10 tips on How to 300-075 Test Like a Badass [131 to 140]


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300-075 Product Description:
Exam Number/Code: 300-075 vce
Exam name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
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Certification: Cisco Certification
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Exam Code: 300-075 (Practice Exam Latest Test Questions VCE PDF)
Exam Name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Certification Provider: Cisco
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2016 May 300-075 Study Guide Questions:

Q131. Refer to the exhibit. 


Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable? 

A. the phone device CSS 

B. the phone line CSS 

C. the phone line/device combined CSS 

D. the SAF CSS configured on the CCD requesting service 

E. the phone AAR CSS configured at the phone device 

F. No special CSS is required. If SAF patterns are accessible, the PSTN reroute is automatic. 

Answer: E 


Q132. Which statement about technology implementation strategy is true? 

A. Cisco Unified Communications Manager Express can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. 

B. Cisco Unified Communications Manager Express in SRST mode can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. 

C. SRST can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. 

D. SRST and MGCP fallback can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. 

Answer: A 


Q133. When an incoming PSTN call arrives at an MGCP gateway, how does the calling number get normalized to a global E.164 number with the + 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 MGCP gateway. 

Answer: D 

Explanation: 

Incorrect Answer: A, B, C, E Configuring calling party normalization alleviates issues with toll bypass where the call is routed to multiple locations over the IP WAN. In addition, it allows Cisco Unified Communications Manager to distinguish the origin of the call to globalize or localize the calling party number for the phone user. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallpn.html 


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Q134. Which statement about SIP precondition is most correct? 

A. When configuring SIP precondition, the SIP trunk must have access to an RSVP agent. 

B. When configuring SIP precondition, the IP phones must have access to an RSVP agent. 

C. When configuring SIP precondition, the IP phones and SIP trunk must have access to an RSVP agent. 

D. RSVP agents are only required for the IP phones. SIP trunks require RSVP agents only when fall back to local RSVP is configured. 

E. SIP trunk will always require RSVP agents regardless of what RSVP type is configured. 

Answer: D 


Q135. Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two.) 

A. H.323 

B. H.245 

C. H.225 

D. intercluster 

E. intracluster 

Answer: C,D 


Q136. A Cisco Unified Communications Manager cluster is installed in headquarters only. 

How can international calls be blocked while national calls are allowed for branch office Cisco IP Phones during a WAN failure? 

A. Configure CSS and partitions in Cisco Unified Communications Manager and apply the CSS and partitions to the SRST ISR. 

B. Configure CSS and partitions in the SRST ISR. 

C. Configure COR in the SRST ISR. 

D. Configure voice translations in the SRST ISR. 

Answer: C 


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Q137. What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage? 

A. B2BUA 

B. SIP server 

C. SIP proxy 

D. SIP SRST router 

E. SIP registrar 

Answer: A 


Q138. You are deploying a Cisco Unified Communications Manager solution with MGCP gateways at multiple locations. Which firewall and ACL configuration must you perform to allow the MCGP gateways to function correctly? 

A. Allow access to TCP port 2428. 

B. Block TCP port 1720. 

C. Open access to all TCP and UDP ports. 

D. Allow access to TCP port 1720. 

E. Block access to TCP ports 2427 and 2428. 

Answer: A 


Q139. Which two statements describe RSVP-enabled locations-based CAC? (Choose two.) 

A. RSVP can be enabled selectively between pairs of locations. 

B. Using RSVP for CAC simply allows admitting or denying calls based on a logical 

configuration that is ignoring the physical topology. 

C. RSVP is topology aware, but only works with full mesh networks. 

D. An RSVP agent is a Media Termination Point that the call has to flow through. 

E. RSVP and RTP are used between the two endpoints. 

Answer: A,D 

Explanation: 

Incorrect Answer: B, C The RSVP policy that is configured for a location pair overrides the default interlocation RSVP policy that configure in the Service Parameter Configuration window. RSVP supports audio, video, and data pass-through. Video data pass-through allows video and data packets to flow through RSVP agent and media termination point devices Link: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02rsvp.ht ml#wp1070214 


Q140. Which configuration command disables the secondary dial tone on the branch office ISR for users calling from the PSTN into the branch office during a WAN failure? 

A. direct-inward-dial 

B. voice translation-rule 

C. incoming called-number 

D. application 

Answer: A 



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