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300-815 Exam Dumps - Cisco CCNP Collaboration Questions and Answers

Question # 34

An administrator is implementing a new dial-plan on Cisco Unified Border Element. The administrator must ensure that incoming dial-peers are matched based on the IP address from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?

Options:

A.

dial-peer voice 1 voip

incoming url via

B.

dial-peer voice 1 voip

incoming url request

C.

dial-peer voice 1 voip

incoming called-number

D.

dial-peer voice 1 voip

incoming url to

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Question # 35

Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?

Options:

A.

request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP translation profile configuration

B.

sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration

C.

request Invite sip-header Diversion modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP profile configuration

D.

request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration

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Question # 36

Refer to the exhibit.

A collaboration engineer investigates an issue where the user calls that match the local Jabber domain are routed back to Cisco UCM. Which action resolves this problem?

Options:

A.

Check the check box to block the pattern for the local domain.

B.

Create another route SIP pattern for the local domain with the block pattern selected.

C.

Create a translation pattern to block calls to the local domain.

D.

Create a transformation pattern to block calls to the local domain.

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Question # 37

Refer to the exhibit.

A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?

Options:

A.

No codecs are configured on the dial peers

B.

The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.

C.

The destination pattern is incorrect for the dialed number.

D.

VAD was not disabled on the outgoing dial poor.

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Question # 38

Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)

Options:

A.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

B.

Ask the firewall administrator to change the ports to TCP.

C.

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

D.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

E.

Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

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Question # 39

Refer to the exhibit.

Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?

Options:

A.

rule 1 /^9\(.*\)/A1/

B.

rulel /.*\(3548S\)/^1/

C.

rulel /^9\(\d*\)/^1/

D.

rule 1/^9123548/^1/

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Question # 40

Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites. How is the issue solved without increasing bandwidth utilization on the WAN links?

Options:

A.

Disable Call Admission Control and let the calls use the amount of bandwidth they require.

B.

Configure Call Queuing so that the user waits until there is bandwidth available

C.

Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.

D.

Reroute all calls through the PSTN and avoid using WAN.

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Question # 41

An IP Telephony administrator is deploying IP phones The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111 2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two )

Options:

A.

Option A

B.

Option B

C.

Option C

D.

Option D

E.

Option E

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Question # 42

Refer to the exhibit.

An administrator just upgraded to Cisco UCM to version 14 and started SIP implementation with some new SIP trunks. During the testing, an error was reported when making a call. Which action resolves the issue?

Options:

A.

Change to a valid range.

B.

Change DTMF Signaling Method to "RFC 4833*.

C.

Make sure the destination number is configured correctly and the device is registered.

D.

Disable SIP Rel1XX Options.

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Question # 43

What are the elements for Device Mobility configuration?

Options:

A.

physical location, device pool, and Device Mobility group

B.

device pool, Device Mobility group, and region

C.

physical location. Device Mobility group, and region

D.

device pool, Device Mobility group, and Cisco IP phone

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Exam Code: 300-815
Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
Last Update: Feb 23, 2025
Questions: 174
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