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2021 Aug icc 300-075 exam:
Q121. 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
Q122. Which three options are supplementary services that are affected by MTP? (Choose three.)
A. Call Hold
B. Call Transfer
C. Call Park
D. Call Pickup
E. Speed Dial
F. Call Back
Answer: A,B,C
Q123. Which two options are valid service parameter settings that are used to set up proper video QoS behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)
A. DSCP for Video Calls when RSVP Fails
B. Default Intraregion Min Video Call Bit Rate (Includes Audio)
C. Default Interregion Max Video Call Bit Rate (Includes Audio)
D. DSCP for Video Signaling
E. DSCP for Video Signaling when RSVP Fails
Answer: A,C
Q124. Which technologies provide remote-site redundancy for Cisco IP Phones during a WAN failure?
A. SRST and MGCP fallback
B. SRST and TEHO
C. TEHO and MGCP fallback
D. SRST and AAR
Answer: A
Q125. Which statement about SIP precondition is most correct?
A. When configuring SIP precondition, the SIP trunk must have access to an RSVP agent.
B. When configuring SIP precondition, the IP phones must have access to an RSVP agent.
C. When configuring SIP precondition, the IP phones and SIP trunk must have access to an RSVP agent.
D. RSVP agents are only required for the IP phones. SIP trunks require RSVP agents only when fall back to local RSVP is configured.
E. SIP trunk will always require RSVP agents regardless of what RSVP type is configured.
Answer: D
Up to date icc 300-075 exam:
Q126. When an incoming PSTN call arrives at an H.323 gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager?
A. Normalization is done by configuring the significant digits for inbound calls on the H.323 gateway configuration in Cisco Unified Communications Manager.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming calling party prefixes based on number type.
D. Normalization is achieved by local route group that is assigned to the H.323 gateway.
Answer: A
Q127. Refer to the exhibit.
When the user of a phone registered to the Cisco Unified Communications Manager places a call to 3001 when the SAF network is down, what happens?
A. The call fails.
B. The call is rerouted to the PSTN with the constructed PSTN number as +442288223001
C. The call is rerouted to the PSTN with the constructed PSTN number as 442288223001
D. The call is rerouted to the PSTN with the constructed PSTN number as 0002288223001
E. The call is rerouted to the PSTN with the constructed PSTN number as +0002288223001
Answer: A
Explanation:
Incorrect Answer: B, C, D When the SAF forwarder loses network connection with its call-control entity, the SAF forwarder withdraws those learned patterns that were published by the call control entity. In this case, CCD requesting service marks those learned patterns as unreachable via IP, and the calls get routed through the PSTN gateway. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html
Q128. Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with video endpoints registered on both systems. Users find that video endpoints registered on Call manager can call each other and likewise for the endpoints registered on the VCS server. The administrator for Company X realizes he needs a SIP trunk between the two systems for any video endpoint to call any other video endpoint. Which two steps must the administrator take to add the SIP trunk? (Choose two.)
A. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server.
B. Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
C. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone.
D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.
E. Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate.
Answer: A,C
Q129. Which two configurable options are available to enable Early Offer for calls over a Cisco Unified Communications Manager SIP trunk? (Choose two.)
A. No Media Termination Point Required
B. Media Termination Point Required
C. Accept Audio Codec Preferences in Received Offer
D. Early Offer support for voice and video calls Mandatory (insert MTP if needed)
E. Use Trusted Relay Point
Answer: B,D
Q130. If the device pool in the phone record does not match the device pools in the matching subnet, what will the system consider the phone to be?
A. roaming
B. unregistered
C. unknown
D. new device
Answer: A