[Nov 2018] 300 075 ciptv2


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New Cisco 300-075 Exam Dumps Collection (Question 16 - Question 22)

New Questions 16

If your IP telephony administrator asks you to configure a local zone for your dial plan to control the volume of calls between two end points in a centralized multisite environment, which two types of Call Admission Control can be implemented? (Choose two.)

A. locations based

B. automated alternate routing

C. gatekeeper based

D. SRST

E. Cisco Unified Communications Manager based

Answer: A,B

Explanation: Incorrect: CDE

Location-based call admission control (CAC) manages WAN link bandwidth in Cisco Unified Communications Manager. Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number.

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html

#wp1067747


New Questions 17

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


New Questions 18

What happens if location-based CAC is used and there is no bandwidth available when a remote caller is placed on hold?

A. Cisco Unified Communications Manager sends TOH rather than MOH.

B. Cisco Unified Communications Manager terminates the call.

C. Cisco Unified Communications Manager plays default MOH.

D. Cisco Unified Communications Manager attempts to reconnect the call immediately.

Answer: A


New Questions 19

Refer to the exhibit.

Which statement about the configuration between the Default and BR regions is true?

A. Calls between the two regions can use either 64 kbps or 8 kbps.

B. Calls between the two regions can use only the G.729 codec.

C. Only 64 kbps will be used between the two regions because the link is "lossy".

D. Both codecs can be used depending on the loss statistics of the link. When lossy conditions are high, the G.711 codec will be used.

Answer: B


New Questions 20

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


New Questions 21

This is the configuration on the voice gateway:

telephony-service max-ephones 30

max-dn 60 preference 0 srst mode auto-provision all srst dn line-mode dual

srst dn template 3

srst ephone description srst fallback auto-provision phone srst ephone template 5

Which ephone-dn would be expected upon activation of SRST?

A. ephone-dn 1 dual-line number 7001

description 7001

name 7001

ephone-dn-template 5

This DN is learned from srst fallback ephones

B. ephone-dn 1 dual-line number 7001

description 7001

name 7001

ephone-dn-template 3

This DN is learned from srst fallback ephones

C. ephone-dn 1

number 7001

description 7001

name 7001

ephone-dn-template 5

This DN is learned from srst fallback ephones

D. ephone-dn 1

number 7001

description 7001

name 7001

ephone-dn-template 3

This DN is learned from srst fallback ephones

Answer: A


New Questions 22

You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2- Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?

A. G.722

B. G.711

C. G.729

D. iSAC

E. GSM-FR

F. iLBC

Answer: C



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