Exam Code: 300-075 (Practice Exam Latest Test Questions VCE PDF)
Exam Name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Certification Provider: Cisco
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2021 Apr 300-075 free exam questions
Q81. 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.
Q82. What happens if location-based CAC is used and there is no bandwidth available when a remote caller is placed on hold?
A. Cisco Unified Communications Manager sends TOH rather than MOH.
B. Cisco Unified Communications Manager terminates the call.
C. Cisco Unified Communications Manager plays default MOH.
D. Cisco Unified Communications Manager attempts to reconnect the call immediately.
Q83. Which two options for a Device Mobility-enabled IP phone are true? (Choose two.)
A. The phone configuration is not modified.
B. The roaming-sensitive parameters of the current (that is, the roaming) device pool are applied.
C. The user-specific settings determine which location-specific settings are downloaded from the Cisco Unified Communications Manager device pool.
D. If the DMGs are the same, the Device Mobility-related settings are also applied.
Q84. 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.
Q85. 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?
B. SIP server
C. SIP proxy
D. SIP SRST router
E. SIP registrar
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Q86. When considering Extension Mobility, what happens if a user logs into a phone for which the user does not have a user device profile?
A. The phone reboots with an error.
B. If a default device profile for this phone has been configured, it is loaded.
C. The user cannot log in.
D. Another user device profile is loaded.
Q87. Which three devices support the SAF Call Control Discovery protocol? (Choose three.)
A. Cisco Unified Border Element
B. Cisco Unity Connection
C. Cisco IOS Gatekeeper
D. Cisco Catalyst Switch
E. Cisco IOS Gateway
F. Cisco Unified Communications Manager
Q88. Refer to the exhibit
When the Cisco Unified Communications Manager advertises the Hosted DN Pattern, which pattern would be advertised?
A. 2XXX and the T0D1D will be 0:+498950555
B. 2XXX and the ToDID will be 0:+4989531 21
C. 4989S05552XXX and the ToDiD will be 0:
D. + 4989631 21 2XXX and the ToDiD will be 0:
E. Both +4989505552XXXand +4989531 21 2XXX will be advertised with ToDID of 0:
Incorrect Answer: B, C, D, E PSTN failover prepend digit is +498950555 Link:
Q89. You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2-Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?
Q90. For which VoIP protocol does a gatekeeper provide address translation and control access?