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300-815
Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) Questions and Answers

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Viewing questions 31-40 out of questions

Questions # 31:

Refer to the exhibit.

Question # 31

The Cisco Unified Border Element receives an INVITE matching inbound dial peer 5002. The outbound dial peer supports only iLBC. and a Local Transcoding Interface is allocated. Based on the configuration and SDP from the INVITE message, which codec is chosen by Cisco Unified Border Element for the inbound call leg?

Options:

G.711 A-law

G.711 U-law

G.729r8

G.729br8

Answer

Questions # 32:

Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button Which configuration parameter causes this problem?

Options:

The voicemail pilot number for call forwarding is missing from the ephone-dn

The voicemail pilot number is missing from the call handling on Cisco Unity Express

The voicemail pilot number is missing from the telephony service configuration on Cisco UCME

The voicemail pilot number for call forwarding is missing from the ephone

Answer

Questions # 33:

Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

Options:

Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.

A unicast music on hold audio source must be configured.

Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.

The maximum number of callers allowed in queue must be 10.

The phone button template must have the Queue Status Softkey configured.

Answer

A, C

Explanation

[Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide- 1201_chapter_01001101.html#CUCM_RF_C960BC9A_00, ]

Questions # 34:

Refer to the exhibit.

Question # 34

In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)

Options:

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

Answer

A, D

Questions # 35:

An administrator must control the number of calls to a remote specific site to reduce bandwidth constraints. The users on that remote site report bad quality of the calls passing through that WAN link. Which action must the administrator take in Cisco UCM to resolve the issue?

Options:

Use RSPV.

Use Location Bandwidth Manager.

Use Expressway deployment.

Use Call Allow Controller.

Answer

Questions # 36:

An administrator is trying to apply configuration changes on Cisco CME. When the users registered on Cisco CME to dial a local number to a PSTN call, the Cisco CME sends an incorrect number of digits. What translation rule fixes the issue and sends the correct number of digits?

Options:

voice translation-rule 1

rule 1 /^4...$/2404\0/ type any national plan any Isdn

voice translation-rule 1 rule 1 // // type any subscriber plan any isdn

voice translation-rule 1 rule 1 /^4...S/ /9132404 0/ type any subscriber plan any Isdn

voice translation-rule 1

rule 1 /^4...V /2404\0/ type any subscriber plan any isdn

Answer

Questions # 37:

What is first preference condition matched in a SIP-enabled incoming dial peer?

Options:

incoming uri

target carrier-id

answer-address

incoming called-number

Answer

Explanation

[Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8, ]

Questions # 38:

Some users report having issues dialing some external numbers when traveling to other locations within the company. The company has five locations in five cities in one country and has an egress gateway in each location for TEHO. The configuration has no specific entry stating that the roaming users are using the local gateway, but calls are going out. How is a verification of the call routing in such a specific configuration performed to further identify the problem?