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Q71. How is the region assigned to a device such as an IP phone?
A. Regions are assigned directly in the device configuration page.
B. Regions can be assigned only through a device pool.
C. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device pool region configuration takes precedence.
D. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device region configuration takes precedence.
Answer: B
Q72. Which statement best describes globalized call routing in Cisco Unified Communications Manager?
A. All incoming calling numbers on the phones are displayed as an E.164 with the + prefix.
B. Call routing is based on numbers represented as an E.164 with the + prefix format.
C. All called numbers sent out to the PSTN are in E.164 with the + prefix format.
D. The CSS of all phones contain partitions assigned to route patterns that are in global format.
E. All phone directory numbers are configured as an E.164 with the + prefix.
Answer: B
Explanation:
Incorrect Answer: A, C, D, E For the destination to be represented in a global form common to all cases, we must adopt a global form of the destination number from which all local forms can be derived. The + sign is the mechanism used by the ITU's E.164 recommendation to represent any PSTN number in a global, unique way. This form is sometimes referred to as a fully qualified PSTN number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1153205
Q73. Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for international calls to the U.S. Assuming the PSTN does not accept globalized numbers with + prefix. What should the Called Party Transformation Pattern at the U.S. gateway be configured as?
A. \+.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: +
B. \+1.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
C. \+408.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: 1
D. \+1408.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
E. \+1.408! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
Answer: D
Q74. When Cisco Extension Mobility is implemented, how is the audio source for the MOH selected?
A. The audio source that is configured at the user device profile is selected.
B. The audio source that is configured at the home phone of the user is selected.
C. The audio source that is configured at the physical phone used for the Cisco Extension Mobility login is selected.
D. The audio source that is configured in the IP Voice Media Streaming parameters is selected.
Answer: A
Explanation:
Incorrect Answer: B, C, D To specify the audio source that plays when a user initiates a hold action, choose an audio source from the User Hold MOH Audio Source drop-down list box from device profile configuration settings. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06dvprf.html
Q75. To preserve analog calls in an MGCP switchback event, which three commands must be configured in the MGCP fallback router? (Choose three.)
A. h323
B. mgcp-switchback-graceful
C. voice service voip
D. mgcp-graceful
E. preserve-h323
F. no h225 timeout keepalive
Answer: A,C,F
Explanation:
Incorrect Answer: B, D, E these additional command for call preservation when using MGCP fallback: voice service voip h323 no h225 timeout keepalive Reference. CCNP Voice CIPT2 642-457 Quick Reference, 2nd Edition
Q76. Which method can be used to address variable-length dial plans?
A. Overlap sending and receiving.
B. Add a prefix for all calls that are longer than 10-digits long
C. Use nested translation patterns to eliminate inter-digit timeout
D. Use the @macro on the route pattern
E. Use MGCP gateways, which support variable-length dial plans
Answer: A
Explanation:
Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsintrcm.htm
Q77. For which VoIP protocol does a gatekeeper provide address translation and control access?
A. H.323
B. SIP
C. Skinny
D. H.248
Answer: A
Q78. When an H.323 trunk is added for Call Control Discovery, which statement is true?
A. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Enable SAF check box should be selected in the trunk configuration.
B. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Trunk Service Type should be Call Control Discovery.
C. The H.323 trunk is added by selecting Call Control Discovery Trunk and then selecting
H.323 as the protocol to be used.
D. The H.323 trunk is added by selecting H.323 Trunk, and selecting Inter-Cluster Trunk as the Device Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF.
Answer: A
Explanation:
Reference. Implementing Cisco Unified Communications Manager Part 2 (CIPT2), Chapter3: Implementing Multisite Connections, pg 70-71, Fig 3-14 and Fig 3-15
Q79. What is the difference between an H.323 gateway and a SIP gateway?
A. An H.323 gateway requires that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. The H.323 gateway can be added in Cisco Unified Communications Manager under gateway type as H.323 Gateway. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk.
C. A SIP gateway requires a call agent for PSTN calls to be placed and received. An H.323 gateway does not require a call agent for PSTN calls to be placed and received.
D. An H.323 gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The H.323 gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The SIP gateway must be configured in Cisco Unified Communications Manager using the domain name.
Answer: B
Q80. Which two statements about SAF service identifier numbers are true? (Choose two.)
A. They are generated in the format service:sub-service:instance.instance.instance.instance.
B. They are 16-bit decimal identifiers.
C. They are generated in the format data-source:sub-service:instance.matrix.fifty.saf.
D. They are 32-bit decimal identifiers.
E. They are generated in the format data.saf.cucm-publisher.asf@domain.local.
F. They are generated in the format telco.cisco.saf-forwader.db.replicate.data.local.
Answer: A,B