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Exam Code: 300-075
Exam Name: CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
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New Cisco 300-075 Exam Dumps Collection (Question 9 - Question 18)
Question No: 9
Which device is needed to integrate H.320 into the Cisco video solution? (SourcE. Introducing IP Video Telephony)
A. video gateway
B. MGCP gateway
C. H.323 gatekeeper
Explanation: Incorrect: ABD
As with H.323 MCUs, H.320 gateways are provisioned in Cisco Unified CallManager as
H.323 gateways, and then route patterns are configured to extend calls to these devices. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42video.html#wp1046523
Question No: 10
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
Question No: 11
Which two actions are performed by the Call Control Discovery service after the local Cisco Unified Communications Manager loses its TCP connection with the primary and secondary Service Advertisement Framework? (Choose two.)
A. Calls are routed to the PSTN gateway after the Call Control Discovery Learned Pattern IP Reachable Duration parameter expires.
B. All learned patterns are purged from the local cache after the Call Control Discovery PSTN Failover Duration parameter expires.
C. The Service Advertisement Framework forwarder contacts all the remaining Service Advertisement Framework forwarders in the cluster.
D. All the remaining Service Advertisement Framework forwarders are notified for their learned patterns.
E. The Cisco Unified Communications Manager establishes a connection with the primary and secondary Service Advertisement Framework after the Learned Pattern IP Reachable Duration parameter expires.
F. Call Control Discovery immediately redirects all the calls to the PSTN gateway based on the learned patterns.
Question No: 12
When configuring Digit Manipulation, where exactly should you apply Digit Stripping?
A. globally to all calls
B. voice port
C. Plain Old Telephone Systems (POTS) dial peer
D. Voice Over Internet Protocol (VOIP) dial peer
E. trunk group
F. Non-Facility Associated Signaling (NFAS) interface
Question No: 13
Refer to the following exhibits.
Users in the U.S dial Germany by calling 9011 49 followed by the remaining digits. What would be the most suitable configuration for Connection X?
A. Configure a SIP trunk to 10.140.1.1 and a SIP route pattern +49T in Cisco Unified Communications Manager.
B. Configure a SIP trunk to the Cisco Unified Border Element and route pattern +49T in Cisco Unified Communications Manager.
C. configure a SIP trunk to the Cisco Unified Border Element. Configure a translation pattern for 9011.49T using DDI Predot prefix + and CSS to point to a route pattern partition
\+! which uses the SIP trunk.
D. Configure a SIP trunk to the ITSP. Configure a translation pattern for 9011.49T using DDI predot prefix + and CSS to point to a route pattern partition \+! which uses the SIP trunk.
Explanation: Incorrect: ABD
SIP trunks for public switched telephone network (PSTN) access are an important new access method for business collaboration. Service providers throughout the world offer SIP trunking as an alternative to traditional TDM (T1/E1) connections.
A discard digits instruction (DDI) removes a portion of the dialed digit string before passing the number on to the adjacent system. A DDI must remove portions of the digit string, for example, when an external access code is needed to route the call to the PSTN, but the PSTN switch does not expect that access code
Question No: 14
Which CAC configuration on a gatekeeper restricts to 10 G.711 audio calls?
A. Use the command bandwidth 10.
B. Use the command bandwidth 1280.
C. Use the command bandwidth 160.
D. Use the command bandwidth session 10.
Question No: 15
Which command should you use on the gatekeeper to specify the address of a Cisco Unified Communications Manager IP address?
Question No: 16
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 u2018SAFu2019 to indicate that this trunk is used for SAF.
Question No: 17
Which interface can you use to configure alarms and traces in Cisco Unified
A. Cisco Unified Reporting
B. Cisco Unified Serviceability
C. Cisco Unified Communications Manager Administration
D. disaster recovery system
E. Cisco Unified OS Administration
F. real-time monitoring
Question No: 18
Refer to the exhibit.
All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL. For the HQ phones always to use the hardware conference bridge as a first choice, which configuration should be implemented?
A. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. Ensure that the instance ID for the hardware conference bridge is 0.
B. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. The hardware conference bridge must be configured first.
C. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first.
D. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Configure an additional HQ_MRGL_2. Add the HQ_MRG to HQ_MRGL. Add HQ_MRG_2 to HQ_MRGL_2. The HQ_MRGL should be assigned to the HQ phones. The HQ_MRGL_2 should be assigned to the HQ device pool.
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