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QUESTION 41
Which two statements are requirements regarding hunt group options for B-ACD implementation on Cisco Unified Communications Manager Express routers? (Choose two.)

A.    The ephone hunt group is mandatory.
B.    Either the ephone hunt group or the voice hunt group is acceptable.
C.    Hunt group members must be SCCP IP phones.
D.    Hunt group members can include both SCCP or SIP IP phones.
E.    Hunt group members must be SIP IP phones.
F.    The member hunting mechanism must be set to sequential.

Answer: AC
Explanation:
The ephone hunt group is mandatory, and while ephone hunt groups only support Cisco Unified SCCP IP phones, a voice hunt group supports either a Cisco Unified SCCP IP phone or a Cisco Unified SIP IP phone
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_v1ht.html

QUESTION 42
Which call hunt mechanism is only supported by the voice hunt group in a Cisco Unified Communications Manager Express router?

A.    sequential
B.    peer
C.    longest idle
D.    parallel
E.    overlay

Answer: D
Explanation:
Parallel Hunt-Group, allows a user to dial a pilot number that rings 2-10 different extensions simultaneously. The first extension to answer gets connected to the caller while all other extensions will stop ringing. A timeout value can be set whereas if none of the extensions answer before the timer expires, all the extensions will stop ringing and one final destination number will ring indefinitely instead. The final number could be another voice hunt-group pilot number or mailbox The following features are supported for Voice Hunt-Group:
Calls can be forwarded to Voice Hunt-Group
Calls can be transferred to Voice Hunt-Group
Member of Voice Hunt-Group can be SCCP, ds0-group, pri-group, FXS or SIP phone/trunk
Max member of Voice Hunt-Group will be 32

QUESTION 43
Which Cisco Unified Communications Manager Express ephone button configuration separator enables overflow lines when the primary line for an overlay button is occupied by an active call?

A.    o
B.    c
C.    w
D.    x
E.    :

Answer: D
Explanation:
x expansion/overflow, define additional expansion lines that are used when the primary line for an overlay button is occupied by an active call.

QUESTION 44
Which two statements describe characteristics of Cisco Unified Border Element high availability, prior to Cisco IOS release 15.2.3T, using a box-to-box redundancy configuration? (Choose two.)

A.    It leverages HSRP for router redundancy and GLBP for load sharing between a pair of routers.
B.    Cisco Unified Border Element session information is check-pointed across the active and standby router pair.
C.    It supports media and signal preservation when a switchover occurs.
D.    Only media streams are preserved when a switchover occurs.
E.    It can leverage either HSRP or VRRP for router redundancy.
F.    The SIP media signal must be bound to the loopback interface.

Answer: BD

QUESTION 45
Refer to the exhibit. From this NFAS-enabled T1 PRI configuration on a Cisco IOS router, how many bearer channels are available to carry voice traffic?

451

A.    91
B.    92
C.    93
D.    94
E.    95

Answer: D
Explanation:
In NFAS one channel is used for signaling so according to this we will have 94 channel for with bearer capability.

QUESTION 46
Refer to the exhibit. Assuming this NFAS-enabled T1 PRI configuration on a Cisco IOS router is fully functional, what will the controller T1 1/1 D-channel status be in the output of the show isdn status command?

461

A.    MULTIPLE_FRAME_ESTABLISHED
B.    TEI_ASSIGNED
C.    AWAITING_ESTABLISHMENT
D.    STANDBY
E.    INITIALIZED

Answer: B
Explanation:
TEI_ASSIGNED, which indicates that the PRI does not exchange Layer 2 frames with the switch. Use the show controller t1 x command to first check the controller t1 circuit, and verify whether it is clean (that is, it has no errors) before you troubleshoot ISDN Layer 2 problem with the debug isdn q921.

QUESTION 47
Refer to the exhibit. In an effort to troubleshoot a caller ID delivery problem, a customer emailed you the voice port configuration on a Cisco IOS router. Which type of voice port is it?

471

A.    FXS
B.    E&M
C.    BRI
D.    FXO
E.    DID

Answer: D

QUESTION 48
The iLBC codec operates at 38 bytes per sample per 20-millisecond interval. What is its codec bit rate in kilobits per second?

A.    6.3
B.    13.3
C.    15.2
D.    16
E.    24

Answer: C
Explanation:
The internet Low Bit Rate Codec (iLBC) is designed for narrow band speech and results in a payload bit rate of 13.33 kbits per second for 30-millisecond (ms) frames and 15.20 kbits per second for 20 ms frames. When the codec operates at block lengths of 20 ms, it produces 304 bits per block, which is packetized as defined in RFC 3952. Similarly, for block lengths of 30 ms it produces 400 bits per block, which is packetized as defined in RFC 3952.The iLBC has built-in error correction functionality to provide better performance in networks with higher packet loss

QUESTION 49
Assume 6 bytes for the Layer 2 header, 1 byte for the end-of-frame flag, and a 40-millisecond voice payload, how much bandwidth should be allocated to the strict priority queue for five VoIP calls that use a G.729 codec over a multilink PPP link?

A.    87 kb/s
B.    134 kb/s
C.    102.6 kb/s
D.    77.6 kb/s
E.    71.3 kb/s

Answer: A
Explanation:
Voice payloads are encapsulated by RTP, then by UDP, then by IP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technology in use by each router interface: A single voice call generates two one-way RTP/UDP/IP packet streams. UDP provides multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering.

QUESTION 50
Assume 20 bytes of voice payload, 6 bytes for the Layer 2 header, 1 byte for the end-of-frame flag, and the IP, UDP, and RTP headers are compressed to 2 bytes, how much bandwidth should be allocated to the strict priority queue for six VoIP calls that use a G.729 codec over a multilink PPP link with cRTP enabled?

A.    80.4 kb/s
B.    91.2 kb/s
C.    78.4 kb/s
D.    69.6 kb/s
E.    62.4 kb/s

Answer: D
Explanation:
Voice payloads are encapsulated by RTP, then by UDP, then by IP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technology in use by each router interface: A single voice call generates two one-way RTP/UDP/IP packet streams. UDP provides multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering.

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QUESTION 21
Refer to the exhibit. A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP Dial Rule with a pattern value of 2001, which statement about the call setup process of this call is true?

211

A.    Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML
event, and Cisco Unified Communications Manager will extend the call as soon as the collected digits
match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.
B.    Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML
event. When the collected digits match the extension of the SCCP IP phone, Cisco Unified Communications
Manager
will extend the call only if the class of service configuration on both phones permits this action.
C.    As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service
configuration on both IP phones.
D.    As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call only if class of service configuration on both phones permits this action.
E.    The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of
service configuration on both IP phones.

Answer: D
Explanation:
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.

QUESTION 22
What does a comma accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified Communications Manager?

A.    It inserts a 500-millisecond pause between digits.
B.    It causes the phone to generate a secondary dial tone.
C.    It is a delimiter and has no significant dialing impact.
D.    It indicates a timeout value of 5000 milliseconds.
E.    It is an obsolete parameter and will be ignored.

Answer: B
Explanation:
Comma is accepted in speed dial as delimiter and pause. -Comma used to delineate dial string, FAC, CMC, and post connect digits For post connect digits, commas insert a 2 second delay Commas may be duplicated to create longer delays

QUESTION 23
Which Call Admission Control mechanism is supported for the Cisco Extension Mobility Cross Cluster solution?

A.    Location CAC
B.    RSVP CAC
C.    H.323 gatekeeper
D.    intercluster Enhanced Location CAC
E.    visiting cluster’s LBM hub

Answer: B
Explanation:
Configuring extension mobility cross cluster (EMCC) is nothing you should take lightly. EMCC requires a lot of configuration parameters including the exporting and importing of each neighbor cluster’s X.509v3 digital certificates. EMCC is supported over SIP trunks only. Presence is another feature that’s only supported over SIP trunks. If you want to be able to perform scalable Call Admission Control (CAC) in a distributed multi- cluster call processing model, you will need to point an H.225 or Gatekeeper controlled trunk to an H.323 Gatekeeper for CAC… but if you want to support presence and EMCC between clusters and maintain CAC.

QUESTION 24
Which two Cisco Unified Communications Manager SIP profile configuration parameters for a SIP intercluster trunk are mandatory to enable end-to-end RSVP SIP Preconditions between clusters? (Choose two.)

A.    Set the RSVP over SIP parameter to Local RSVP.
B.    Set the RSVP over SIP parameter to E2E.
C.    Set the SIP Rel1XX Options parameter to Disabled.
D.    Set the SIP Rel1XX Options parameter to Send PRACK If 1xx Contains SDP.
E.    Set the SIP Rel1XX Options parameter to Send PRACK for All 1xx Messages.
F.    Check the Fall Back to Local RSVP check box.

Answer: BD
Explanation:
Each Unified Communications Manager cluster and Unified CME should have the same configuration information. For example, Application ID should be the same on each Unified Communications Manager cluster and Unified CME. RSVP Service parameters should be the same on each Unified Communications Manager cluster.

QUESTION 25
What is the number of directory URIs with which a Cisco Unified Communications Manager directory number can be associated?

A.    1
B.    up to 2
C.    up to 3
D.    up to 4
E.    up to 5

Answer: E
Explanation:
Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.

QUESTION 26
Which Cisco Unified Communications Manager partition will be associated with a directory URI that is configured for an end user with a primary extension?

A.    null
B.    none
C.    directory URI
D.    default
E.    any partition that the Cisco Unified Communications Manager administrator desires

Answer: C
Explanation:
Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.

QUESTION 27
Which Call Control Discovery function allows the local Cisco Unified Communications Manager to listen for advertisements from remote call-control entities that use the SAF network?

A.    CCD advertising service
B.    CCD requesting service
C.    SAF forwarder
D.    SAF enabled trunks
E.    CCD registration service

Answer: B
Explanation:
SAF and CCD will allow large distributed multi-cluster deployments to have the directory number (DN) ranges of each call routing element advertised dynamically over SAF. Cisco routers act as SAF Forwarders (SAFF), while the call routing elements (e.g. CUCM) act as clients that register with the routers to advertise their DN ranges and listen to the advertisements of other routers.

QUESTION 28
Which codec complexity mode, when deployed on Cisco IOS routers with DSPs using the C5510 chipset, supports the most G.711 calls per DSP?

A.    Low
B.    Medium
C.    High
D.    Secure
E.    Flex

Answer: E
Explanation:
The flex parameter allows the complexity to automatically adjust to either medium or high complexity depending on the needs of a call. For example, if a call uses the G.711 codec, the C5510 chipset automatically adjusts to the medium-complexity mode. However, if the call uses G.729, the C5510 chipset uses the high complexity mode

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QUESTION 11
Which two responses are examples of client error responses in SIP protocol? (Choose two.)

A.    302 Moved Temporarily
B.    404 Not Found
C.    503 Service Unavailable
D.    502 Bad Gateway
E.    604 Does Not Exist Anywhere
F.    408 Request Timeout

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QUESTION 11
Which two responses are examples of client error responses in SIP protocol? (Choose two.)

A.    302 Moved Temporarily
B.    404 Not Found
C.    503 Service Unavailable
D.    502 Bad Gateway
E.    604 Does Not Exist Anywhere
F.    408 Request Timeout

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QUESTION 1
Which two SCCP call signaling messages are sent by an IP phone to Cisco Unified Communications Manager? (Choose two.)

A.    SoftKeyEvent
B.    OpenReceiveChannelAck
C.    StartMediaTransmission
D.    SelectSoftKeys
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QUESTION 481
An engineer is converting all gateways to SIP and wants to ensure that the device is protected from traffic with malicious intent? Which two parts must the engineer enable to protect the Cisco Unified Communications Manager from SIP attacks on any newly created SIP trunks? (Choose two.)

A.    SIP Station TCP Port Throttle Threshold
B.    Denial – of – Service Protection Flag
C.    SIP TCP Unused Connection Timer
D.    SIP Max incoming Message Headers
E.    SIP Max incoming Message Size
F.    SIP trunk TCP Port Throttle Threshold

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QUESTION 461
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QUESTION 441
Refer to the Exhibit, Which three cisco Unified CM internal extension match the globalization pattern shown and provide a globalized calling party number? (Choose three)

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QUESTION 421
A Call is made between two desk phones enabled with single number reach that are registered to a cisco unified CM cluster. The device pool for each device has a local route group defined. When the call is placed to exit the system, which device pool control the destination gateway?

A.    Destination RDP
B.    Source Phone
C.    Source RDP
D.    Destination phone

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QUESTION 401
A collaboration engineer is designing Cisco IM&P implementation to support instant messaging logging for compliance.
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QUESTION 381
A user is troubleshooting an FXO line on a Cisco IOS router that remains connected even after the call ends. Which three disconnect methods can be configured to fix this problem? (Choose three)

A.    Loop Current Feed Open Signaling Disconnect
B.    Hookflash Duration Signaling Disconnect
C.    CP-TONE Dual Supervisory Disconnect
D.    Power Denial-based Supervisory Disconnect
E.    Ground-start Signaling Disconnect
F.    Tone-based Supervisory Disconnect

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QUESTION 361
A Cisco Unified Contact Center Express manager wants to add database integration to the selfservice interactive voice response application. Which four types of licensing and database servers support this requirement? (Choose four.)

A.    The server must have enhanced licensing.
B.    The server must have premium licensing.
C.    A server running Sybase Adaptive Server is required.
D.    A server running Oracle is required.
E.    A server running Postgress SQL is required.
F.    A server running SAP SQL server is required.
G.    A server running Microsoft SQL server is required.
H.    The server must have standard licensing.

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