Cisco 300-075 ExamCIPTV2 Implementing Cisco IP Telephony and Video, Part 2

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2016 May 300-075 Study Guide Questions:

Q166. 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 Answer: A, C, D, E 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 


Q167. Which method can be used to address variable-length dial plans? 

A. Overlap sending and receiving. 

B. Add a prefix for all calls that are longer than 10-digits long 

C. Use nested translation patterns to eliminate inter-digit timeout 

D. Use the @macro on the route pattern 

E. Use MGCP gateways, which support variable-length dial plans 

Answer: A 

Explanation: 

Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsintrcm.htm 


Q168. 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. What should the TEHO-US route list configuration consist of? 

A. First route group should point only to the U.K. gateway. The second route group should point to the U.S. gateway. 

B. First route group should be only the local route group. The second route group should point to the U.S. gateway. 

C. First route group should point only to the U.S. gateway. The second route group should be the local route group. 

D. The TEHO-US route list should contain only the local route group. The globalized configuration means that the appropriate gateway will be selected automatically. 

E. The \+! route pattern should point directly to the U.S. gateway. 

Answer: C 

Explanation: 

Incorrect Answer: A, B, D The route group points to one or more gateways and can choose the gateways for call routing based on preference. The route group can serve as a trunk group by directing all calls to the primary device and then using the secondary devices when the primary is unavailable. One or more route lists can point to the same route group. Link: 

http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08 gw.html#wp1167274 


Q169. 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 


Q170. Which two locations are the best locations that an end user can use to determine if an IP phone is working in SRST mode? (Choose two.) 

A. Cisco Unified Communications Manager Administration 

B. IP phone display 

C. Cisco Unified SRST Router 

D. Cisco Unified MGCP Fallback Router 

E. physical IP phone settings 

Answer: B,E 

Explanation: 

Incorrect Answer: A, C, D IP Phone display and Physical phone IP settings are two locations were an end user can determine if an IP phone is working in SRST mode. Link: http://my.safaribooksonline.com/book/telephony/1587050757/survivable-remote-site-telephony-srst/529 


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Q171. 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: E 


Q172. 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 


Q173. When a SIP trunk is added for Call Control Discovery, which statement is true? 

A. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The Enable SAF check box should be selected. 

B. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The Trunk Service Type should be Call Control Discovery. 

C. The SIP trunk is added by selecting Call Control Discovery Trunk and then selecting SIP as the protocol to be used. 

D. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF. 

Answer: B 


Q174. Which two configurations can you perform to allow Cisco Unified Communications Manager SIP trunks to send an offer in the INVITE? (Choose two.) 

A. Enable the Media Termination Point Required option on the SIP trunk. 

B. Enable the Early Offer Support for Voice and Video Calls option on the SIP profile. 

C. Select the Display IE Delivery check box in the gateway configuration. 

D. Select the Enable Inbound FastStart check box on the Cisco Unified Communications Manager servers. 

E. Select the SRTP Allowed check box on the SIP trunk. 

F. Execute the isdn switch-type primary-ni command globally. 

Answer: A,B 


Q175. 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: D 


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Free 300-075 questions:

Q176. Which option configures the secondary dial tone option for SRST mode to let the users hear the dial tone for PSTN calls? 

A. voice service voipsecondary dialtone 0 

B. call-manager-fallbacksecondary dialtone 0 

C. dial-peer voice 1 potssecondary dialtone 0 

D. ccm-manager secondary dialtone 0 

Answer: B 


Q177. Refer to the exhibit. 


The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number. Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. 

What should the AAR group prefix be? 

A. 9 

B. 91 

C. none 

D. + 

E. +1 

Answer: C 


Q178. The following exhibit shows configs for H.323 gateway. You have been asked to implement TEHO from a remote branch office with area code 301 to the HQ office with area code 201 using Cisco Unified Communications Manager. The remote office has an MGCP gateway and the HQ office has an H.323 gateway. Once the call arrives at the HQ, it should break out to the PSTN as a seven-digit local call. Which statement about the route pattern is true? 


A. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot and Prefix 9 

B. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot 

C. route pattern should be 91201.[2-9]XXXXXX 

D. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot 

E. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot and Prefix 9 

Answer: A 

Explanation: 

Incorrect Answer: B, C, D, E 

Destination pattern is 91, HQ office area code is 201 . 


Q179. 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 


Which device configuration option will allow an administrator to assign a device to specific rights for making calls to specific DNs? 

A. Media Resource Group List 

B. Device Pool 

C. Location 

D. AAR Group 

E. Calling Search Space 

Answer: E 

Explanation: 

A partition comprises a logical grouping of directory numbers (DNs) and route patterns with similar reachability characteristics. Devices that are typically placed in partitions include DNs and route patterns. These entities associate with DNs that users dial. For simplicity, partition names usually reflect their characteristics, such as "NYLongDistancePT," "NY911PT," and so on. A calling search space comprises an ordered list of partitions that users can look at before users are allowed to place a call. Calling search spaces determine the partitions that calling devices, including IP phones, softphones, and gateways, can search when attempting to complete a call. 


Q180. Which task must you perform before deleting a transcoder? 

A. Delete the dependency records. 

B. Unassign it from a media resource group. 

C. Use the Reset option. 

D. Remove the device pool. 

E. Remove the subunit. 

F. Delete the common device configuration. 

Answer: B 


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