cisco 300 075 (1 to 10)

300-075 Guide

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  • Cisco
  • Exam Number/Code 300-075
  • Product Name CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
  • Questions and Answers
  • 245 Q&As
  • Last Updated
  • Dec 03,2018
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cisco 300 075 (1 to 10)

Download of 300 075 pdf exam price materials and bundle for Cisco certification for IT candidates, Real Success Guaranteed with Updated 300 075 pdf pdf dumps vce Materials. 100% PASS Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) exam Today!

Q1. Refer to the exhibit. 

How many calls can be placed to Cluster B? 

A. three G.729 calls 

B. one G.711 call 

C. one G.711 and three G.729 calls 

D. There is no limit for incoming calls to Cluster B. Outgoing calls are limited to one G.711 and three G.729 calls. 

Answer:


Q2. Which three items must you configure to enable SAF Call Control Discovery? (Choose three.) 

A. the SIP or H.323 trunk 

B. hosted DN groups 

C. hosted DN patterns 

D. route patterns 

E. a calling search space 

F. translation patterns 

Answer: A,B,C 


Q3. Which CAC configuration on a gatekeeper restricts to 10 G.711 audio calls? 

A. Use the command bandwidth 10. 

B. Use the command bandwidth 1280. 

C. Use the command bandwidth 160. 

D. Use the command bandwidth session 10. 

Answer:


Q4. Which option configures call preservation for H.323-based SRST mode? 

A. voice service voip h323 call preserve 

B. call preservation not possible with H.323 

C. call-manager-fallback preserve-call 

D. dial-peer voice 1 voip call preserve 

Answer:


Q5. 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:


Q6. How is a SIP trunk in Cisco Unified Communications Manager configured for SIP precondition? 

A. The configuration is done by selecting a SIP precondition trunk for trunk type. 

B. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk. 

C. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk. 

D. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. The new SIP profile must then be assigned to the SIP trunk. 

Answer:


Q7. 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:


Q8. Refer to the exhibit. 

With the Mobile Connect feature configured, when the PSTN phone calls the Enterprise user at extension 3001, the Enterprise user's mobile phone does not ring. Which CSS is responsible for ensuring that the correct partitions are accessed when calls are sent to the Enterprise user's mobile phone? 

A. the gateway CSS 

B. the Phone Device CSS 

C. the Remote Destination Profile CSS 

D. the Remote Destination Profile Rerouting CSS 

E. the Phone Line (DN)CSS 

Answer:

Explanation: 

Incorrect Answer: A, B, C, E 

Ensure that the gateway that is configured for routing mobile calls is assigned to the partition that belongs to the Rerouting Calling Search Space. Cisco Unified Communications Manager determines how to route calls based on the remote destination number and the Rerouting Calling Search Space. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmfeat/fsmobmgr .html 


Q9. Which statement about the host portion format in Cisco Unified Communications Manager URI dialing is false? 

A. The host portion cannot start or end with a hyphen. 

B. The host portion is not case sensitive. 

C. The host portion accepts characters a-z, A-Z, 0-9, hyphens, and periods. 

D. The host portion can have two periods in a row. 

Answer:


Q10. 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 9971 Video IP Phone. The Cisco VCS is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

SX20 System Information 

DX650 Configuration 

MRGL 

DP 

Locations 

AARG 

CSS 

Movi Failure 

Movi Settings 

What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two.) 

A. Incorrect username and password 

B. Wrong SIP domain configured. 

C. User is not associated with the device. 

D. IP or DNS name resolution issue. 

E. CSF Device is not registered. 

F. IP Phone DN not associated with the user. 

Answer: B,D 


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