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2021 Apr 300-075 exam question

Q1. Which option best describes a service that assembles a network model from configured locations and link data in one or more clusters? 

A. LBM 

B. Weight 

C. LBM Hub 

D. Shadow 

Answer: A 


Q2. Refer to the exhibit. 

The H.323 Gateway is showing status “unknown”. Which statement is true? 

A. The gateway must be reset in Cisco Unified Communications Manager. 

B. The no gateway command followed by the gateway command must be issued in Cisco IOS. 

C. The mgcp commands must be removed. 

D. H.323 gateways do not register with Cisco Unified Communications Manager H.323 gateways always show status "Unknown". 

E. VUAN 1 20 may be down and so the H.323 gateway appears offline to the Cisco Unified Communications Manager 

Answer: D 

Explanation: 

Incorrect Answer: A, B, C, E 

After a gateway is registered with Cisco Unified Communications Manager, gateway registration status may display in Cisco Unified Communications Manager Administration as unknown. 

Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06gtway. html 


Q3. 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 


Q4. What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager? 

A. EF/46 

B. CS6/48 

C. AF41/34 

D. CS3/24 

E. CS4/32 

Answer: E 


Q5. Which three configuration settings are included in a default region configuration? (Choose three.) 

A. Audio Codec 

B. Video Call Bandwidth 

C. Link Loss Type 

D. Location Description 

E. Immersive Bandwidth 

F. Real Time Protocol 

Answer: A,B,C 


Replace 300-075 test engine:

Q6. 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 


Q7. Refer to the following exhibit. 

The MGCP gateway has the following configurations: 

called party transformation CSS HQ_cld_pty CSS (partition=HQ cld_pty.Pt) call.ng party transformation CSS HQ_clng_pty CSS (partition=HQ_clng_pty Pt) 

All translation patterns have the check box "Use Calling Party's External Phone Number Mask" enabled. 

When the IP phone at extension 3001 places a call to 9011 49403021 56001# what is the resulting called and calling number that is sent to the PSTN? 

A. The called number is 01 1 49403021 56001. The calling number will be 5553001 and number type set to subscriber. 

B. The called number is 011 49403021 56001. The calling number will be 5215553001 and number type set to national. 

C. The called number is 4940302156001 with number type set to international. The calling number will be 5215553001 and number type set to national. 

D. The called number is +49403021 56001 with number type set to international. The calling number will be 5215553001 and number type set to subscriber. 

Answer: A 

Explanation: 

Incorrect Answer: B, C, D Check the check box "Use Calling Party's External Phone Number Mask" if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls. You may also configure an External Phone Number Mask on all phone devices. Link: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00805 b6f33.shtml 


Q8. 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. 

How many route lists and route groups should be configured for AAR at a minimum? 

A. a single route list with a local route group for each site 

B. two route lists and two route groups for each site 

C. a single route list and four route groups for each site 

D. None. The AAR CSS can point directly to the route pattern. 

Answer: A 


Q9. What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage? 

A. B2BUA 

B. SIP server 

C. SIP proxy 

D. SIP SRST router 

E. SIP registrar 

Answer: A 


Q10. What is the difference between an MGCP gateway and a SIP gateway? 

A. An MGCP gateway that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers. 

B. An MGCP gateway can be added in Cisco Unified Communications Manager under the Gateway Type field using the gateway model. 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 MGCP gateway does not require a call agent for PSTN calls to be placed and received. 

D. An MGCP gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown". 

E. The SIP gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The MGCP gateway must be configured in Cisco Unified 

Communications Manager using the domain name. 

Answer: B