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2021 Nov cisco 300-075 vce:
Q11. What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
A. EF/46
B. CS6/48
C. AF41/34
D. CS3/24
E. CS4/32
Answer: E
Q12. Company A has deployed a VCS Control and is attempting to register a third-party endpoint. The engineer has confirmed that no traffic is being blocked for the endpoint and it is receiving a valid IP address. Which option could be the cause of this registration failure?
A. Third-party endpoints are not compatible with VCS Control, only with VCS Expressway.
B. Cisco Unified Communications Manager is required in addition to the VCS Control.
C. An incorrect SIP domain is configured on the VCS Control for the endpoint.
D. The VCS Control must be deployed together with VCS Expressway before endpoints can register to either one.
Answer: C
Q13. 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
Q14. Which commands are needed to configure Cisco Unified Communications Manager Express in SRST mode?
A. telephony-service and srst mode
B. telephony-service and moh
C. call-manager-fallback and srst mode
D. call-manager-fallback and voice-translation
Answer: A
Q15. How is a SIP trunk in Cisco Unified Communications Manager configured for SIP precondition?
A. The configuration is done by selecting a SIP precondition trunk for trunk type.
B. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk.
C. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk.
D. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. The new SIP profile must then be assigned to the SIP trunk.
Answer: D
Latest voice tut 300-075 exam:
Q16. 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
Q17. 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 configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001), the calls fail from the PSTN. Which two of the following configurations if applied to the router, would remedy this situation? (Choose two.)
A. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:15
B. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:13
C. voice translation-rule 1rule 1 /228821 …$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in
D. voice translation-rule 1rule 1 /228822…$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in
E. The router does not need to be configured
Answer: A,D
Q18. Which process can localize a global E.164 with + prefix calling numbers for inbound calls to an IP phone so that users see the calling number in a local format?
A. Calling number localization is done using translation patterns.
B. Calling number localization is done using route patterns.
C. Calling number localization is done by configuring a calling party transformation CSS at the phone.
D. Calling number localization is done by configuring a calling party transformation CSS at the gateway.
E. Calling number localization is done by configuring the phone directory number in a localized format.
Answer: C
Q19. When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country code 1 and the 10 digits. How can this be accomplished?
A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI.
B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns.
C. Use a calling party transform mask for each route group in the corresponding route list configuration. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns.
D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls.
Answer: C
Explanation:
Incorrect Answer: A, B, D calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03trpat.ht ml
Q20. 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