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QUESTION 222
Cisco Telepresence Systems that are calling from a CUCM cluster to an H.323 registered device on a Cisco VCS Control do not work.
However, calls from the H.323 VCS registered devices to the same CUCM registered systems do work. What are two possible reasons?

A. the call from CUCM to VCS Control does not match any search rule
B. Calls from CUCM add “:5060” or “:5061” after the SIP address, unlike the VCS Control
C. both systems do not support TLS encryption
D. the SIP trunk has not enabled bidirectional mode
E. Cisco VCS Control is registered in the wrong partition

Answer: A

QUESTION 223
Which CUCM database replication issue would cause CUCM nodes to generate the error “Reverse DNS lookup failed”?

A. DNS server is down
B. DNS server reported that no IP address was returned for a requested A record lookup
C. DNS server reported that no HOSTNAME was returned for a requested A record lookup
D. DNS server reported that no IP address existed for a requested SRV record lookup
E. DNS server reported that no HOSTNAME was returned for a requested PTR record lookup

Answer: C

QUESTION 224
After you upgrade a CUCM cluster,users are unable to log in to their phones.
Which three actions must you take to correct the problem? (3)

A. restart TOMCAT service on all servers
B. regenerate the Tomcat.pem certificate on the PUB only
C. reconfigure the Extension Mobility feature on the PUB and rebuild the PUB
D. regenerate the Tomcat.pem certificate on the PUB & SUB
E. reboot all servers in the cluster
F. restart the TVS and TFTP services on all servers

Answer: ADF

QUESTION 225
Users in your enterprise can establish PSTN calls, but users notice that when they attempt to perform a transfer the call FAILS.
Which two actions must you take to troubleshoot the problem?

A. Verify that media resources are assigned to the transcoder
B. verify that MTP resources are registered with CUCM
C. restart CUCM services
D. Use RTMT Performance monitoring to verify that an MTP device is available to support supplementary services
E. restart Cisco Serviceabilty tool

Answer: BD

QUESTION 226
You have received an SNMP notification that the phones in your enterprise are failing to receive TFTP updates.
Which action must you take to troubleshoot the problem?

A. Verify that Unity Connection is replicating with a status of 5
B. verify that port monitoring is enabled
C. reboot the TFTP server
D. Verify DB replication
E. Debug the voice gateway to locate SIP traces

Answer: C

QUESTION 227
After you deploy a cluster with LDAP authentication, you receive multiple reports that users are unable to log in to their phones with EXT MOB.
Which two steps must you take to troubleshoot the problem?

A. Restart the DirSync service in the CUCM PUB
B. check whether the LDAP configuration requires SSL
C. check whether the LDAP server is acting as a global catalog server
D. Check whether CUCM is running
E. download and install the Sun LDAP connector

Answer: BD

QUESTION 228
IP phone users on your network report hearing echoes of their own voices during calls.
Which three actions correct the problem (3)?

A. adjust padding & receive levels
B. confirm that the users turn off their headsets when their phones are in speaker mode
C. verify that the most recent software versions are in use and the latest patches are applied
D. install a new motherboard in Cisco IOS Router
E. Bind media to the switch port instead of the incoming dial peer
F. remove the bearer capability for voice traffic

Answer: BC

QUESTION 229
An organization is setting up integration of its CUCM cluster to a newly deployed VCS cluster. THe requirement is to allow the CUCM registered endpoints to be able to call VCS registered video endpoints and send a call from IP phone to VCS whenever an IP phone dials any number with the prefix 555 followed by an extension.
Which CUCM construct must be configured to fulfill this req?

A. SIP Route Pattern
B. Local Route Group
C. Translation Pattern
D. SIP Dial Rules
E. Route Pattern

Answer: A

QUESTION 230
After migrating a cisco IP Phone to a new cluster the phone continue to register with its old CUCM cluster. IP Phone status ERROR: TRUST LIST UPDATE FAILED
Which two actions must you take to troubleshoot the problem?

A. CallManager services are not running on the destination cluster
B. the new TFTP server is not in the ITL file
C. the phone is not provisioned correctly on the destination cluster
D. the phone cannot reach its new TFTP server

Answer: BD

QUESTION 231
An engineer has a SIP TRUNK configured between CUCM and VCS cluster.
When a call is made from a Telep EX60 that is registered to the VCS to an IP PHONE 9971 that is registered on the CUCM, it rings.
But upon picking up the call, a busy tone is heard.
What should be checked to resolve this issue?

A. CUCM zone on the VCS
B. SIP trunk registration
C. authentication on the SIP trunk
D. SIP trunk and phone region settings

Answer: D

QUESTION 232
After you install Cisco Jabber client, it fails to register with the CUCM server.
Which two actions must you take to troubleshoot the problem?

A. verify that the name of the configuration file is correct
B. verify Layer 3 connectivity on the gateway
C. verify that the corporate firewall allows connections TO and FROM Jabber client
D. verify that the LMHOST files is installed on the PC
E. reboot CUBE gateway

Answer: BC

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2018.Jan.9 new 300-075 Questions updated!

QUESTION 189
Which of the following are two functions that ensure that the telephony capabilities stay operational in the remote location Cisco Unified SRST router? (Choose two)

A. Automatically detecting a failure in the network.
B. Initiating a process to provide call-processing backup redundancy.
C. Notifying the administrator of an issue for manual intervention.
D. Proactively repairing issues in the voice network.

Answer: AB

QUESTION 190
Which three of the following are steps in configuring MGCP Fallback and Cisco Unified SRST? (Choose three)

A. Define the SRST reference for phones in the Device Pool configuration
B. Enable and configure the MGCP fallback and Cisco Unified SRST features on the IOS gateways.
C. Implement a simplified SRST dial plan on the remote-site-gateways to ensure connectivity for remote-site phones in SRST mode.
D. Enable SIP trunking between both remote and hub sites to provide mesh coverage.
E. Define the SRST reference in the configuration on the IP Phones.
F. Enable and configure the MGCP fallback on the IOS gateway but not Cisco Unified SRST since it is enabled automatically.

Answer: ABC

QUESTION 191
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:
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.

QUESTION 192
Refer to the exhibit. Which trunks would be most suitable for Connection Y?

A. an H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (gatekeeper-controlled)
C. a SIP trunk on the U.S. cluster and an intercluster trunk on the remote cluster
D. intercluster trunk (nongatekeeper-controlled)
E. No extra connections are required. As long as IP connectivity exists, you need only configure a route pattern for each site. The Cisco Unified Communications Manager will automatically forward the calls over the WAN if the destination directory number is not registered locally.

Answer: D

QUESTION 193
Which two features require or may require configuring a SIP trunk? (Choose two.)

A. SIP gateway
B. Call Control Discovery between a Cisco Unified Communications Manager and Cisco Unified Communications Manager Express
C. Cisco Device Mobility
D. Cisco Unified Mobility
E. registering a SIP phone

Answer: AB
Explanation:
All protocols require that either a signaling interface (trunk) or a gateway be created to accept and originate calls. Device mobility allows Cisco Unified Communications Manager to determine whether the phone is at its home location or at a roaming location. Cisco Unified Mobility gives users the ability to redirect incoming IP calls from Cisco Unified Communications Manager to different designated phones, such as cellular phones.

QUESTION 194
A Cisco 3825 needs to be added in Cisco Unified Communications Manager as an H.323 gateway. What should the gateway type be?

A. H.323 gateway
B. Cisco 3825
C. Cisco 3800 series router. The specific model will be selected after the Cisco 3800 is selected.
D. The gateway can be added either as an H.323 gateway or Cisco 3800 series router.
E. The gateway can be added either as an H.323 gateway or Cisco 3825 series router.

Answer: A

QUESTION 67
When an incoming PSTN call arrives at an MGCP gateway, how does the calling number get normalized to a global E.164 number with the + 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 MGCP gateway.

Answer: D
Explanation:
Configuring calling party normalization alleviates issues with toll bypass where the call is routed to multiple locations over the IP WAN. In addition, it allows Cisco Unified Communications Manager to distinguish the origin of the call to globalize or localize the calling party number for the phone user.

QUESTION 195
When an incoming PSTN call arrives at an MGCP gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager?

A. Normalization is done by configuring the significant digits for inbound calls on the MGCP gateway.
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 MGCP gateway.

Answer: A

QUESTION 196
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