300-075 Guide

300-075 Exam Royal Pack (In Stock.)

 
  • Cisco
  • Exam Number/Code 300-075
  • Product Name CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
  • Questions and Answers
  • 383 Q&As
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  • July 24,2017
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Download of cisco 300 075 exam price materials and training tools for Cisco certification for client, Real Success Guaranteed with Updated 300 075 dump pdf dumps vce Materials. 100% PASS Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) exam Today!

Q51. 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:


Q52. In a Centralized Call processing architecture, you have deployed Extension Mobility (EM) feature. After the deployment of EM, when one of the end-users tries to login to the IP phone, the Error 25 is displayed on the screen. What three things should you do to resolve this issue? (Choose three.) 

A. upgrade the firmware of the IP Phone to the latest version 

B. activate EM feature service under Cisco Unified Serviceability 

C. associate EM Device profile with the end-user 

D. subscribe the MAC address of the IP Phone to EM Service 

E. update EM Phone Service URL to point to the publisher 

F. subscribe device profile to EM phone service in case the enterprise subscription of EM Service is disabled 

Answer: B,C,D 


Q53. 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:


Q54. Which ability does the Survivable Remote Site Telephony feature provide? 

A. a means to allow the local site to continue to send and receive calls in the event of a WAN failure 

B. a means to route calls on-net through other sites during high utilization periods 

C. a method that allows for backup calls in the event that your gateway fails 

D. the ability to force a call out of a certain trunk when the Cisco Unified Communications Manager is being upgraded 

Answer:


Q55. 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:

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 


Q56. Which device must be accessible from the public Internet in a Collaboration Edge environment? 

A. Expressway-C 

B. Cisco Unified Communications Manager 

C. Cisco IM and P 

D. Expressway-E 

E. VCS Control 

Answer:


Q57. What is the purpose of a SAF Client? 

A. To decode address information and route calls to and from the end points 

B. To pass IP information from the CUCM to the endpoint 

C. To learn about and advertise or subscribe information about SAF network services 

D. To reside in the Cisco IOS software, and to communicate with the SAF forwarder 

Answer:


Q58. The administrator at Company X is trying to set up Extension Mobility and has done these steps: 

-Set up end users accounts for the users who need to roam 

-Set up a device profile for the type of phones users will be allowed to log in Users have reported to the administrator that they are unable to log in to the phones 

designated for Extension Mobility. Which two options are the two reasons for this issue? (Choose two.) 

A. The user device profile is not associated to the correct end user. 

B. The username must be numeric only and must match the DN. 

C. The Extension Mobility service has not been enabled under the Cisco Unified Serviceability Page. 

D. Extension Mobility has not been enabled under Enterprise Parameters. 

E. The user must ensure that their main endpoint is online and registered, otherwise they cannot log in elsewhere. 

Answer: A,C 


Q59. 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 


Q60. Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two.) 

A. Configure all SIP trunks with DNS SRV 

B. Configure all SIP trunks with Cisco Unified Border Element 

C. Configure all SIP trunks to point to a SIP gateway 

D. Configure SIP trunks to be members of route groups and route lists 

E. Configure all SIP trunks to allow TCP ports 5060 

F. Configure all SIP trunks to point to a gatekeeper through SIP to H.323 gateway 

Answer: A,D 

Explanation: 

Incorrect Answer: B, C, E, F For SIP trunks, Cisco Unified Communications Manager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS host name. The order of the IP addresses depends on the DNS response and may be identical in each DNS query. The OPTIONS request may go to a different set of remote destinations each time if a DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses, or if a host name (configured on the SIP trunk) resolves to more than 10 IP addresses. Thus, the status of a SIP trunk may change because of a change in the way a DNS query gets resolved, not because of any change in the status of any of the remote destinations. 

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html 


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