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Q91. This is the configuration on the voice gateway: 

telephony-service 

max-ephones 30 

max-dn 60 preference 0 

srst mode auto-provision all 

srst dn line-mode dual 

srst dn template 3 srst ephone description 

srst fallback auto-provision phone 

srst ephone template 5 

Which ephone-dn would be expected upon activation of SRST? 

A. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones 

B. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones 

C. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones 

D. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones 

Answer:


Q92. Which sign is prefixed to the number in global call routing? 

A. -

B. + 

C. # 

D. @ 

E. & 

F. * 

Answer:


Q93. 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:


Q94. Refer to the exhibit. 

To permit three G.729 calls, what should the bandwidth value be for the ip rsvp bandwidth command? 

A. 32 

B. 48 

C. 64 

D. 88 

E. 128 

Answer:


Q95. If your IP telephony administrator asks you to configure a local zone for your dial plan to control the volume of calls between two end points in a centralized multisite environment, which two types of Call Admission Control can be implemented? (Choose two.) 

A. locations based 

B. automated alternate routing 

C. gatekeeper based 

D. SRST 

E. Cisco Unified Communications Manager based 

Answer: A,B 

Explanation: 

Incorrect Answer: C, D, E Location-based call admission control (CAC) manages WAN link bandwidth in Cisco Unified Communications Manager. Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wp1067747 


Q96. Which two options are effective mechanisms to restrict the maximum number of voice calls on a WAN link? (Choose two.) 

A. Configure a gatekeeper with an SIP trunk. 

B. Configure a gatekeeper and a gatekeeper-controlled trunk in Cisco Unified Communications Manager with bandwidth control. 

C. Configure Cisco Unified Communications Manager regions. 

D. Configure Cisco Unified Communications Manager locations. 

Answer: B,D 


Q97. For which VoIP protocol does a gatekeeper provide address translation and control access? 

A. H.323 

B. SIP 

C. Skinny 

D. H.248 

Answer:


Q98. Scenario There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 7965 and 9971 Video IP Phones. The Cisco VCS and TMS control the Cisco TelePresence Conductor, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

DNS Servers 

Device Pool 

Expressway 

ILS 

Locations 

MRA 

Speed Dial 

SIP Trunk 

The intercluster URI call routing no longer allows calls between sites. What is the reason why this would happen? (Choose two) 

A. Wrong SIP domain configured. 

B. User is not associated with the device. 

C. IP or DNS name resolution issue. 

D. No SIP route patterns for cisco.lab exist. 

Answer: C,D 


Q99. What user profile is used to define the settings for a user on login? 

A. Device Profile 

B. Group Profile 

C. Pool Profile 

D. Specific Profile 

Answer:


Q100. What is a prerequisite of AAR deployment? 

A. You must have a single distributed call processing deployment. 

B. Calls must be manually rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. 

C. Calls must be automatically rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. 

D. Clustering must be implemented over the WAN. 

E. You must have a centralized call processing deployment. 

Answer:


Q101. Which statement is true when device mobility mode is enabled or disabled in the Phone 

Configuration window? 

A. The device mobility mode phone settings take precedence over the service parameter settings. 

B. The service parameter settings take precedence over the device mobility mode phone settings. 

C. The combined service parameter settings and the device mobility mode phone settings will be used. 

D. The default settings will be used due to the conflicts. 

Answer:


Q102. When Cisco Extension Mobility is implemented, which CSS is used for calling privileges? 

A. The user device profile line CSS combined with the device CSS of the physical phone used to log in the extension mobility user. 

B. The user device profile device CSS combined with the line CSS of the physical phone used to log in the extension mobility user. 

C. Only the user device profile device CSS is used. 

D. The combined line/device CSS of the physical phone is used to log in the extension mobility user. 

E. The combined line/device CSS of the user device profile. 

Answer:


Q103. Which two options are valid service parameter settings that are used to set up proper video QoS behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.) 

A. DSCP for Video Calls when RSVP Fails 

B. Default Intraregion Min Video Call Bit Rate (Includes Audio) 

C. Default Interregion Max Video Call Bit Rate (Includes Audio) 

D. DSCP for Video Signaling 

E. DSCP for Video Signaling when RSVP Fails 

Answer: A,C 


Q104. What is the standard Layer 3 DSCP media packet value that should be set for Cisco TelePresence endpoints? 

A. CS3 (24) 

B. EF (46) 

C. AF41 (34) 

D. CS4 (32) 

Answer:


Q105. 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: