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300-075 Exam Questions - Online Test


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Q1. Refer to the exhibit. 

The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number. Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. 

What should the AAR group prefix be? 

A. 9 

B. 91 

C. none 

D. + 

E. +1 

Answer:

Q2. Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three.) 

A. Configure a phone NTP reference. 

B. Configure an SRST reference. 

C. Configure the SIP registrar. 

D. Configure voice register global dn. 

E. Configure voice register pool. 

F. Configure telephony service. 

Answer: B,C,E 

Q3. On which Cisco Unified Communications Manager configuration parameter does the CODEC that a Cisco IP Phone uses for a call depend? 

A. enterprise parameters 

B. media resources 

C. physical location 

D. region 

E. location 

Answer:

Q4. Refer to the exhibit. 

Assume that NANP is being used and 9 is used for PSTN access code Long distance national calls are preceded with 1. 

How should the HQ Cisco Unified Communications Manager be configured for calls to 3XXX to be sent to the gatekeeper at 1 0 1 6 1 with PSTN backups? 

A. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two route groups The first route group contains the H 225 trunk The second route group contains the MGCP gateway with prefix digits 1 408555 for the outgoing called number. 

B. Configure a route pattern for 1#3XXX Assign this route pattern to a route list that points to a route group that lists the H 225 trunk as first choice and the MGCP gateway as a second choice. 

C. Configure a route pattern for 4085543XXX. Assign this route pattern to a route list that points to two route groups. The first route group contains the H 226 trunk The second route group contains MGCP gateway. 

D. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two route groups The first route group contains the H 225 trunk The second route group contains MGCP gateway with prefix digits 91 408554 for the called number. 

Answer:

Explanation: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b03rtgrp.html #wpxref46617 

Q5. Which statement is true regarding the configuration of SAF Forwarder? 

A. In a multisite dial plan, SAF Forwarders may exist in multiple autonomous systems. 

B. The client label that is configured in Cisco Unified Communications Manager must match the configuration on the SAF Forwarder router. 

C. There should not be multiple nodes of Cisco Unified Communications Manager clusters acting as SAF clients. 

D. The destination IP address must match the loopback address of the SAF router. 

Answer:

Q6. 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 

Q7. Which command displays the detailed configuration of all the Cisco Unified IP phones, voice ports, and dial peers of the Cisco Unified SRST router? 

A. show call-manager-fallback all 

B. show dial-peer voice summary 

C. show ephone summary 

D. show voice port summary 

Answer:

Q8. Which three commands are necessary to override the default CoS to DSCP mapping on interface Fastethernet0/1? (Choose three.) 

A. mls qos map cos-dscp 0 10 18 26 34 46 48 56 

B. mls qos map dscp-cos 8 10 to 2 

C. mls qos 

D. interface Fastethernet0/1mls qos trust cos 

E. interface Fastethernet0/1mls qos cos 1 

F. interface Fastethernet0/2mls qos cos 1 

Answer: A,C,D 

Q9. Which two actions ensure that the call load from Cisco TelePresence Video Communication Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes? (Choose two.) 

A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses. 

B. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses. 

C. Create one neighbor zone in VCS for each Unified CM node. 

D. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node. 

E. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node. 

Answer: A,D 

Q10. Refer to the exhibit. A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. The user in RTP sees the message "Not Enough Bandwidth" on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.) 

A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings. 

B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings. 

C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. 

D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings. 

E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. 

F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings. 

Answer: B,F 

Explanation: 

Incorrect Answer: A, C, D, E Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml