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2021 Mar 300-075 vce
Q111. Which statement is correct about AAR?
A. The end users see, "Network Congestion Rerouting?" but AAR is otherwise transparent to the end user and works without user intervention.
B. AAR will display "not enough bandwidth" on the IP phone while it reroutes the call.
C. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.
D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth.
Answer: A
Explanation:
Incorrect Answer: B, C, D Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml
Q112. When an incoming PSTN call arrives at an H.323 gateway, how does the calling number get normalized to a global E.164 number with + 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 H.323 gateway.
Answer: D
Q113. Scenario
There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 7965 and 9971 Video IP Phones. The Cisco VCS and TMS control the Cisco TelePresence Conductor, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
DNS Servers
Device Pool
Expressway
ILS
Locations
MRA
Speed Dial
SIP Trunk
What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM endpoints? (Choose two.)
A. Media Resource Group List.
B. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
C. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS.
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM.
E. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
Answer: B,D
Q114. Which trunk should you use in an H.323 gatekeeper-controlled network?
A. H.323
B. H.225
C. SIP
D. Intercluster
E. MGCP FXO trunk
F. MGCP T1/E1 trunk
Answer: B
Q115. You have been asked to deploy Cisco Extension Mobility Cross Cluster for a distributed call processing environment. During the initial extension mobility login request, how does the visiting cluster determine if the user is a local user or a remote user?
A. by using a third-party automatic provisioning tool to verify user ID
B. by broadcasting a request to all clusters to verify the user type
C. from user IDs that are created by default when the user logs in
D. by using Extension Mobility Cross Cluster Session Initiation Protocol (SIP) trunks
E. by verifying against the local database
F. by verifying the visiting Trivial File Transfer Protocol
Answer: E
Up to the minute 300-075 braindumps:
Q116. 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
Q117. Which method can be used to address variable-length dial plans?
A. Overlap sending and receiving.
B. Add a prefix for all calls that are longer than 10-digits long
C. Use nested translation patterns to eliminate inter-digit timeout
D. Use the @macro on the route pattern
E. Use MGCP gateways, which support variable-length dial plans
Answer: A
Explanation:
Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsintrcm.htm
Q118. What is the fastest way for an engineer to test the implementation of SRST in a production environment?
A. Shut down the Cisco Unified Communications Manager Servers.
B. Shut down the switch ports connected to the Cisco Unified Communications Manager Servers.
C. Add a null route to the publisher Cisco Unified Communications Manager at the remote router. Remove the null route when the operation is verified.
D. Unplug the IP phones from their switch ports.
E. Verification is not needed.
Answer: C
Q119. Scenario
There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 9971 Video IP Phone. The Cisco VCS and TMS control the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
DP
Locations
CSS
SRST
SRST-BR2 Config
BR2 Config
SRSTPSTNCall
After adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Device Pool cannot be default.
B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router.
C. The router does not support SRST.
D. The SRST enabled router is not configured correctly.
Answer: A
Q120. 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