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Exam Code: 300-075
Exam Name: Implementing Cisco IP Telephony & Video, Part 2(CIPTV2)
Updated: Mar 10, 2017
Exam Number:300-075 CIPTV2
Associated Certifications:CCNP Collaboration
Duration:75 minutes (50-60 questions)
Exam Policies:Read current policies and requirements
Exam Tutorial:Review type of exam questions
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2017 Cisco 300-075 VCE (#11-20) from Pass4itsure:
300-075 vce Q11. 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
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
300-075 vce Q12. 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 is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
SX20 System Information
What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two.)
A. Incorrect username and password
B. Wrong SIP domain configured.
C. User is not associated with the device.
D. IP or DNS name resolution issue.
E. CSF Device is not registered.
F. IP Phone DN not associated with the user.
300-075 vce Q13. 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.
300-075 vce Q14. Which statement about the function of a gatekeeper is true?
A. A gatekeeper improves call routing between servers within a single Cisco Unified Communications Manager cluster.
B. A gatekeeper can replace the dial plan of a Cisco Unified Communications Manager cluster.
C. A gatekeeper can simplify the dial plan between many different Cisco Unified Communications Manager clusters.
D. Gatekeepers can be implemented to deploy RSVP-based CAC.
Q15. Which symbol is required for globalized call routing?
300-075 vce Q16. 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
300-075 vce Q17. Which action configures phones in site A to use G.711 to site B but to use G.729 to site C?
A. Configure Cisco Unified Communications Manager regions.
B. Configure Cisco Unified Communications Manager locations.
C. Configure transcoder resources in Cisco Unified Communications Manager.
D. Configure a gatekeeper.
300-075 vce Q18. Which two entities could be represented by device mobility groups? (Choose two.)
C. directory numbers
300-075 vce Q19. Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server.
B. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
C. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
E. Verify that the H.323 redundant connection is active.
F. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
300-075 vce Q20. Refer to the exhibit.
Assume that the HQ phones have access to the HQ partition, and BR phones have access to the BR partition. Which set of implementations would best address the overlapping directory number extensions for intersite (WAN) calling between the HQ site and the BR site?
A. Configure a route pattern 8222.XXX for site HQ, and assign it to partition HQ. Configure the called party DDI of Predot.Configure a route pattern for site BR 8111.[1-3]XXX, and assign it to partition BR. Configure called party DDI Predot.Use the local gateway at each site. Prefix the appropriate site code for the calling number.
B. Configure a single route pattern for both sites 8[12,12,12].[1-32]XXX. Use a route list that contains the local route group for each site. Prefix the appropriate site code for the calling number.
C. Configure a translation pattern 8222.XXX for site HQ, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.
D. Configure a translation pattern 8222.XXX for site HQ, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.
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