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Product Description:
Exam Number/Code: 300-075
Exam name: CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
n questions with full explanations
Certification: Cisco Certification
Last updated on Global synchronizing

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New Cisco 300-075 Exam Dumps Collection (Question 6 - Question 15)

Question No: 6

Which option describes the reason that transcoding resources are added in Cisco Unified Communications Manager?

A. to enable transcoding resources in a Cisco Unified Communications Manager server

B. to enable Cisco Unified Communications Manager to select the optimal single codec for end-to-end calls

C. to enable transcoding resources in Cisco IP Phones

D. to provide transcoding resources in Cisco IOS gateways to Cisco Unified Communications Manager

Answer: D


Question No: 7

Which action configures CAC utilizing only Cisco Unified Communications Manager software?

A. Configure Cisco Unified Communications Manager regions.

B. Configure Cisco Unified Communications Manager locations.

C. Configure Cisco Unified Communications Manager RSVP-enabled locations.

D. Configure Cisco Unified Communications Manager MTPs.

Answer: B


Question No: 8

When you configure regions in a Cisco Unified Communications Manager multisite cluster, which two are ways to prevent G.722 from being advertised in the cluster? (Choose two.)

A. modify the service parameter

B. modify the enterprise parameter

C. modify the device pool

D. modify the line settings

Answer: A,B


Question No: 9

Which command displays the detailed configuration of all the Cisco Unified IP phones, voice ports, and dial peers of the Cisco Unified SRST router?

A. show call-manager-fallback all

B. show dial-peer voice summary

C. show ephone summary

D. show voice port summary

Answer: A


Question No: 10

Refer to the exhibit. Assume that the HQ phones have access to the HQ partition, and BR phones have access to the BR partition. Which set of implementations would best address the overlapping directory number extensions for intersite (WAN) calling between the HQ site and the BR site?

A. Configure a route pattern 8222.[12]XXX for site HQ, and assign it to partition HQ. Configure the called party DDI of Predot.

Configure a route pattern for site BR 8111.[1-3]XXX, and assign it to partition BR. Configure called party DDI Predot.

Use the local gateway at each site. Prefix the appropriate site code for the calling number.

B. Configure a single route pattern for both sites 8[12,12,12].[1-32]XXX. Use a route list that contains the local route group for each site. Prefix the appropriate site code for the calling number.

C. Configure a translation pattern 8222.[12]XXX for site HQ, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.

Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.

For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.

D. Configure a translation pattern 8222.[12]XXX for site HQ, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.

Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.

For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.

Answer: C


Question No: 11

Refer to the exhibit.

A PSTN call arrived at the MGCP gateway. The calling number was received as 14087071222 with number set to type international. The HQ_clng pty_CSS contains the HQ_clng_pty Pt partition. Which caller ID is displayed on the IP phone?

A. +087071222

B. 14087071222

C. 087071222

D. 4087071222

E. 14087071222

Answer: C

Explanation: Incorrect: ABDE

SIP trunks and MGCP gateways can support sending the international escape character, +, for calls. H.323 gateways do not support the +. QSIG trunks do not attempt to send the +. For outgoing calls through a gateway that supports +, Cisco Unified Communications Manager can send the + with the dialed digits to the gateway. For outgoing calls through a gateway that does not support +, the gateway strips the + when Cisco Unified Communications Manager sends the call information to the gateway.

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html


Question No: 12

Which two statements about the functionality of a gatekeeper are true? (Choose two.)

A. Cisco Unified Communications Manager has gatekeeper functionality built in.

B. Cisco Unified Communications Manager registers with a gatekeeper via SIP.

C. Cisco Unified Communications Manager registers with a gatekeeper via H.323.

D. A gatekeeper can enable CAC and AAR.

E. A gatekeeper can enable CAC, but not AAR.

Answer: C,E


Question No: 13

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: 14

When an incoming PSTN call arrives at an H.323 gateway, how does the calling number get normalized to a global E.164 number with + prefix in Cisco Unified Communications Manager?

A. Normalization is done using translation patterns.

B. Normalization is done using route patterns.

C. Normalization is done using the gateway incoming called party prefixes based on number type.

D. Normalization is done using the gateway incoming calling party prefixes based on number type.

E. Normalization is achieved by local route group that is assigned to the H.323 gateway.

Answer: D


Question No: 15

While operating in SRST, what is needed to route calls outside of the remote site location to the PSTN?

A. SIP trunk

B. CallManager route patterns

C. translation patterns

D. POTS dial peers

E. VOIP dial peers

Answer: D

Explanation: Incorrect: ABCE

in time of srst configuration on router, please configure a dial-peer so that call flow in SRST mode.

Link:


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