[Certified] 300-075 Cisco practice test 131-140 (Mar 2017)

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2017 Mar 300-075 Study Guide Questions:

Q131. 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 

Q132. 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 

Q133. A voice-mail product that supports only the G.711 codec is installed in headquarters. 

Which action allows branch Cisco IP phones to function with voice mail while using only the G.729 codec over the WAN link to headquarters? 

A. Configure Cisco Unified Communications Manager regions. 

B. Configure transcoding within Cisco Unified Communications Manager. 

C. Configure transcoding resources in Cisco IOS and assign to the MRGL of Cisco IP phones. 

D. Configure transcoder resources in the branch Cisco IP phones. 

Answer: C 

300-075  practice test

Updated 300-075 free exam questions:

Q134. Which two options must be selected in the SIP Trunk Security Profile configuration between Cisco Unified Communications Manager and Expressway? (Choose two) 

A. Enable application-level authorization 

B. Accept presence subscription 

C. Accept out-of-dialog refer 

D. Accept unsolicited notification 

E. Accept replaces header 

F. Transmit security status 

G. Allow charging header 

Answer: D,E 

Q135. 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: D 

Q136. 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 

300-075  practice test

Simulation 300-075 guidance:

Q137. Which two statements about SAF service identifier numbers are true? (Choose two.) 

A. They are generated in the format service:sub-service:instance.instance.instance.instance. 

B. They are 16-bit decimal identifiers. 

C. They are generated in the format data-source:sub-service:instance.matrix.fifty.saf. 

D. They are 32-bit decimal identifiers. 

E. They are generated in the format data.saf.cucm-publisher.asf@domain.local. 

F. They are generated in the format telco.cisco.saf-forwader.db.replicate.data.local. 

Answer: A,B 

Q138. 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. To match the US-TEHO pattern \\+!, how should the translation pattern be configured? 

A. 9001.4085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: + 

B. 9.0014085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: +1 

C. 900.14085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: +1 

D. 900.14085551234 with the Called Party Transformation:Discard Digits PreDotPrefix Digits Outgoing Calls: + 

E. 001.4085551234 with the Called Party Transformation:Prefix Digits Outgoing Calls: + 

Answer: D 

Explanation: 

Incorrect Answer: A, B, C The PSTN access code for the UK is 9, International call code is 001, The international escape character, +, signifies the international access code in a complete E.164 number format Link: http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03r p.html 

Q139. 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. Which statement is true? 

A. The AAR group system must be configured on the device configuration of the phones. 

B. The AAR group system must be configured on the line configuration of the phones. 

C. The single AAR group system cannot be used. A second AAR group must be configured in order to have source and destination AAR groups. 

D. The AAR group system must be configured under the AAR service parameters. 

Answer: B 

Q140. 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 

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