[Free] 2017(Oct) Dumps4cert Testinsides Cisco 300-075 Dumps with VCE and PDF Download 161-170

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Implementing Cisco IP Telephony amp; Video, Part 2 v1.0

Question No: 161

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

  1. iSAC

  2. GSM-FR

  3. iLBC

Answer: C

Question No: 162

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?

  1. Normalization is done using translation patterns.

  2. Normalization is done using route patterns.

  3. Normalization is done using the gateway incoming called party prefixes based on number type.

  4. Normalization is done using the gateway incoming calling party prefixes based on number type.

  5. Normalization is achieved by local route group that is assigned to the MGCP gateway.

Answer: D

Explanation: Incorrect answer: ABCE

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.

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

Question No: 163

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

How many calls can be placed to Cluster B?

  1. three G.729 calls

  2. one G.711 call

  3. one G.711 and three G.729 calls

  4. There is no limit for incoming calls to Cluster B. Outgoing calls are limited to one G.711 and three G.729 calls.

Answer: A

Question No: 164

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?

  1. ephone-dn 1 dual-line number 7001 description 7001

    name 7001

    ephone-dn-template 5

    This DN is learned from srst fallback ephones

  2. ephone-dn 1 dual-line number 7001 description 7001

    name 7001

    ephone-dn-template 3

    This DN is learned from srst fallback ephones

  3. ephone-dn 1 number 7001 description 7001 name 7001

    ephone-dn-template 5

    This DN is learned from srst fallback ephones

  4. ephone-dn 1 number 7001 description 7001 name 7001

ephone-dn-template 3

This DN is learned from srst fallback ephones

Answer: A

Question No: 165

Which command is needed to utilize local dial peers on an MGCP-controlled ISR during an SRST failover?

  1. ccm-manager fallback-mgcp

  2. telephony-service

  3. dialplan-pattern

  4. isdn overlap-receiving

  5. voice-translation-rule

Answer: A

Question No: 166

Which two statements regarding IPv4 Static NAT address 209.165.200.230 has been configured on a VCS Expressway are true? (Choose two.)

  1. The Advanced Networking or Dual Network Interfaces option key has been installed.

  2. VCS rewrites the Layer 3 source address of outbound SIP and H.323 packets to 209.165.200.230.

  3. VCS applies 209.165.200.230 to outbound SIP and H.323 payload messages.

  4. With static NAT enabled on the LAN2 interface, VCS applies 209.165.200.230 to outbound H.323 and SIP payload traffic exiting the LAN1 interface.

Answer: A,C

Question No: 167

The network administrator of Enterprise X receives reports that at peak hours, some calls between remote offices are not passing through. Investigation shows no connectivity problems. The network administrator wants to estimate the volume of calls being affected by this issue.

Which two RTMT counters can give more information on this? (Choose two.)

  1. CallsRingNoAnswer

  2. OutOfResources

  3. LocationOutOfResources

  4. RequestsThrottled

  5. CallsAttempted

Answer: B,C

Question No: 168

Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate external calls, VCS Expressway is deployed and traversal server zone is set up there. Video endpoints inside Company X have registered, but are unable to receive calls from outside endpoints. Which option could be the cause of this issue?

A.

The traversal zone on the VCS Control does not have a search rule configured.

B.

The access control list on the VCS Control must be updated with the IP for the external users.

C.

When a traversal zone is set up on VCS Control only outbound calls are possible.

D.

The local zone on the VCS Control does not have a search rule configured.

Answer: B

Question No: 169

Which statement is correct about AAR?

  1. The end users sees, quot;Network Congestion Rerouting?quot; but AAR is otherwise transparent to the end user and works without user intervention.

  2. AAR will display quot;not enough bandwidthquot; on the IP phone while it reroutes the call.

  3. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.

  4. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth.

Answer: A

Explanation: Incorrect answer: BCD

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

Question No: 170

Which option is a valid test scenario to verify that Cisco Unified Communications Manager failover works and endpoints register with the backup call manager?

  1. During a predetermined maintenance window, stop the Cisco IP Phone Services service on the primary Unified CM. Devices should reregister with the backup Unified CM in the Cisco CallManager Group.

  2. During a predetermined maintenance window, stop the Unified CM service on the Publisher. Devices should reregister with the backup Publisher in the Cisco CallManager Group.

  3. During a predetermined maintenance window, stop the TFTP service on the primary call manager. Devices should reregister with the backup Unified CM in the Cisco CallManager Group.

  4. During a predetermined maintenance window, stop the Unified CM service on the primary call manager. Devices should reregister with the backup Unified CM in the CallManager Group.

Answer: D

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