1 PRI
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.
the same audio codec on both legs of the call
B.
an FXO card
C.
a binding IP address
D.
a hardware transcoder -LTI Transcoder Resource
E.
DSP resources -DSP PVDM Resource
F.
a T1 card
Answer: D,E
QUESTION NO: 122 As per dumps A,C is wrong A & B is the Correct answer
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.
DX-650
B.
Cisco Jabber Desktop
C.
CP-7965
D.
EX-60
E.
MX-200
Answer: A,C
QUESTION NO: 123 I selected B,E
I think is better to go with A, E as per previous discussion here
Refer to the exhibit.
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versa.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A.
The Cisco EX60 default gateway of user is missing from the network configuration.
B.
The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.
The Cisco EX60 of user is not responding to requests coming from the TMS server.
D.
The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E.
The router does not have a route back from the DMZ to the internal network.
Answer : B,E
QUESTION NO: 124
Which three messages does a Cisco VCS use to monitor the Presence status of endpoints?
(Choose three.) Options are reduced in Exam (A. start-call; B. end-call; C. call-started; D.
registration ) Answer is D
A.
start-call
B.
in-call
C.
end-call
D.
call-ended
E.
call-started
F.
registration
Answer: B,D,F
Reference:
http://www.cisco.com/c/en/us/td/docs/telepresence/infrastructure/articles/vcs_monitors_presen
ce_
status_endpoints_kb_186.html
QUESTION NO: 125 In Dumps it is A,B,E But “ voice register global dn” I think it’s not a
valid command.
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A.
Configure voice register pool.
B.
Configure voice register global dn.
C.
Configure an SRST reference.
D.
Configure a phone NTP reference.
E.
Configure the SIP registrar.
F.
Configure telephony service.
Answer: A,C,E
QUESTION NO: 126
Which three options are overlapping parameters for roaming when a device is configured for
Device Mobility? (Choose three.)
A.
device pool
B.
location
C.
network locale
D.
codec
E.
MRGL
F.
extension
Answer: B,C,E
Explanation:
The overlapping parameters for roaming-sensitive settings are Media Resource Group List,
Location, and Network Locale. The overlapping parameters for the Device Mobility-related
settings
are Calling Search Space (called Device Mobility Calling Search Space at the device pool),
AAR
Group, and AAR Calling Search Space. Overlapping parameters configured at the phone have
higher priority than settings at the home device pool and lower priority than settings at the
roaming
device pool.
Reference: https://supportforums.cisco.com/document/77096/device-mobility
QUESTION NO: 127
An engineer is working on a Cisco VCS Control routing configuration and wants users to be
able
to dial ccnpcollab and have calls routed to [email protected]. Which option achieves this
aim?
A.
search rules
B.
transforms
C.
access rules
D.
call policy
Answer: B
QUESTION NO: 128 In Dumps it is MGCP call park feature is not available in MGCP and in SCCP
call park feature is available so Correct Answer is SCCP.
An administrator is setting up analog phones that connect to a Cisco VG310. Which type of
gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the
administrator set up to allow the phones to have the call pickup feature?
A.
H.323 gateway
B.
SCCP gateway
C.
H.225 trunk
D.
MGCP gateway
E.
SIP trunk
Answer: B
QUESTION NO: 129 BFCP is Correct answer. In dumps D is Wrong answer.
An engineer must enable video desktop sharing between a Cisco Unified Communications
Manager registered video endpoint and a Cisco VCS registered video endpoint. Which protocol
must be enabled in SIP profile for VCS SIP trunk on Cisco Unified Communications Manager?
A.
RDP
B.
B. H.264
C.
C. H.224
D.
H.263
E.
BFCP
Answer: E
QUESTION NO: 130
An engineer must resolve a call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
A.
lack of audio bandwidth
B.
lack of video bandwidth
C.
lack of transcoding resources
D.
lack of audio or video bandwidth
E.
lack of conferencing resources
Answer: A
QUESTION NO: 131 In Exam Option was modified
While troubleshooting a connectivity issue between Cisco Unified Communications Manager,
Expressway-C, and Expressway-E, an engineer sees this output in the Expressway-E logs.
Event=”Authentication Failed” Service=”SIP” Src-ip=”10.50.2.1”
Src-port=”25723” Detail=”Incorrect authentication credential for user”
Protocol “TLS” Method=”OPTIONS” Level=”1”
What is the cause of this issue?
A.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Zone username/password.
B.
The Expressway-C Traversal Client username/password do not match the Expressway-E
Traversal Server username/password.
C.
The Expressway-C Traversal Client Zone username/password do not match the Expressway-E
Traversal Zone username/password.
D.
The Expressway-C Traversal Zone username/password do not match the Expressway-E
TraversalClient username/password.
E.
The Expressway-C Traversal Server username/password do not match the Expressway-E
Traversal Client username/password.
Answer: B
QUESTION NO: 132 I Selected AAR but as per dumps.Some have suggested “AAR” & “LRG” Please
research & answer.
An engineer is performing an international multisite deployment and wants to create an
effective
backup method to access TEHO destinations in case the call limit triggers. Which technology
provides this ability?
A.
AAR
B.
CFUR
C.
LRG
D.
SRST
Answer: C
QUESTION NO: 134 As per dumps B,D is wrong. I think D,E is correct answer.
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A.
Resolve names outside of the direct control of the Cisco VCS that exist on the public
Internet.
B.
Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C.
Traverse a firewall from a protected network to a public or DMZ network.
D.
Apply registration, authentication, and media encryption policies.
E.
Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of
bandwidth.
Answer: D,E
QUESTION NO: 135
Which situation requires TCP port 443 to be open for packets that are sourced from the
Internet
with a destination in the corporate DMZ?
A.
when video endpoints that reside on the Internet require administrative access to the Cisco
Expressway Edge
B.
when you require encrypted calls to endpoints on your corporate LAN
C.
when you want to enable calls to web applications by using HTTP
D.
when you require administrative access to the Cisco Expressway Edge from the Internet
Answer: D
QUESTION NO: 136
Refer to the exhibit.
Which option describes the effect of this configuration?
A.
It implements Cisco Unified CME redundancy.
B.
It configures a standby Cisco Unified E.
C.
It configures failover.
D.
It implements Cisco IOS redundancy.
E.
It creates dial peers.
F.
It implements HSRP.
Answer: A
QUESTION NO: 137
Which two types of trunks can support Cisco Unified Communications Manager? (Choose two.)
A.
switch port trunks
B.
PIMG trunks
C.
SIP trunks
D.
H.225 trunks
E.
CO trunks
F.
POTS trunks
Answer: C,D
QUESTION NO: 138
Which three tests can you perform to verify redundancy in the customer environment? (Choose
three.)
A.
Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is
disconnected.
B.
Verify that all phones are registered to a second subscriber server.
C.
Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D.
Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E.
Verify that media resources fail over to a secondary subscriber server when the publisher
fails.
F.
Verify that the H.323 redundant connection is active.
Answer: A,B,E
QUESTION NO: 139 I selected C. I think "A" will better comparing with "C"
Which option indicates the best QoS parameters for interactive video?
A.
0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B.
1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C.
1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D.
5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Answer: C
QUESTION NO: 140
Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A.
a calling search space
B.
hosted DN patterns
C.
translation patterns
D.
route patterns
E.
the SIP or H.323 trunk
F.
hosted DN groups
Answer: B,E,F
QUESTION NO: 141
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200
Which two types of devices are affected when an engineer changes the DSCP for TelePresence
Calls service parameter? (Choose two.)
A.
a Cisco Jabber Desktop
B.
DX-650
C.
CP-7965
D.
MX-200
E.
EX-60
Answer: D,E
QUESTION NO: 143
Which three devices or applications support call preservation? (Choose three.)
A.
a software conference bridge
B.
Cisco Unified IP Phone 7962G
C.
an annunciator
D.
SIP trunks
E.
JTAPI applications
F.
TAPI applications
G.
CTI applications
H.
third-party H.323 endpoints
Answer: A,B,D
QUESTION NO: 144 in dumps C is Wrong correct answer is B
An engineer is configuring Global Dial Plan Replication and wants to prevent the local
cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes
this task?
A.
Create a block route pattern.
B.
Create a block learned pattern.
C.
Create a block transformation pattern.
D.
Create a block translation pattern.
Answer: B
QUESTION NO: 145
An administrator is visiting a remote site that has on-net calls with headquarters and one
voice
gateway for PSTN calls. When using RTMT to monitor the bandwidth utilization of the remote
site,
the administrator notices the OutOfResources counter for the site in LBM has been increasing
slowly in last two weeks, but no call failure reports have been sent from this site. Which
description
about this issue is true?
A.
The bandwidth settings of the site are fulfilling on-net call volume.
B.
AAR is routing some of the calls.
C.
The location-based CAC does not work properly.
D.
The LBM service is malfunctioning.
Answer: B
QUESTION NO: 146 in Exam options are reduced to choose 2
I selected G,C from the below options
An engineer is configuring URI calling within the same cluster. Which four actions must be
taken to
accomplish this configuration? (Choose four.)
A.
Configure SIP route patterns.
B.
Configure the directory URI partition and calling search space.
C.
Associate the directory URIs to directory numbers.
D.
Activate the URI service in Cisco Unified Serviceability.
E.
Configure SIP trunk.
F.
Assign directory URIs to users.
G.
Configure the SIP profile.
H.
Configure the URI service parameters.
Answer: B,C,F,H
QUESTION NO: 148 I Selected A,E
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site.
During a network failure between the remote site and the central office, some of the phones
located at the remote site are unable to make phone calls. Which two options are potential
causes
of the problem? (Choose two.)
A.
The site has exceeded the number of SRST endpoints supported by the voice gateway.
B.
The ccm-manager fallback command is configured incorrectly on the voice gateway.
C.
Phones at the remote site are assigned to the incorrect device pool.
D.
The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E.
The site has exceeded the number of simultaneous calls allowed in SRST mode.
Answer: A,E
QUESTION NO: 149
Options Modified B,D Is the Correct answer from modified option
(A. h.323 fast start;
B. IPV6 -IPV4 transform;
C. DTMF inband RTP-NTE (rfc2833);
D. Sip delay offer )
Which function can be implemented without MTP resources?
A.
DTMF relay conversion
B.
terminating a media stream that uses the same codec
C.
music on hold
D.
SIP early offer
Answer: B
QUESTION NO: 150
An engineer is setting up a Cisco VCS Cluster with SIP endpoints only. While configuring the
Cisco VCS peers, which signaling protocol is used between peers to determine the best route
for
calls?
A.
SIP
B.
H.323
C.
SCCP
D.
MGCP
Answer: B
Reference:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X87/Cis
co-VCS-Cluster-Creation-and-Maintenance-Deployment-Guide-X8-7.pdf
(page 4, basic Configuration is done, third point)
QUESTION NO: 151 Please check the documents I think SAF will not be a dialing function.
Which three globalization dialing functions are enhanced in Cisco Unified Communications
Manager 7.x and later? (Choose three.)
A.
MGRL
B.
TEHO
C.
CER
D.
AAR
E.
SAF
F.
click-to-call
Answer: B,C,D
QUESTION NO: 152 One command is to verify SCCP phone and another command to view SIP IP
phones In dumps it wrong answer C,E is Correct answer
Which two commands verify Cisco IP Phone registration? (Choose two.)
A.
show telephony-service ephone-dn
B.
show voice register session-server
C.
Show ephone registered
D.
showccm-manager hosts
E.
show sip-ua status registrar
Answer: C,E
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/troubleshooting/guide/ts_phreg.html
(see the steps)
QUESTION NO: 153 I selected F . DNS Server
In Exam options are reduced to choose 1,
Refer to the exhibit.
An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP
address and the system name. After logging into the Cisco VCS Expressway admin page, the
engineer sees this output. Which four options must be configured to complete the Cisco VCS
Expressway system configuration? (Choose four.)
A.
NTP server
B.
SIP server
C.
LDAP server
D.
security certificate
E.
DHCP server
F.
DNS server
G.
SIP URI
H.
Cisco Unified Communications Manager IP address
Answer: A,C,D,F
QUESTION NO: 154
After forgetting to log out of his IP phone in the main office, an Extension Mobility user is
unable to
log in to a different IP phone at a remote office. Which option is a possible reason for the
problem?
A.
The phone at the remote location is a different model than the phone in the user’s main
office.
B.
The user’s Extension Mobility profile is misconfigured.
C.
The user can log in to only one device at a time.
D.
The device pool is misconfigured.
Answer: C
QUESTION NO: 155
Which solution is needed to enable presence and extension mobility to branch office phones
during a WAN failure?
A.
SRST without MGCP fallback
B.
SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C.
SRST with MGCP fallback
D.
Cisco Unified Communications Manager Express in SRST mode
Answer: D
QUESTION NO: 157
An engineer is configuring a new DX-80 in Cisco Unified Communications Manager. Where can an
engineer verify the default DSCP value of AF41?
A.
enterprise phone configuration
B.
common phone profile
C.
service parameters
D.
enterprise parameters
Answer: C
QUESTION NO: 158
Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit
numbers. Which option should be configured if you want outgoing calls from 4-digit internal
directory number to be presented as a 10-digit number?
A.
calling party transformation pattern
B.
AAR group
C.
translation pattern
D.
route pattern
Answer: A
QUESTION NO: 159
Which three configuration settings are included in a default region configuration? (Choose
three.)
A.
Immersive Bandwidth
B.
Video Call Bandwidth
C.
Audio Codec
D.
Link Loss Type
E.
Real Time Protocol
F.
Location Description
Answer: B,C,D
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/bccm-
801cm/b02regio.html#wp1077135
QUESTION NO: 160 In Exam Options to choose 1.
I selected "A"
Refer to the exhibit.
Which three statements about when user A calls user using SIP are true? (Choose three.)
A.
SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B.
Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking
option key.
C.
Cisco VCS Control and Cisco VCS Expressway support static NAT.
D.
Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
E.
RT and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F.
The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
Answer: A,B,E
QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose
three.)
A.
The SIP route patterns have not been properly configured.
B.
The Tomcat certificates do not match.
C.
The Cisco Unified Resource Identifier service needs a restart.
D.
The ILS authentication password does not match.
E.
The cluster ID does not match.
F.
One cluster is using TLS certificate, and the other is using Password.
Answer: B,D,F
Thank for your help.
Can you please also take some time answering this other questions:
¨¨¨¨¨¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints
inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.
Dumb select option: "...The access control list on the VCS Control must..."
I saw other users selecting option D.
Traversal zone is for external call, that mean outside network. Local zone is for local network. in this case the question is for "Outside Call". I think D is correct answer.
I selected "The traversal zone on the VCS Control does not have a search rule configured" but I failed the exam.
Not sure which is the correct one.
¨¨¨¨Question 123¨¨¨¨¨¨
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user B and vice versa.
User A can hear user C, however user C cannot hear user A.
User B can hear user C, however user C cannot hear user B.
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user C is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user C is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E.The router does not have a route back from the DMZ to the internal network.
My Answer : C and E but I failed the exam.
¨¨¨¨¨¨QUESTION 125¨¨¨¨¨¨
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.
Seeing a lot of conflicting answers from everyone.
Dump answer: B / D / E
I selected options: BCE.
Just refer to the SRST admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf
page 190
But I failed the exam, not sure which are the corrects, any thoughts?
****QUESTION 128******
An administrator is setting up analog phones that connect to a Cisco VG310. Which type of gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the administrator set up to
allow the phones to have the call pickup feature?
A. H.323 gateway
B. SCCP gateway
C. H.225 trunk
D. MGCP gateway
E. SIP trunk
Dump: H323 GW
But I selected option: SCCP Gateway
Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-series-gateways/product_data_sheet09186a00801d87f6.html - Table 1
But I failed the exam, not sure which is the correct, any thoughts?
****QUESTION 134******
Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two.)
A. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet.
B. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities.
C. Traverse a firewall from a protected network to a public or DMZ network.
D. Apply registration, authentication, and media encryption policies.
E. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.
Dumb answer: B D
My final answer: C and D
Reason: Per Cisco's VCS Administrator guide, "Subzones are used to control the bandwidth used by various parts of your network, and to control the VCS's registration, authentication, and media encryption
policies." So D for sure. E maybe, however it seems tricky because I don't think you can manage bandwidth limits based on the type of endpoint (like standard definition). Bandwidth is managed based on
Within subzone, to/from other subzones, and total bandwidth of all calls. I like C better - the traversal subzone is utilized when calls need to traverse a firewall (when VCS Expressway is deployed).
But I failed the exam, not sure which is the correct, any thoughts?
****QUESTION 135******
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a
destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway
Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
Answer provided: B
I selected: D
Reason: Per Cisco's MRA Deployment guide, 443 is opened from internet to DMZ only for administrative access to VCS Expressway (which is strongly discouraged). See firewall port reference on the following
guide: http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-5/Mobile-Remote-Access-via-VCS-Deployment-Guide-X8-5-2.pdf
But I failed the exam, not sure which is the correct, any thoughts?
****QUESTION 140******
Which three items must you configure to enable SAF Call Control Discovery ( choose three)
A. Calling Search Space
B. Hosted DN Groups
C. Translation Patterns
D. The SIP or H323 trunk
E. Hosted DN Patterns
F. Route Patterns
Dump answer: A / D / E
I selected below answerS:
B. Hosted DN Groups
D. The SIP or H323 trunk
E. Hosted DN Patterns
But I failed the exam, not sure which is the correctS, any thoughts?
****QUESTION 142******
A presales engineer is working on a quote for a major customer and must evaluate how many Cisco VCS Expressway traversal call licenses for which to plan. Calls to and from which three routes must the
engineer include in the tally? (Choose three.)
A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS
Dump: C E F
I selected answer : C, D and F
For a VCS Expressway, calls to or from a traversal client (Firewall traversal calls). Traversal clients include ""other VCSs(F), gatekeepers(D), Border Controllers(C), or traversal-enabled endpoints.
Reference : http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html
But I failed the exam, not sure which is the correctS, any thoughts?
****QUESTION 144******
An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes this
task?
A.Create a block route pattern.
B.Create a block learned pattern.
C.Create a block transformation pattern.
D.Create a block translation pattern.
Dump answer: C
I selected option: D Create a block translation pattern.
But I failed the exam, not sure which is the correct, any thoughts?
****QUESTION 148******
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phones
located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two.)
A.The site has exceeded the number of SRST endpoints supported by the voice gateway.
B.The ccm-manager fallback command is configured incorrectly on the voice gateway.
C.Phones at the remote site are assigned to the incorrect device pool.
D.The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
What will be the correct answer of this.
I selected A and E.
Reason:
1. Cisco 2821 voice bundle with PVDM2-32, SRST featuring 48-phone license,
2. If BCD are correct then some phones will not work.
But I failed the exam, not sure which is the corrects, any thoughts?
****QUESTION 149******
Which function can be implemented without MTP resources? SELECT TWO
A.DTMF relay conversion
B. terminating a media stream that uses the same codec
C. Multicast music on hold
D.SIP early offer
E.IPV4 to IPV6 conversation
Dump answer: B terminating a media stream....
I selected option C "Multicast music on hold" and E "IPV4 to IPV6 conversation"
But I failed the exam, not sure which is the corrects, any thoughts?
****QUESTION 152******
Which two commands verify Cisco IP Phone registration? (Choose two.)
A.show telephony-service ephone-dn
B.show voice register session-server
C.show ephone registered
D.show ccm-manager hosts
E.show sip-ua status registrar
Dump answer: B C
But I believe that is incorrect:
A: false, this command is used to see the phone confiuration, not registration.
B: false, command do not exists.
C. Correct, for SCCP phones.
D. False, the command is used for MGCP gateway
E. Correct, used to display all SIP endpoint registered.
So I selected: C and E:
for reference:
Step 3 show sip-ua status registrar
Use this command to display all the SIP endpoints currently registered with the contact address.
From http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080557e81.html
But I failed the exam, not sure which is the corrects, any thoughts?
****QUESTION 153******
An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP address and the system name. After logging into the Cisco VCS Expressway admin page, the engineer sees this
output.
Which four options must be configured to complete the Cisco VCS Expressway system configuration? (Choose four.)
A. NTP server
B. SIP server
C. LDAP server
D. security certificate
E. DHCP server
F. DNS server
G. SIP URI
H. Cisco Unified Communications Manager IP address
If have to mark one answer, I think the right is: D - DNS Server
this questions wasnt in my exam.
****QUESTION 154******
After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the
problem?
A.The phone at the remote location is a different model than the phone in the user’s main office.
B.The user’s Extension Mobility profile is misconfigured.
C. The user can log in to only one device at a time.
D.The device pool is misconfigured.
Dump answer: A
I selected option: C.
reasons:
A: does not matter, if the phone is a different model, it will take the default profile for its type (I remember there is also another question in the dump regarding this).
B: Don't think so, because he was able to log in to another phone.
C: Yes, true, because of the "Multiple login allowed" parameter, in the extension mobility parameters
D: Inconsistent question. Device pool does not impact Extension Mobility.
For reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsgd-861-cm/fsem.html
•You can set the service parameter to allow for multiple logins. If you set multiple login not allowed, Cisco Extension Mobility supports only one login at a time for a user. Subsequent logins on other
devices will fail until the user logs out on the first device
****QUESTION 155******
Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?
A. SRST without MGCP fallback
B. SRST with VoIP dial peers to Cisco Unified Communications Manager Express
C. SRST with MGCP fallback
D. Cisco Unified Communications Manager Express in SRST mode
DUMP ANSWER: C
I selected option: D CME in SRTS mode.
reasoning:
http://www.ciscopress.com/articles/article.asp?p=1744068&seqNum=4
CUCME in SRST Mode Usage
Examples of features that are provided only by CUCM Express in SRST mode are Call Park, Presence, Cisco Extension Mobility, and access to Cisco Unity Voice Messaging services using SCCP.
Correct Answer: None!
Reason: This one is tricky.. the closest answer is D since presence and extension mobility are both CUCME features, however while in SRST these enhanced features are not supported. I will pick D if I get
this question, but hopefully this is one of the "not graded" questions...
I selected D but I failed the exam, not sure if that is the correct one. Any thoughts?
I failed for the second time the test yesterday, with a 847 score.
I obtain the same % score for question categories 1, 3, 4 and 8, so definitively I am wrong on some answers.
Please help.
1- VCS Control: 50%
2- Collaboration Edge(VCS Expressway): 63%
3- Configure CUCM Video Service parameters: 86%
4- Describe and Implement Centralized Call Processing Redundancy: 60%
5- Describe and configure a Multi-Site Dial Plan for CUCM: 80%
6- Implement Call Control Discovery/ILS: 100%
7- Implement Video Mobility Features: 100%
8- Implement Bandwidth Management and Call Control on CUCM: 57%
Q1: A
Q7: A,C,E
Q9: A,D
Q13: D
Q14: A,C
Q15: B,D
Q16: D
Q17: C
Q22: A,C
Q34: A,D
Q35: A,C
Q48: D
Q75: B
Q84: A,C,D
Q91: A,B
Q93: B
Q95: B
Q96: B
Q103 B
Q104: C
Q115: Transform (both search rules and transform were proposed)
Q116: C,D
Q117: A,B,F
Q118: D (How many CUCM nodes can a skinny phone establish an SCCP connection to at the same time?)
Q121: A,B (A. PVDM or DSP resource; B. LTI local transcode resource; C. ref2833; D. one audio codec; E. T1 PRI card)
Q122: A,B
Q123: A,E (Someone could give the right answer?????)
Q124: D (A. start-call; B. end-call; C. call-started; D. registration)
Q125: A,B,C
Q126: B,C,E
Q127: search rules (both search rules and transform were proposed)
Q128: B
Q129: E
Q130: A
Q131: B
Q132: A
Q134: B,D
Q135: D
Q136: C
Q137: C,D (A was H246 Video Trunk, I should chose A because Trunk H323 is not really supported it is Trunk H225)
Q138: A,B,E (Someone could give the right answer?????)
Q139: A (appears to be best compare to C)
Q140: B,E,F
Q141: D,E
Q143: A,B,D
Q144: B
Q145: B
Q146: D,E (A. Configure SIP route patterns; B. Activate the URI service in Cisco Unified Serviceability; C. Configure SIP trunk; D. Assign directory URIs to users; E. Configure the SIP profile)
Q148: A,E
Q149: B,C (A. h.323 fast start; B. IPV6 -IPV4 transform; C. DTMF inband RTP-NTE (rfc2833); D. delayed offer h.323)
Q150: B
Q152: C,E
Q153: B (A. LDAP server; B. DNS server; C. Cisco Unified Communications Manager IP address; D. DHCP server)
Q154: C
Q155: D
Q157: C
Q158: A
Q159: B,C,D
Q160: C (Someone could give the right answer?????)
Q161: B,D,F
Answer: C
A Cisco Unified Communications Manager Group specifies a prioritized list of up to three Cisco Unified
Communications Managers. The first Cisco Unified Communications Manager in the listserves asthe primary
Cisco Unified Communications Manager for that group, and the other members of the group serve assecondary
and tertiary (backup) Cisco Unified Communications Managers
REF:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100/CUCM_BK_C95ABA82_00_admin-guide-100_chapter_0100.pdf
PAGE 1
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 ??
When considering CUCM failover, how many backup servers can be configured in a CUCM Group?
The answer says 3, and of course you can configure 3 servers on a CUCM group, but only 2 of them will be backup servers.
Am I wrong?
Answer is A, C and E.
Reference : http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_8_9_and_X7-2.pdf
Which three statements about configuring an encrypted trunk between Cisco TelePresence Video Communication Server and Cisco Unified Communications Manager are true? (Choose three.)
A. The root CA of the VCS server certificate must be loaded in Cisco Unified Communications Manager.
B.A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP.
C. The Cisco Unified Communications Manager trunk configuration must have the destination port set to 5061.
D. A SIP trunk security profile must be configured with Device Security Mode set to TLS.
E.A SIP trunk security profile must be configured with the X.509 Subject Name from the VCS certificate.
F. The Cisco Unified Communications Manager zone configured in VCS must have SIP authentication trust mode set to On.
G. The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set to Off.
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.
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.
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.
What steps are needed to set up h.323 to SIP and vise versa, pick two.
a. Protocol>Sip>Sip on
b. H.323-SIP internetworking mode On
c. H.323-SIP internetworking mode Off
d. H.323-SIP internetworking mode Registered Only
e. Protocol>H.323>On
f. Protocol>Sip>configuration>Sip On
An engineer must resolve a video call failure issue. When using RTMT, the engineer notices that the
Location Bandwidth Manager-OutOfResources counter is showing a positive value. Which option
is the cause of the call failure?
A.lack of audio bandwidth
B.lack of video bandwidth
C.lack of transcoding resources
D.lack of audio or video bandwidth
E.lack of conferencing resources
2 new questions were:
2 Types of Trunks in CUCM
A. H.246 Trunks
B. SIP Trunks
C. H.343 Trunks
D. MGCP Trunks
E. CO Trunks
F. POTS Trunks.
I Chose A. H.246 Trunks and B. SIP Trunks but I think Answer should be SIP Trunks and CO Trunks. Any ideas?
QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password
I chose B,D,F, not sure if this was right.
QUESTION 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish this configuration? (Choose four.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.
I Chose E and F, option C was not an option in the test.
Which 2 things do not utlise MTP
A. H.323 fast start
B. IPV6 -IPV4
C. DTMF inband RTP-NTE (rfc2833)
D. delayed offer h.323
I chose A and B. I think A is wrong.
Can someone share the right answers for this?
I will later add all my answers like Steve.
About questions 115 e 127??
In the two questions I think the right answer is TRANSFORM.
And how about your opinion?
These are the answers I chose that differed from yours
Q115- Transform
Q116- B, C
Q122- DX650 and Jabber. SRND guide does say Jabber does pull DSCP info from CUCM
Q123- A, E
Q124- in-all, call-ended, registration
Q125- voice register pool, voice register global dn, SIP registrar
Q127- I chose Transform but I feel like I should have chosen Search Rules
Q128- SCCP
Q129- BFCP
Q132- AAR just because it asks “Call Limit”
Q135- it only asks for 1 and I chose A
Q136- C Failover
Q137- SIP and H323. Gatekeeper control uses H.323. If tests asks for H.225 I will chose that
Trunk Types in Cisco Unified Communications Manager Administration
Your choices for configuring trunks in Cisco Unified Communications Manager depend on whether the IP WAN uses gatekeepers to handle call routing. Also, the types of call-control protocols that are used in the call-processing environment determine trunk configuration options.
You can configure the trunk types in Cisco Unified Communications Manager Administration listed in this section.
• H.225 Trunk (Gatekeeper Controlled)
• Intercluster Trunk (Gatekeeper Controlled)
• Intercluster Trunk (Non-Gatekeeper Controlled)
• SIP Trunk
Q138- A, B, E
Q139- C
Q140- B, E, F
Q144- C – block transformation pattern
Q145- B AAR is routing some of the calls
Q146. I believe this one is tricky but it does say within the same cluster. I chose Sip route patterns and assign URI’s to users
Q148- A and E
Q152 – C and E
Q153- DNS
Q154 – C
Q158- A Calling party transformation pattern
Q160- I chose E (anyone that passed please correct)
Q161 – B, D, F
•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
Which three items must you configure to enable SAF Call Control Discovery ( choose three)
A. Calling Search Space
B. Hosted DN Groups
C. Translation Patterns
D. The SIP or H323 trunk
E. Hosted DN Patterns
F. Route Patterns
Verified Answer
B,D,E
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
which pass4 you used?
Good Look
First of all you are mentioning about the dump ete 300-075 161q file??
QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.
QUESTION NO: 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
120: D
the main prob is "unable to receive calls from outside endpoints".
A. is the solution for unable to calls from inside to outside endpoints
C. you need to add search rule too
123: E
(the exam only ask for 1 answer)
*135,138,160: same
Did you use the original answer of the DUMP or answer by or self? Regards
A. default gateway
B. MCU
C. TMS
D. DNS
E. Gatekeeper
My guess on this is TMS, Telepresence Management Server
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
Could it be C and D? I read through several Cisco docs (I didn't have the actual servers to log into)...it looks like yes "registered only" is better than "on". Definitely C is the answer, but as per the cisco guide, the sip mode is on by default and stays on (we don't turn if off on any step).
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.
"
What are the tasks required to route calls from H323 to SIP and viceversa?
A. Config-protocols-Interworking-On
B. Config-protocols-Interworking-Off
c. Config-protocols-Interworking-registered only
D. Config-protocols-Sip-Config-Mode-On
e. Config-protocols-H323-H323Mode-On
why the answer should be AE?
I suspect it to be A and C"
What you're looking at in the answers is the pathway you would take in the VCS-E menu to make configuration changes.
The interworking config is, by default, set to "Registered Only." One of the earlier posters stated they chose A & E, but based on the question breakdown given at the end of the exam and the percentage they received, they believe the A&E answer is wrong.
Given that the question asks what is REQUIRED configuration, not optimal configuration (which IMO would be 'On', I believe one of the answers to be C.
Both of you said A & C, but given how configuration is handled in VCS-E, it has to be one or the other, Configuration cannot be set to both On and Registered Only.
I believe the second answer to this question is E, H323. When you hover over the information Icon under H323 config in VCS-E, it reads "Determines whether or not the VCS will provide H.323 gatekeeper functionality."
When you do the same for SIP, it reads "Determines whether or not the VCS will provide SIP registrar and SIP proxy functionality.
This mode must be enabled in order to use either the Presence Server or the Presence User Agent."
Therefore, I believe the correct answer to be C and E.
Best of Luck to you buddy!! Keep us posted!! I am taking the exam for the 3rd time on the 29th.
Good luck.
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
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
160 is B
161 is one answer A
125 IS B,C,E
The bad news is that I failed with a 797!
I did not go verbatim on what the dump said as there seemed to be many wrong or questionable. Those I had researched and I answered what I found to be "correct" from the research. Looks like I should have followed the dump.
The results showed this:
VCS Control 50%
Collaboration Edge: 63%
Configure CUCM Video Parameters: 100%
Central Call Processing Redundancy: 60%
Multi-site Dial Plan: 63%
CCD/ILS: 67%
Video Mobility Features: 83%
Bandwidth Mgmt and CAC: 57%
Here's the breakdown of which questions I got and what I answered, anyone care to shed some light on which ones were the ones I got wrong?
1:A
9:A,D
13: D
14:A,C
15:B,D
16: D
17: G
22: A,C
27: A,B,E
34: A,D
35: A,C
79: B
84: A,C,D
88: A,B,C
91: A,B
93: B
95: B
96: B
103: B
104: C
109: C
114: E
115: B
116: B,C
117: A,B,F
118: D. Question actually said :How many CUCM Nodes can a skinny phone establish an SCCP connection to at the same time"
119: B
120: B
121: D,E. D was Listed as a specific type of transcoding on the IOS, A was just One Audio Codec, no B
122: A,B
123:C,E
124: F. Pick 1, had End-Call, Call-Started, Registration, one more
125: A,B,C
126: B,C,E
127: D
128: D
129: E
130: A
131: B
132: A
134: B,D
135: A
136: A
137: C,D
138: A,B,E
139: C
140: A,E,F
141: D,E
143: A,B,D
144: B
145: B
146: C,D. Choose 2, no F or H
148: A,E
149: Can't recall, Pick Two, had DTMF Relay, MOH, SIP Early Offer, h323 early offer, one more
150: B
152: B,C
153: F. Pick 1 out of 4, had NTP, LDAP, DNS and CUCM IP
154: C
155: D
156: C
157: C
158: D
159: B,C,D
160: B. Pick 1 out of 4, had A,B,C and F
161: B,D,F
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)
Can you advise to what question #'s you are referring to in your reply back to me?
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
@Hari: Like Scotty says you need to check "Use Fully Qualified Domain Name in SIP Requests" in sip profile (the default is unchecked).
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?
one last set, @king @fen @Sunday if you can share your thoughts and comments. i will be taking the exam next week.
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
Answer: B
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which component
allows for standardized caller addresses between the endpoints?
A.search rules
B.SIP route pattern
C.policy service
D.transform
Answer: D
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user and vice versA.
User A can hear user C, however user cannot hear user A.
User can heat user C, however user cannot hear user .
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E.The router does not have a route back from the DMZ to the internal network.
Answer : A E
An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications
Manager and enables the option “Use Fully Qualified Domain Name” in SIP Requests. Which result is
achieved by enabling this option?
A.Resolve FQDN using DNS type SRV record.
B.Resolve FQDN using DNS type A record.
C.Ensure FQDN is used in SIP Identity header.
D.Ensure FQDN is used in SIP Request header.
Answer: D
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with
a destination in the corporate DMZ?
A.when video endpoints that reside on the Internet require administrative access to the Cisco
Expressway Edge
B.when you require encrypted calls to endpoints on your corporate LAN
C.when you want to enable calls to web applications by using HTTP
D.when you require administrative access to the Cisco Expressway Edge from the Internet
Answer: D
Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer
Answer C
An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from
routing the Vice President number 5555555555 to the remote cluster. Which action accomplishes this
task?
A.Create a block route pattern.
B.Create a block learned pattern.
C.Create a block transformation pattern.
D.Create a block translation pattern.
Answer: D
Your company’s main number is 408-526-7209, and your employee’s directory numbers are 4-digit
numbers. Which option should be configured if you want outgoing calls from 4-digit internal directory
number to be presented as a 10-digit number?
A.calling party transformation pattern
B.AAR group
C.translation pattern
D.route pattern
Answer A
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.
What are your answers on the below 5 questions.
QUESTION 120
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone
is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The traversal zone on the VCS Control does not have a search rule configured.
B. The access control list on the VCS Control must be updated with the IP for the external users.
C. When a traversal zone is set up on VCS Control only outbound calls are possible.
D. The local zone on the VCS Control does not have a search rule configured.
QUESTION 123
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user В and vice versa.
User A can hear user C, however user С cannot hear user A.
User В can heat user C, however user С cannot hear user В.
Which two properties are the most likely reasons for this issue? (Choose two.)
A. The Cisco EX60 default gateway of user С is missing from the network configuration.
B. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C. The Cisco EX60 of user С is not responding to requests coming from the TMS server.
D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control.
E. The router does not have a route back from the DMZ to the internal network.
QUESTION 135
Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ?
A. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge
B. when you require encrypted calls to endpoints on your corporate LAN
C. when you want to enable calls to web applications by using HTTP
D. when you require administrative access to the Cisco Expressway Edge from the Internet
QUESTION 138
Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
B. Verify that all phones are registered to a second subscriber server.
C. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
F. Verify that the H.323 redundant connection is active.
QUESTION 160
Refer to the exhibit.
Which three statements about when user A calls user С using SIP are true? (Exam asks only one)
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Deploying a Cisco VCS Expressway behind a NAT mandates the use of the Advanced Networking option key.
C. Cisco VCS Control and Cisco VCS Expressway support static NAT.
D. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.
E. RTР and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
F. The NAT device must translate from 10.X.X X to 193.1.1.X and vice versa.
When provisioning for Interactive Video (IP Videoc onferencing) traffic, the following guidelines are recommended:
• Interactive Video traffic should be marked to DSCP AF41; excess Interactive-Video traffic can be marked down by a policer to AF42 or AF43.
• Loss should be no more than 1 %.
• One-way Latency should be no more than 150 ms.
• Jitter should be no more than 30 ms.
• Overprovision Interactive Video queues by 20% to accommodate bursts Because IP Videoconferencing (IP/VC) includes a G.711 audio codec for voice, it has the same loss, delay, and delay variation requirements as voice, but the traffic patterns of videoconferencing are radically different from voice.
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 NO: 160
Which statement about when user A calls user C using SIP is true?
A. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa.
B. Cisco VCS Control and Cisco VCS Expressway support static NAT.
C. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking
option key.
D. RTP and RTCP ports must be opened at the firewall from internal to DMZ and vice versa
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?
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.
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.
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?
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.
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A.the same audio codec on both legs of the call
B.an FXO card
C.a binding IP address
D.a hardware transcoder
E.DSP resources
F.a T1 card
Answer: D,E
My Understanding : Same
Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer
Answer: B
My Understanding is B and C
Reason for C: "The following restriction exists for multicast music on hold (MOH) When an MTP resource gets invoked in a call leg at a site that is using multicast MOH, the caller receives silence instead o music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH"
Reference : http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsgd-712-cm/fsmoh.html
I used q161, questions are correct but answer are incorrect. They selected the questions that we have doubts, they are posting the difficult questions :-S
I tried to do a research with the official cisco documents and trying to find the correct answers and I used those answers today but I still failed the exam.
We need to work together to pass this exam.
We need assistance from people that passed the exam recently to please provide the answers they used.
PLEASE HELP in below questions:
*****QUESTION 68****
What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM endpoints? (Choose two.)
A Media Resource Group List.
B.Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
C.Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS.
D.Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM.
E.Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
Dump answer
C. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified C
I used below answers in the exam (not sure if it was good):
B. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS
E. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM
****QUESTION 80******
What is the correct value to use for the "DSCP for TelePresence Calls" Cisco CallManager service parameter?
A. 28 (011100)
B. 34 (100010)
C. 41 (101001)
D. 46 (101110)
Answer: B
Q161 indicates A
I used B 34 but not sure if it was correct.
******QUESTION 98******
Which functionality does ILS use to link all hub clusters in an ILS network?
A. Fullmesh
B. Automesh
C. ILS updates
D. multicast
DUMP ANSWER: B
What is the correct one?
EXPLAIN:
When a new hub cluster registers to another hub cluster in an existing ILS network, ILS automatically creates a full mesh connection between the new hub cluster and all the existing hub clusters in the ILS network.
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/configure___intercluster_lookup_service.pdf
question no 160 was in the exam but you need to select only one answer.
******QUESTION 109******
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only
use G.711 between regions.
Answer: B
161Q says the Correct answer is C but I think the Correct answer is B
I selected B but I failed the exam.
¨¨¨¨¨¨QUESTION 120¨¨¨¨¨¨
Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.
Dumb select option: "...The access control list on the VCS Control must..."
I saw other users selecting option D.
Traversal zone is for external call, that mean outside network. Local zone is for local network. in this case the question is for "Outside Call". I think D is correct answer.
I selected "The traversal zone on the VCS Control does not have a search rule configured" but I failed the exam.
Not sure which is the correct one.
¨¨¨¨¨¨QUESTION 122¨¨¨¨¨¨
The Cisco Unified Communications system of a company has five types of devices:
•Cisco Jabber Desktop
•CP-7965
•DX-650
•EX-60
•MX-200
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
read "Set DSCP Values". so answer should be A,B.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_6/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide_appendix_01111.html
A. The SAF Client Control is a configurable inherent component of Cisco Unified Communications Manager.
B. The SAF Client Control is a non-configurable inherent component of Cisco Unified Communications Manager.
C. The SAF Client Control is a non-configurable inherent component of the Cisco IOS Routers.
D. The SAF Client Control is a configurable inherent component of the Cisco IOS Routers.
Is B??
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 ?
Thank you so much for this and congrats on passing, its going to be very helpful for this horrible test
can you confirm the 161 dumps you used that had the wrong answers was from Actual tests or was this the Exam collection file?
Asking as both these vendors have different answers even to the ones outside of the answers you posted?
Also could you tell me which answers you used for the below questions?
QUESTION 146
An engineer is configuring URI calling within the same cluster. Which four actions must be taken to accomplish
this configuration? (Choose four.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Associate the directory URIs to directory numbers.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.
Answer: As people are saying the exam only asks for two answers, was it C and F?
QUESTION 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A. Implement local failover.
B. Implement SIP to POTS.
C. Load-balance PRI connections.
D. Load-balance route lists within the cluster.
E. Implement ICT trunks to remote locations.
F. Implement centralized failover.
Correct answer A and F?
QUESTION 148
You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. During a network
failure between the remote site and the central office, some of the phones located at the remote site are unable
to make phone calls. Which two options are potential causes of the problem? (Choose two.)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway.
C. Phones at the remote site are assigned to the incorrect device pool.
D. The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
Answer A and E?
Thanks a lot
Ramesh
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?
QUESTION NO: 160
-you choose 4 answers or 4 answers but you need to choose 3?
There will be 4 options only and need to choose 1.
QUESTION NO: 153
-NTP Server, SIP Server, Security Cert and DNS. will NTP be the best answer?
I think the correct answer is DNS Server in Exam
QUESTION NO: 136
-Should be the failover?
Yes Failover
QUESTION NO: 127
-my guess is search rule?
-No I think Transforms
QUESTION NO: 123
-Can you explain your answer to this.
-As I said Can't remember but the answer might be IOS router.
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.
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?
C. The user can log in to only one device at a time.
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 "
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?
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.
Thank you
@ Akira
Q153
DNS
Server Certificate (Under security certificate)
NTP Server
SIP Domains
So, DNS is first choice
Q127
Search rules - @domain
Transforms - @example.com
I concur transforms as right answer
Q111
B. the destination alias, without the domain portion
Both Q127 & Q111: http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Single-VCS-Control-Deployment-Guide-X8-2.pdf
Still searching for Q160 and Q123
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).
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?
A. The traversal server zone on Expressway-C must have a TLS verify subject name configured.
B. The traversal client zone and the traversal server zone Media encryption mode must be set to Force encrypted.
C. The traversal client zone and the traversal server zone Media encryption mode must be set to Auto.
D. The traversal client zone on Expressway-C Media encryption mode must be set to Auto.
E. The traversal client zone and the traversal server zone must be set to SIP TLS with TLS verify mode set to On.
Correct Answer: BE ????
A. Resolve FQDN using DNS type SRV record.
B. Resolve FQDN using DNS type A record.
C. Ensure FQDN is used in SIP Identity header.
D. Ensure FQDN is used in SIP Request header.
Is D??
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.
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
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
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
I am still looking for the correct answers of the below questions, I am planning to place the exam next week..
QUESTION NO: 43
Which component is needed to set up SAF CCD?
A.SAF-enabled H.323 intercluster (gatekeeper controlled) trunk
B.SAF forwarders on Cisco routers
C.Cisco Unified Communications cluster
D.SAF-enabled H.225 trunk
------
QUESTION NO: 109
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A.G.711 requires 128K of bandwidth per call.
B.G.729 requires 24K of bandwidth per call.
C.The default codec does not matter if you have defined a hardware MTP in your Cisco Unified
Communications Manager environment.
D.To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and
only use G.711 between regions.
-------
QUESTION NO: 115
A voice engineer is enabling video capabilities between H.323 and SIP endpoints. Which
component allows for standardized caller addresses between the endpoints?
A.search rules
B.SIP route pattern
C.policy service
D.transform
-------
QUESTION NO: 147
Which two steps must you take when implementing TEHO in your environment? (Choose two.)
A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.
-------
QUESTION NO: 161
When configuring intercluster URI dialing, an engineer gets the error message “Local cluster
cannot connect to the ILS network”. Which three reasons for this error are true? (Choose three.)
A.The SIP route patterns have not been properly configured.
B.The Tomcat certificates do not match.
C.The Cisco Unified Resource Identifier service needs a restart.
D.The ILS authentication password does not match.
E.The cluster ID does not match.
F.One cluster is using TLS certificate, and the other is using Password
Please you help
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
Could you please share your thought on the exam. is 161Q still valid.
I'm taking the test for the 3rd time tomorrow hopefully. These were the last scores I got
VCS Control - 63
VCS Expressway - 50
CUCM Video - 100
Centralized Call - 60
Multi-Site - 88
ILS - 67
Mobility - 83
CAC - 57
Lets try to work together and get past this horrible test
¨¨¨¨QUESTION 122¨¨¨¨¨¨
Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
My Answer: A,B
Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two.)
A. the same audio codec on both legs of the call
B. an FXO card
C. a binding IP address
D. a hardware transcoder
E. DSP reources
F. a T1 card
Note: Dump answer are D & E. But upon searching a hardware transcoder is different from hardware MTP.
Hardware MTP configured on Cisco IOS routers:
+DSP resources are required. Configure this MTP type by using the maximum session hardware
+Use of the same audio codec but different packetization on both call legs is possible.
__Question 139___
Which option indicates the best QoS parameters for interactive video?
A. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
C. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
Note: Dump answer is C but other said that it should be A?
__Question 148___
You have deployed a Cisco 2821 ISR to poerform as an SRST voice gateway at a remote site. During a network failure between the remote site and the central office, some of the phone located at the remote site are unable to make phone calls. Which two options are potential causes of the problem? (Choose two)
A. The site has exceeded the number of SRST endpoints supported by the voice gateway.
B. The ccm-manager fallback command is configured incorrectly on the voice gateway
C. Phone at the remote site are assigned to the incorrect device pool
D. The ccm-manager fallback-mgcp cimmand is configured incorrectly on the voice gateway.
E. The site has exceeded the number of simultaneous calls allowed in SRST mode.
Note: Dump answer are B & D but other said that the answer are A & E.
__Question 154__
After forgetting to log out of his IP phone in the main office, an Extension Mobility user is unable to log in to a different IP phone at a remote office. Which option is a possible reason for the problem?
A. The phone at the remote location is a different model than the phone in the user's main office.
B. The user's Extension Mobility profile is misconfigured
C. The user can log in to only one device at a time.
D. The device pool is misconfigured.
Note: Dump answer is A but other said that the answer is C.
___Question 155___
Which solution is needed to enable presence and extension monility to branch office phones during a WAN failure?
A. SRST without MGCP fallback
B. SRST with VOIP dial peers to CME
C. SRST with MGCP fallback
D. CUCM Express in SRST mode
Note: Dump answer is C but other said that the answer is D.
Any advise guys? I will take the exam this coming Monday. Thanks!
My friend thanks a lot for sharing the right questions!!!!
You mean that all other questions from 161q are correct and only the ones you posted are wrong??
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
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 [email protected]. 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?
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 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!
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..
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
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
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
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.
Should be A and F.
What must the administrator configure on the firewall to stabilize this deployment?
A. The VCS Control should not be on the LAN, but it must be located in the DMZ with the Expressway.
B. The firewall at Company X must have all SIP ALG functions disabled.
C. The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the same network that the VCS Control is on.
D. A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway and the VCS Control servers.
Correct Answer: B????
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
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
Here is the difference between my answers and your answers
115 B
116 B, C
121 B, E
125 B,C,E
127 B
135 B
136 C
139 C
146 F, C
151 B,C,D
153 F
160 B
161 A
An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms:
User A can hear user B and vice versa.
User A can hear user C, however user C cannot hear user A.
User B can hear user C, however user C cannot hear user B.
Which two properties are the most likely reasons for this issue? (Choose two.)
A.The Cisco EX60 default gateway of user C is missing from the network configuration.
B.The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ.
C.The Cisco EX60 of user C is not responding to requests coming from the TMS server.
D.The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS
Control.
E.The router does not have a route back from the DMZ to the internal network.
E is definite. However I feel like no other answer applies. I suppose A would be the next best answer but here are my thoughts:
A. In my opinion if the EX60 didn't have a default gateway, then I would think the call would never have been setup in the first place, hence we wouldn't even be talking about one way audio.
B. The port direction they are describing is backwards, otherwise it would be right.
C. I don't see what the TMS has to do with one way audio.
D. If there was no response to SIP INVITE then the call wouldn't be setup, hence no one way audio.
I didn't use ete, used pdf instead.
@Jpb and chase
These are my answers and rest the same.
Q116: My Ans: B,C
Q122: My Ans: A,C
Q123: My Ans: B,E
Q127: My Ans: B
Q134: My Ans: D,E
Q137: My Ans: H.323 trunk , SIP trunk
Q139: My ANs: C
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).
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
@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
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?
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?
I have same opinion for 123, 135,138 and 160.
Q123: A and E
Q135: D
Q138: A B E
Q160: A
But Q120: D, Local Zone we will use for internal calls in the VCS. For external calls we need to have a traversal zone.
We have 2 options with traversal Zone(A&C).
C. When a traversal zone is set up on VCS Control only outbound calls are possible.(Not outbound only, Both inbound and outbound are possible)
So i feel it is A.
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?
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.
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
I noticed that you passed the exam yesterday with 897.
Can you please see my latest post with several questions and let us know how you answer those questions in your exam?
You score 897 which means most of those question you answered properly so we really appreciate you can tell us how to answer those questions.
Please PLEASE Please help help.
congrats!
Appreciate if you can share the list of questions with their correct answers with us.
thank you!
Correct answers of 161 are
The Tomcat certificates do not match.
One cluster is using TLS certificate, and the other is using Password.
The ILS authentication password does not match.
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
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.
Thanks.
Failed today with 723 :(
FYI. The 161q is still valid. I just passed my exam with a score 869. :)
This was my second try, cause I faild my first attempt with 835.
I've got a recommendation for all of you, who are using dump to prepare for the exam. ALWAYS READ ALL ANSWERS CARFULLY! For example. Q136, where A is marked for right answer. Have use seen, that Cisco UNITED CME is written and not UNIFIED!
One more thing. Most of the questions were from the last part of the dump (~60).
I wish you all good luck!
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.
Explain:
ILS uses automesh functionality to create a full mesh connection between all hub clusters within an ILS network.
When a new hub cluster registers to another hub cluster in an existing ILS network, ILS automatically creates a full mesh connection between the new hub cluster and all the existing hub clusters in the ILS network.
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
A. You do not need to configure search rules for traversal calls.
B. You need to configure the firewall to allow communication from the Cisco VCS Expressway to the Cisco VCS Control.
C. The username on the Cisco VCS Control and Cisco VCS Expressway are local and do not need to match.
D. The Cisco VCS Expressway is the Traversal Server.
D IS RIGHT??
Can anyone please advise on recent exam results?
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?
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?
Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones?
(Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.
Seeing a lot of conflicting answers from everyone. Here's what I will choose: BCE. Just refer to the SRST admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf
QUESTION 132
An engineer is performing an international multisite deployment and wants to create an effective backup
method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. CFUR
C. LRG
D. SRST
I originally thought A, but now I agree with others that C is best answer. Within context of TEHO, you should make sure LRG (Local Route Groups) are utilized as backup when WAN is down/CAC limit is reached.
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
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!
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!
Thanks for clarification. I will try to exam shortly...
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
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...
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
Can anyone confirm if it still valid? Please update us all. 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?
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.
A. the full URI, including the domain portion
B. the destination alias, without the domain portion
C. the E.164 number that is assigned to the Cisco TelePresence EX90
D. the directory number that is assigned to the Cisco TelePresence EX90
Answer: B is right??
What is your plan for retake?
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
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.
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
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
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 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.
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)
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.
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
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?
Thank you!
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
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
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?
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.
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!
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?
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
@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
Used dump from Passleader Q355
Good luck everyone!
Does someone tried the new 161Q, is it valid?
The 239Q is still valid. Only 5 new:
VCS Control: 100%
Collaboration Edge: 100%
Configure CUCM Video Service Parameters: 100%
Describe an Implement Centralizaed Call Processing Redundancy: 70%
Describe and Configure a Milti-Site Dial Plan for CUCM: 100%
Implement Call Control Discovery/ILS: 100%
Implement Video Mobility Features: 83%
Implement Banwidth Management and CAC on CUCM: 100%
new questions
1.a trace RTMT logs (about no enough bandwidth)
ANS. location out of resource
2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,Annunciator
3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway
4.Question about normalization global call routing. 3 answers
not remember options:
I marked 3 about "gateways" word (I failed here, I think)
5. Scenario: CUCM in HQ, Branch with srst+PSTN+DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-calls
d) The service phone does not working.
not remember more...
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
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.
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
All questions are in the 239q dump.
Few Q&A are inconsistent, I strongly recommend to comment (there is a button to add a comment in the test)
Some exhibits are not displayed correctly (exactly as shown in the 239q dump : you cannot read the ip addresses on the picture). I recommend to do a request to the local VUE-PROMETRICS representative as a technical incident, and add a comment in the exam.
If you fails, please open a case to Cisco www.cisco.com/go/certsupport and ask for your exam to be reviewed: you may be refunded with a voucher
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.
The file 239q is 100% VALID, I passed the exam few days ago, all questions been from dump !
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.
Thanks Guys
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
Thanks for the clarification on Question 123.
When are you planning to take next exam.?
Thanks
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)
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
239q still valid about 6-7 new questions
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
new questions
1.a trace RTMT logs (location out of resource)
ANS. Out of bandwidth is the only valid answer.
2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,Annunciator (Need to verify)
3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway (Need to verify)
4.Question about normalization global call routing. (normalization is on the Q239)
5. Scenario: CUCM in HQ, remote site iwth end points, Branch with srst+PSTN+DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-branch calls
(I gotta look into this one, but HQ would definitely have issues if they called the remote site. Branch will work on SRST, but would not be able to call the remote site.)
Any body has information about updated Dump 160?
any body try it?
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.
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
Dump 239 is still valid, scored 9XX , 5 new questions.
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
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?
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
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
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.
with the 239Q dump as it is valid.
Did anyone passed after 25-Sep-2017?
Please Update. Thanks...
A. AAR
B. SRST
C. CFUR
D. LRG
Pls. What is the correct answer?
CCNP Collab now !
239Q 110% vaild.
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
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?
Anyone tryed this one?
Please.
Could you tell us yr experience.
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?
D. The traversal zone on the VCS Control does not have a search rule configured.
Did anyone tryed this one?
Hope you have a note of the correct answers, Could you share with us.
Thanks
Sultan Al Arif
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!
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
Thanks,
Rahul
Some research on the new q's and wanted to share, mostly agree with David SS but hopefully the explanations below will help everyone out
1.a trace RTMT logs (about no enough bandwidth)
ANS. location out of resources
Explanation
Two potential counters here: OutOfResources, LocationOutOfResources
2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,SIP TRUNK
Explanation
Call Preservation
The call preservation feature of Cisco Unified Communications Manager ensures that an active call does not get interrupted when a Cisco Unified Communications Manager fails or when communication fails between the device and the Cisco Unified Communications Manager that set up the call.
Cisco Unified Communications Manager supports full call preservation for an extended set of Cisco Unified Communications devices. This support includes call preservation between Cisco Unified IP Phones, Media Gateway Control Protocol (MGCP) gateways that support Foreign Exchange Office (FXO) (non-loop-start trunks) and Foreign Exchange Station (FXS) interfaces, and, to a lesser extent, conference bridge, MTP, and transcoding resource devices.
Enable H.323 call preservation by setting the advanced service parameter, Allow Peer to Preserve H.323 Calls, to True.
The following devices and applications support call preservation. If both parties connect through one of the following devices, Cisco Unified Communications Manager maintains call preservation:
Cisco Unified IP Phones
SIP trunks
Software conference bridge
Software MTP
Hardware conference bridge (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Transcoder (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Non-IOS MGCP gateways (Catalyst 6000 24 Port FXS Analog Interface Module, Cisco DT24 , Cisco DE30 , Cisco VG200)
Cisco IOS H.323 gateways (such as Cisco 2800 series, Cisco 3800 series)
Cisco IOS MGCP Gateways (Cisco VG200, Catalyst 4000 Access Gateway Module, Cisco 2620, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3810)
Cisco VG248 Analog Phone Gateway
The following devices and applications do not support call preservation:
Annunciator
H.323 endpoints such as NetMeeting or third-party H.323 endpoints
CTI applications
TAPI applications
JTAPI applications
3.call preserve on cisco unified SRST on sip phone. Where to configure:
ANS:SIP Trunk in CUCM, voice service voip /preserve in gateway
Explanation
First Part
1. Navigate to the trunk device > trunk.
2. Select new or existing trunk
3. Check "SRTP Allowed - When this flag is checked, Encrypted TLS needs to be configured in the network to provide end to end security. Failure to do so will expose keys and other information.
Consider Traffic on This Trunk SecureRequired Field"
4. Select either when using bot sRTP or TLS, when using sRTP
Second Part
1. enable
2. configure terminal
3. voice service voip
4. srtp fallback
5. allow-connections sip to h323
6. allow-connections sip to sip
7. end
4.Question about normalization global call routing. 3 answers
not remember options:
I marked 3 about "gateways" word (I failed here, I think)
Not quote enough info here so maybe reading of this will help, most normalization is done by the gateway but some are also available on trunks
More in-depth detail
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fsgd-851-cm/fscallpn.html
5. Scenario: CUCM in HQ, Branch with srst PSTN DID. what happens if the WAN is down?
Options:
a) Users in HQ not known about the problem
b) User in Branch can to make calls with no problem
c) User in Branch can not to make intra-calls
d) The service phone does not working.
not remember more...
Thinking B from the selection given is correct as SRST provides remote site redundancy during a WAN failure
Heard something about SAF ports
Question about saf port:
a: tcp port 5050
Explanation
The service family used is IPv4 and TCP port 5050. The keepalive timer for the Cisco SAF external client is optionally set to 360,000 milliseconds (the default is 9600 milliseconds).
service-family external-client listen ipv4 505
external-cliet client-a
username safuser
password safpass456
keepalive 360000
King: good luck keep us posted :)
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.
The dumps Ace.102q, Ahmad.91q and Virgil.93q are totally invalid.
I made this test at 09/12/2017 and I didn't pass it, 736 on 860. About 30-35 new questions of 52 questions appear in this test. Only the questions about configurations (DX950, jabber, and so on) are still maintained.
Please, if anyone has the new dump and can send to us as soon as possible, I appreciate a lot.
Thanks.
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
[email protected]
dimka2000 at yahoo.com
new questions
1.a failed call and RTMT logs (no enough band width)
ANS. i believe it is location out of resource
2.application or device that preserve call
a.CTI,TAPI,JTAPI,SIP TRUNK,Announcator,software conferece bridge,ip phone
ANS. Software conference bridge,ip phone,Annunciator
3.call preserve on cisco unified SRST on sip phone
SIP Trunk, i don't remember read about it.
next week i am going to attempt this exam using 114Q file! can any one confirm is it still valid? did any one pass this exam after 15th Dec?
Thank you in advance
anyone tried it ?
Which dumps did study?. Is the different from Q239.
Thanks.
My self have already failed this test twice , so please, anyone passed the exam recently that can clarify this ambiguity?
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.
Dump 239 is still valid, scored 944 , new questions as mentioned by David SS
I need to enter the exam and schedule for a test but i need to make sure if there is a valid dump or not
There was the 114Q and 277Q then a new 180Q then the newest 180Q
Are all of them invalid or what is our next action here ?
with the 239Q dump as it is valid
The 239Q dump this really valid?
@Everyone, anyone confirm if this new dump ETE File this valid?
Thanks for advance
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
there is a new 239Q dump that people says it is valid, two persons advised that they have passed using it
Can someone please update us with the exam status. Are these dumps still Valid..?
Please Reply. Thanks...
Only 4-5 questions were new.
How can Cisco not displayed on Practice..?
Bad news first: I failed today (28/01/) working with the 109q + 114q + 180q.
Good news now: As far as I can remember a significant part of the missing questions are in the 239q that I got too late. It looks like the dump version has just been updated.
Please share you experience, I need to pass it before mid-February.
I got a score of 860. Passing score was 860. In the print out from the exam center, they have written that I have passed. When one gets 860/860 is that a pass or can Cisco change the exam result to a Fail?
most of the exam questions are from ~180-239 range
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?
can you please send me lateset dumps at [email protected]
my CCNP voice cert is expiring in next week ,need to give CIPTV2 in this week..please help me.
I failed by 744 score
dump file q239 steel valid.
The 114 and 146 Q&A are not valid many new questions (10-15) present on the exam
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
I looked at the 'newer' 180q dump, and even though it says version 6.0, it looks like it is even worse than the older dump I used. It looks like it has a bunch of much older questions from CIPT2 exam.
If you look at the exam objectives, 25% of the questions are about VCS-C and VCS-E, and also integration between VCS and CUCM. Your dump does not seem to have those questions.
Also, the several big scenario questions (presenting mixed VCS/CUCM environment) with multiple sub-questions from 114q ActualTest were still on the exam, while 180q dump does not have them.
I'd say this is a bad/outdated dump and will not work for the 300-075
can you pls send me updated dumps at [email protected]
My expiratory of ccnp voice cert is due in next week I desperately need to give CIPTV2 as soon as possible.Please help me
Can you please confirm which dump u've used ?
Exams questions are written in unclear manner and it's impossible to pass!
This is quite weird/embarrassing to know that out of the Whole WORLD not even a Single Collaboration Expert is a REAL WORTHY COLLAB ENGINEER..? including me (^_^)
Even Dump Experts are helpless...to be able to help us. Anyways Guys give it a try and keep us posted. Thanks...
- Cisco Aspirant.
But I think in combination with the 102q file it will be possible to pass the exam. Mine is on February the 2nd.
Thanks a lot
what are the new questions if any?
my exam is tomorrow.
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
Some Questions remembered, not on Q277 or any other:
SAF Port number usage : UDP 5050
Fill in the blank on intercluster and intracluster : not sure of the answer
Anyone know others missing?
I have the 180 Q with me.Just wanted to check whether it is accurate.Anyone sitting for the exam before feb 2nd ?
Regards. and have a greate day.
with the 239Q dump as it is valid
Can you please send me the latest dumps also at
[email protected]
Many Thanks !!
please some buddy share with as the 114 Q
[email protected]
[email protected]
I used 102q
Please share it to me also [email protected]
Thanks
with the 239Q dump as it is valid
Please, can you send me the new dump in pdf?
mailpersonal999 at gmail.com
Thank you!
Any update if the exam is still valid?
Thanks.
Please share your opinions/Experiences/Suggestions with me. Thanks...
- Aspirant.