300 075 pdf (4 to 13)

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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

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New Cisco 300-075 Exam Dumps Collection (Question 4 - Question 13)

Question No: 4

Refer to the exhibit. A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. The user in

RTP sees the message "Not Enough Bandwidth" on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.)

A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings.

B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings.

C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.

D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.

E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.

F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.

Answer: B,F

Explanation: Incorrect: ACDE

Explanation: Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party.

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml



Question No: 5

Cisco Unified Communications Manager is configured with CAC for a maximum of 10 voice calls.

Which action routes the 11th call through the PSTN?

A. Configure an SIP trunk to the ISR.

B. Configure Cisco Unified Communications Manager AAR.

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

D. Configure Cisco Unified Communications Manager locations.

Answer: B



Question No: 6

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 PreDot

Prefix Digits Outgoing Calls: +

B. 9.0014085551234 with the Called Party Transformation: Discard Digits PreDot

Prefix Digits Outgoing Calls: +1

C. 900.14085551234 with the Called Party Transformation: Discard Digits PreDot

Prefix Digits Outgoing Calls: +1

D. 900.14085551234 with the Called Party Transformation: Discard Digits PreDot

Prefix Digits Outgoing Calls: +

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

Answer: D

Explanation: Incorrect: ABC

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



Question No: 7

Which Cisco IOS command is used to verify that a SAF Forwarder that is registered with Cisco Unified Communications Manager has established neighbor relations with an adjacent SAF Forwarder?

A. show eigrp service-family ipv4 neighbors

B. show eigrp address-family ipv4 neighbors

C. show voice saf dndball

D. show saf neighbors

E. show ip saf neighbors

Answer: A

Explanation: Incorrect: BCDE

Router# show eigrp service-family ipv4 4453 neighbors

Link: http://www.cisco.com/en/US/docs/ios/saf/command/reference/saf_s1.html



Question No: 8

Which option describes a function of SIP preconditions?

A. SIP preconditions enable end-to-end RSVP over an SIP trunk.

B. SIP preconditions enable RSVP between Cisco IP Phones.

C. SIP preconditions can be enabled in a gatekeeper.

D. SIP preconditions enable end-to-end RSVP for calls through the PSTN.

Answer: A



Question No: 9

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



Question No: 10

When implementing a dial plan for multisite deployments, what must be present for SRST to work successfully?

A. dial peers that address all sites in the multisite cluster

B. translation patterns that apply to the local PSTN for each gateway

C. incoming and outgoing COR lists

D. configuration of the gateway as an MGCP gateway

Answer: B



Question No: 11

You are the Cisco Unified Communications Manager in Certpaper.com. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost. How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three.)

A. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN.

B. The default service must be enabled globally.

C. The command ccm-manager mgcp-fallback must be configured.

D. COR needs to be configured to disallow outbound calls.

Answer: A,B,C

Explanation: Incorrect: D

Class of restriction: Cisco Unified Communications Manager Business Edition 3000 supports class of service (CoS) with respect to geographic reach as follows:

u2013 Campus

u2013 Local

u2013 National

u2013 International

u2013 Emergency services

u2022 Call waiting

u2022 Default ringtones

u2022 MoH

u2022 Speed dials: Single-button, not BLF Link:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps11370/data_sheet_c78-651909.html



Question No: 12

Which of the following are two functions that ensure that the telephony capabilities stay operational in the remote location Cisco Unified SRST router? (Choose two)

A. Automatically detecting a failure in the network.

B. Initiating a process to provide call-processing backup redundancy.

C. Notifying the administrator of an issue for manual intervention.

D. Proactively repairing issues in the voice network.

Answer: A,B



Question No: 13

Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth? (SourcE. Configuring Cisco Unified Video Advantage)

A. 768 kbps

B. 384 kbps

C. 512 kbps

D. 192 kbps

Answer: B

Explanation: Incorrect answer : ACD

Explanation : A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729

(at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. Link:

http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.html#wp1059726



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