Ucertify offers free demo for 300 075 ciptv2 exam. "Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)", also known as 300 075 pdf exam, is a Cisco Certification. This set of posts, Passing the Cisco 300 075 ciptv2 exam, will help you answer those questions. The 300 075 vce Questions & Answers covers all the knowledge points of the real exam. 100% real Cisco 300 075 vce exams and revised by experts!
Q1. 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
DX650 Configuration
MRGL
DP
Locations
AARG
CSS
Movi Failure
Movi Settings
Which device configuration option will allow an administrator to assign a device to specific rights for making calls to specific DNs?
A. Media Resource Group List
B. Device Pool
C. Location
D. AAR Group
E. Calling Search Space
Answer: E
Explanation:
A partition comprises a logical grouping of directory numbers (DNs) and route patterns with similar reachability characteristics. Devices that are typically placed in partitions include DNs and route patterns. These entities associate with DNs that users dial. For simplicity, partition names usually reflect their characteristics, such as "NYLongDistancePT," "NY911PT," and so on. A calling search space comprises an ordered list of partitions that users can look at before users are allowed to place a call. Calling search spaces determine the partitions that calling devices, including IP phones, softphones, and gateways, can search when attempting to complete a call.
Q2. Refer to the exhibit.
Which configuration change is needed to enable NANP international dialing during MGCP fallback?
A. Change the dial peer to dial-peer voice 901 voip.
B. Change the dial peer to dial-peer voice 9011 pots.
C. Add the command prefix 011 to the dial peer.
D. Add the command prefix 9011 to the dial peer.
Answer: C
Q3. Which statement about Service Advertisement Framework is true?
A. SAF requires that the EIGRP be configured on all routers, including non-SAF routers.
B. SAF requires that the EIGRP be configured only on SAF routers. Non-SAF routers act as an IP cloud.
C. SAF has no dependency on the underlying routing protocol, as long as it is a dynamic routing protocol. Static routes are not supported.
D. SAF operates on any dynamic or static IP routing configuration. SAF is totally independent of the underlying routing protocol.
Answer: D
Explanation:
Because Cisco SAF is independent of IP routing and uses underlying Cisco routing technology to distribute service advertisements in a reliable and efficient manner, Cisco SAF will run in networks over any routing protocol they may have in place such as Enhanced Interior Gateway Routing Protocol (EIGRP), Open Shortest Path First (OSPF), Exterior Border Gateway Protocol (EBGP) over an MPLS service, or static routing (Figure 2).
http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6554/ps6599/ps10822/whitepaper_c11-636604.html
Q4. Which Cisco IOS command is used to verify that a SAF Forwarder that is registered with Cisco Unified Communications Manager has established neighbor relations with an adjacent SAF Forwarder?
A. show eigrp service-family ipv4 neighbors
B. show eigrp address-family ipv4 neighbors
C. show voice saf dndball
D. show saf neighbors
E. show ip saf neighbors
Answer: A
Explanation:
Incorrect Answer: B, C, D, E
Router# show eigrp service-family ipv4 4453 neighbors
Link: http://www.cisco.com/en/US/docs/ios/saf/command/reference/saf_s1.html
Q5. When using Cisco Unified Communications Manager Express in SRST mode, how many multicast music on hold streams can be utilized by the system at any given time?
A. 3
B. 6
C. 2
D. 4
E. 1
F. 5
Answer: B
Q6. 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.
Answer: C
Explanation:
Incorrect Answer: A, B, D 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.
Link: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03rp.html
Q7. Which statement is not true about GARP?
A. GARP attacks require access to the target LAN or VLAN.
B. GARP can be used for a man-in-the-middle attack.
C. GARP is normally used for HSRP.
D. GARP can be disabled at Cisco IP phones.
Answer: C
Explanation:
Incorrect Answer: A, B, D
GARP (Gratuitous ARP) announce the presence of IP Phone on the network.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/4_0_1/secuphne.html
Q8. You recently implemented call redundancy at a new remote site, and users report that calls are dropped when the remote site supposedly is in SRST. Which two actions must you take to troubleshoot the problem? (Choose two.)
A. Confirm that SRST is configured on the voice gateway.
B. Confirm that the site has an SRST reference that is correctly associated with the Cisco Unified Communications Manager group.
C. Confirm that a calling search space is assigned to the voice gateway in Cisco Unified Communications Manager.
D. Confirm that the site devices are associated with a Cisco Unified Communications Manager group and that four Cisco Unified Communications Manager servers are available.
E. Check the Region settings in Cisco Unified Communications Manager.
F. Restart Cisco Unified Communications Manager services to confirm that the server is working correctly.
Answer: A,B
Q9. What is the purpose of a SAF Client?
A. To decode address information and route calls to and from the end points
B. To pass IP information from the CUCM to the endpoint
C. To learn about and advertise or subscribe information about SAF network services
D. To reside in the Cisco IOS software, and to communicate with the SAF forwarder
Answer: C
Q10. Which two statements are true regarding the implementation of globalized call-routing in terms of localized call egress? (Choose two.)
A. Calling-party numbers are routed from the gateway or trunks to phones.
B. Called-party numbers are routed from the gateway or trunks to phones.
C. Calling-party numbers of internal calls are routed from the gateway or trunks.
D. Calling-party calls are routed to the gateway and trunks.
Answer: A,D
Q11. An update of the configuration using the Cisco CTL client not needed when _______.
A. a Cisco Unified CallManager has been removed
B. an LSC of the IP phone is upgraded
C. a security token is added to the system
D. an IP address of the Cisco TFTP server has been changed
Answer: B
Explanation:
Incorrect Answer: A, C, D The CTL file contains entries for the following servers or security tokens:
. System Administrator Security Token (SAST)
. Cisco CallManager and Cisco TFTP services that are running on the same server
. Certificate Authority Proxy Function (CAPF) .
. TFTP server(s)
. ASA firewall
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_6_1/secugd/secuauth.h tml#wp1028878
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 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
Q13. Which option is known as the location attribute that the global dialplan replication uses to advertise its dial plan information?
A. location controller
B. route pattern
C. route string
D. URI
Answer: C
Q14. When implementing a dial plan for multisite deployments, what must be present for SRST to work successfully?
A. dial peers that address all sites in the multisite cluster
B. translation patterns that apply to the local PSTN for each gateway
C. incoming and outgoing COR lists
D. configuration of the gateway as an MGCP gateway
Answer: B
Q15. Which task must you perform before deleting a transcoder?
A. Delete the dependency records.
B. Unassign it from a media resource group.
C. Use the Reset option.
D. Remove the device pool.
E. Remove the subunit.
F. Delete the common device configuration.
Answer: B