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Q71. With Cisco Extension Mobility, when a user logs in to a phone type which has no user device profile, what will happen to the phone?
A. The phone takes on the default clusterwide device profile.
B. The phone creates a new device profile automatically.
C. The phone immediately logs the user off.
D. The phone crashes and reboots.
Answer: A
Q72. Where do you specify the device mobility group and physical location after they have been configured?
A. phones
B. DMI
C. device mobility CSS
D. device pool
E. MRGL
F. locale
Answer: D
Explanation:
Incorrect Answer: A, B, C, E Before you configure a device pool, you must configure the following items if you want to choose them for the device pool, Cisco Unified Communications Manager group (required), Date/time group (required). Region (required) , SRST reference (optional). Media resource group list (optional), Calling search space for auto-registration (optional). Reverted call focus priority (optional), Device mobility group (optional), Device mobility calling search space, Physical location (optional). Location, AAR group. AAR calling search space. Link:
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b02devpl.ht ml
Q73. Refer to the exhibit.
The Cisco Unified Communications Manager at HQ has been configured for end-to-end RSVP. The Cisco Unified Communications Manager at BR has been configured for local RSVP.
RSVP between the locations assigned to the IP phones and SIP trunks at each site are configured with mandatory RSVP. When a call is placed from the IP phone at the BR site to the IP phone at the HQ site, which statement is true?
A. The Cisco Unified Communications Manager at BR will fall back to local RSVP and place the call. No RSVP end-to-end will occur.
B. RSVP end-to-end will occur.
C. The Cisco Unified Communications Manager at BR will use local RSVP. The HQ Cisco Unified Communications Manager will use end-to-end RSVP.
D. The call will fail.
E. The call will proceed as a normal call with no RSVP reservation.
Answer: A
Q74. Which CAC configuration on a gatekeeper restricts to 10 G.711 audio calls?
A. Use the command bandwidth 10.
B. Use the command bandwidth 1280.
C. Use the command bandwidth 160.
D. Use the command bandwidth session 10.
Answer: B
Q75. 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: C
Explanation:
Expiation:
To ensure that the hardware bridge is utilized first with all its resources BEFORE the software bridge is used … you need to have two separate MRG’s and list the hardware MRG 1st in the MRGL …
Q76. Refer to the following exhibit.
The MGCP gateway has the following configurations:
called party transformation CSS HQ_cld_pty CSS (partition=HQ cld_pty.Pt) call.ng party transformation CSS HQ_clng_pty CSS (partition=HQ_clng_pty Pt)
All translation patterns have the check box "Use Calling Party's External Phone Number Mask" enabled.
When the IP phone at extension 3001 places a call to 9011 49403021 56001# what is the resulting called and calling number that is sent to the PSTN?
A. The called number is 01 1 49403021 56001. The calling number will be 5553001 and number type set to subscriber.
B. The called number is 011 49403021 56001. The calling number will be 5215553001 and number type set to national.
C. The called number is 4940302156001 with number type set to international. The calling number will be 5215553001 and number type set to national.
D. The called number is +49403021 56001 with number type set to international. The calling number will be 5215553001 and number type set to subscriber.
Answer: A
Explanation:
Incorrect Answer: B, C, D Check the check box "Use Calling Party's External Phone Number Mask" if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls. You may also configure an External Phone Number Mask on all phone devices. Link: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00805 b6f33.shtml
Q77. A Cisco 3825 needs to be added in Cisco Unified Communications Manager as an H.323 gateway. What should the gateway type be?
A. H.323 gateway
B. Cisco 3825
C. Cisco 3800 series router. The specific model will be selected after the Cisco 3800 is selected.
D. The gateway can be added either as an H.323 gateway or Cisco 3800 series router.
E. The gateway can be added either as an H.323 gateway or Cisco 3825 series router.
Answer: A
Q78. 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
Q79. The corporate WAN has been extended to two newly acquired sites and it includes gatekeeper support. Each site has a Cisco CallManager and an H.323 gateway that allows connection to the PSTN. Which connection method is best for these two new customers?
A. H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (non-gatekeeper controlled)
C. SIP trunk
D. intercluster trunk (gatekeeper-controlled)
Answer: D
Q80. Refer to the exhibit.
The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number. Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. Which statement is true?
A. The AAR group system must be configured on the device configuration of the phones.
B. The AAR group system must be configured on the line configuration of the phones.
C. The single AAR group system cannot be used. A second AAR group must be configured in order to have source and destination AAR groups.
D. The AAR group system must be configured under the AAR service parameters.
Answer: B
