300-075 Exam Questions - Online Test


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New Cisco 300-075 Exam Dumps Collection (Question 2 - Question 11)

Q2. You need to verify if the Media Control Gateway Protocol gateway is enabled and active. Which command should you use for this purpose?

A. show running-config

B. show fallback-mgcp

C. show gateway

D. show ccm-manager fallback-mgcp

E. show running-config gateway

F. show fallback-mgcp ccm-manager

Answer: D


Q3. Which command should you use on the gatekeeper to specify the address of a Cisco Unified Communications Manager IP address?

A. gw-type-ip

B. gw-type-prefix-ip

C. gw-ip-cm

D. gw-type-prefix

E. gw-ip-type

F. gw-prefix-ip

Answer: D


Q4. Refer to the exhibit.

When a call between two HQ users is being conferenced with a remote user at BR, which configuration is needed?

A. The BR_MRG must contain the transcoder device. The BR_MRGL must be assigned to the BR phones.

B. The HQ_MRG must contain the transcoder device. The HQ_MRGL must be assigned to the HQ phones.

C. A transcoder should be configured at the remote site and assigned to all remote phones through the BR_MRGL.

D. The HQ_MRG must contain the transcoder device. The HQ_MRGL must be assigned to the software conference bridge.

E. Enable the software conference bridge to support the G.711 and G.729 codecs in Cisco Unified Communications Manager Service Parameters.

Answer: D


Q5. Which two configurations are needed to implement SRST in Cisco Unified Communications Manager? (Choose two.)

A. SRST Gateway setting in Cisco Unified Communications Manager

B. SRST Reference configured in Cisco Unified Communications Manager

C. Device Pool SRST Reference setting

D. Call Manager Group setting

E. Cisco Unified Communications Locations setting

Answer: B,C


Q6. Which two features require or may require configuring a SIP trunk? (Choose two.)

A. SIP gateway

B. Call Control Discovery between a Cisco Unified Communications Manager and Cisco Unified Communications Manager Express

C. Cisco Device Mobility

D. Cisco Unified Mobility

E. registering a SIP phone

Answer: A,B

Explanation: Incorrect: CDE

All protocols require that either a signaling interface (trunk) or a gateway be created to accept and originate calls. Device mobility allows Cisco Unified Communications Manager to determine whether the phone is at its home location or at a roaming location. Cisco Unified Mobility gives users the ability to redirect incoming IP calls from Cisco Unified Communications Manager to different designated phones, such as cellular phones.

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html# wpxref77849


Q7. Refer to the exhibit.

IT shows an H.323 gateway configuration in a Cisco Unified Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on the H.323 gateway shows the correct called-party number as 5015552001. Which two configuration changes can correct this issue? (Choose two.)

A. Add port 1/0:23 to dial-peer voice 123 pots.

B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4.

C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the

voice translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is left at default (All).

D. Add the command h323-gateway voip id on interface vlan120.

E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the Significant Digits for inbound calls to 4.

Answer: B,E

Explanation: Incorrect: ACD

Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called.

Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06trunk.html


Q8. Which bandwidth amounts are correct for configuring locations?

A. 8 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722

B. 8 kb/s for G.729, 64 kb/s for G.711, and 16 kb/s for G.722

C. 64 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722

D. 8 kb/s for G.729, 8 kb/s for G.711, and 8 kb/s for G.722

Answer: A


Q9. What is the purpose of a SAF Client?

A. To decode address information and route calls to and from the end points

B. To pass IP information from the CUCM to the endpoint

C. To learn about and advertise or subscribe information about SAF network services

D. To reside in the Cisco IOS software, and to communicate with the SAF forwarder

Answer: C


Q10. Which statement about AAR functionality is true?

A. AAR reroutes calls through the PSTN upon WAN failure.

B. AAR reroutes calls through the PSTN when Cisco Unified Communications Manager

CAC blocks a call.

C. AAR reroutes calls through the WAN when the PSTN connection fails.

D. AAR reroutes calls to another phone number when a Cisco IP Phone unregisters.

Answer: B


Q11. Refer to the exhibit. When a user presses a speed dial to +442079460255 when the SAF network is down, which event should occur?

A. The call will reroute via the PSTN with the constructed PSTN number as 442079460255.

B. The call will reroute via the PSTN with the constructed PSTN number as

+442079460255.

C. The call will reroute via the PSTN with the constructed PSTN number as 00442079460255.

D. The call will fail because the ToDID is 0:.

E. The call will fail because the called number will be 2079460255.

Answer: B


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