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Q31. What is a prerequisite of AAR deployment? 

A. You must have a single distributed call processing deployment. 

B. Calls must be manually rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. 

C. Calls must be automatically rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. 

D. Clustering must be implemented over the WAN. 

E. You must have a centralized call processing deployment. 

Answer:


Q32. 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. 

B. TLC uses its local time for all systems. 

C. TLC queries for the time zone as part of configuration. 

D. TLC produces an error and must be run remotely. 

Answer:


Q33. Scenario 

There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 9971 Video IP Phone. The Cisco VCS and TMS control the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

DP 

Locations 

CSS 

SRST 

SRST-BR2 Config 

BR2 Config 

SRSTPSTNCall 

After configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001), the calls fail from the PSTN. Which two of the following configurations if applied to the router, would remedy this situation? (Choose two.) 

A. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:15 

B. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:13 

C. voice translation-rule 1rule 1 /228821 …$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

D. voice translation-rule 1rule 1 /228822…$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

E. The router does not need to be configured 

Answer: A,D 


Q34. 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. To match the US-TEHO pattern \+!, how should the translation pattern be configured? 

A. 9001.4085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: + 

B. 9.0014085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: +1 

C. 900.14085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: +1 

D. 900.14085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: + 

E. 001.4085551234 with the Called Party Transformation:Prefix Digits Outgoing Calls: + 

Answer:

Explanation: 

Incorrect Answer: A, B, C The PSTN access code for the UK is 9, International call code is 001, The international escape character, +, signifies the international access code in a complete E.164 number format Link: http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03r p.html 


Q35. Which bandwidth amounts are correct for configuring locations? 

A. 8 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722 

B. 8 kb/s for G.729, 64 kb/s for G.711, and 16 kb/s for G.722 

C. 64 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722 

D. 8 kb/s for G.729, 8 kb/s for G.711, and 8 kb/s for G.722 

Answer:


Q36. How is a SIP trunk in Cisco Unified Communications Manager configured for SIP precondition? 

A. The configuration is done by selecting a SIP precondition trunk for trunk type. 

B. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk. 

C. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk. 

D. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. The new SIP profile must then be assigned to the SIP trunk. 

Answer:


Q37. Refer to the exhibit. 

To permit three G.729 calls, what should the bandwidth value be for the ip rsvp bandwidth command? 

A. 32 

B. 48 

C. 64 

D. 88 

E. 128 

Answer:


Q38. Refer to the exhibit. 

All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL. For the HQ phones always to use the hardware conference bridge as a first choice, which configuration should be implemented? 

A. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. Ensure that the instance ID for the hardware conference bridge is 0. 

B. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. The hardware conference bridge must be configured first. 

C. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first. 

D. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Configure an additional HQ_MRGL_2. Add the HQ_MRG to HQ_MRGL. Add HQ_MRG_2 to HQ_MRGL_2. The HQ_MRGL should be assigned to the HQ phones. The HQ_MRGL_2 should be assigned to the HQ device pool. 

Answer:


Q39. Which three commands are necessary to override the default CoS to DSCP mapping on interface Fastethernet0/1? (Choose three.) 

A. mls qos map cos-dscp 0 10 18 26 34 46 48 56 

B. mls qos map dscp-cos 8 10 to 2 

C. mls qos 

D. interface Fastethernet0/1mls qos trust cos 

E. interface Fastethernet0/1mls qos cos 1 

F. interface Fastethernet0/2mls qos cos 1 

Answer: A,C,D 


Q40. Refer to the exhibit. 

IT shows an H.323 gateway configuration in a Cisco Unified Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on the H.323 gateway shows the correct called-party number as 5015552001. Which two configuration changes can correct this issue? (Choose two.) 

A. Add port 1/0:23 to dial-peer voice 123 pots. 

B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4. 

C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the voice translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is left at default (All). 

D. Add the command h323-gateway voip id on interface vlan120. 

E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the Significant Digits for inbound calls to 4. 

Answer: B,E 

Explanation: 

Incorrect Answer: A, C, D Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06trunk.html