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Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)

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Total Questions : 174

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Question # 1

Refer to the exhibit.

Question # 1

A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

Options:

A.  

Allow Passthrough of Configured Line Device Caller Information must be enabled.

B.  

Accept Audio Codec Preferences in Received Offer must be set to On.

C.  

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

D.  

Early Offer for G Clear Calls must be enabled.

Discussion 0
Question # 2

CollabCorp is a global company with two clusters, emea.collab corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make inter-cluster calls via URI. but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)

Options:

A.  

directory URI partition

B.  

SIP route pattern

C.  

intercluster trunk

D.  

calling search space and partition

E.  

SIP trunk

Discussion 0
Question # 3

A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through 8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?

Options:

A.  

813510[012345]

B.  

813510[12345]

C.  

813510[^0-5]

D.  

81XXXXX

Discussion 0
Question # 4

An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CM

E.  

Which configuration must be used to start this process?

Options:

A.  

voice service volp

Ip address trusted list

B.  

voice service volp

enablo ip address trust authentication

C.  

voice service volp

enable Ip address trust list

D.  

voice service volp

ip address trusted authenticate

Discussion 0
Question # 5

Refer to the exhibit.

Question # 5

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:

A.  

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

B.  

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

C.  

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.

D.  

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.

E.  

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.

Discussion 0
Question # 6

An engineer must route all SIP calls in the form of @example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)

Options:

A.  

example.com@gateway.corporate.local

B.  

*@example.com

C.  

gateway.corporate.local

D.  

example.com

E.  

*.*

Discussion 0
Question # 7

Refer to the exhibit.

Question # 7

An administrator just Implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily. Which action resolves the issue?

Options:

A.  

Disable SIP Red XX Options.

B.  

Change to a range that does not result in MTP.

C.  

Change OTMF Signaling Method to "No Preference".

D.  

Change DTMF Signaling Method to "RFC 4733".

Discussion 0
Question # 8

End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Options:

A.  

Contact: header of the 200 OK response

B.  

Allow: header if the 200 OK response

C.  

o= line of SDP content

D.  

c= line of SDP content

Discussion 0
Question # 9

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

Discussion 0
Question # 10

After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

Options:

A.  

Router(config)# voice service voip

Router(conf-voi-serv)#allow-connections h323 to h323

B.  

Router(config)#dial-peer voice 2 voip

Router(config-dial-peer)#no vad

C.  

Router(config)# voice service voip

Router(conf-voi-serv)#allow-connections voice-mail mod

D.  

Router(config)# voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

Discussion 0
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