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300-815 Cisco Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) Free Practice Exam Questions (2025 Updated)

Prepare effectively for your Cisco 300-815 Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) certification with our extensive collection of free, high-quality practice questions. Each question is designed to mirror the actual exam format and objectives, complete with comprehensive answers and detailed explanations. Our materials are regularly updated for 2025, ensuring you have the most current resources to build confidence and succeed on your first attempt.

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Total 221 questions

An administrator is working on an issue between the customer s Cisco Unified Border Element and the service provider. The provider only wants to see mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?

A.

midcall-signallng passthru

B.

midcall-signaling preserve-codec

C.

no update-callerid

D.

midcall-signaling passthru media-change

Refer to the exhibit.

A collaboration engineer is troubleshooting an issue where external callers cannot leave voicemail messages. Also, internal users report hearing the reorder tone (fast busy) when they attempt to retrieve voicemail messages from their Cisco IP phones. Which action resolves the issue?

A.

Ensure that Cisco UCM can resolve the destination address via DNS.

B.

Start the Cisco Call Manager service at the destination.

C.

Ensure that the SIP Trunk Security Profile is configured to use UDP for transport.

D.

Verify that the correct port numbers are used for the SIP trunk.

An engineer is troubleshooting an intersite call between two endpoints where the call fails and the message “Not Enough Bandwidth” is displayed. G.729 codec is in use on both sites. First, calls are being properly routed, and the issue happens after the third call is established and the bandwidth utilization between the two sites is under 50%. Which configuration in Cisco UCM must be adjusted to resolve the issue?

A.

transcoder

B.

location

C.

route pattern

D.

translation pattern

Users report silence on the line when they try to connect to external voice numbers. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. The MGCP server seems to be responding. Which action must the engineer take to begin to investigate the media for this type of issue?

A.

Run RTMP on the Cisco router.

B.

Verify that the MGCP service is running under serviceability in Cisco UCM.

C.

Run the debug mgcp media command on the Cisco router.

D.

Run the debug H.323 gateway command on the Cisco router.

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

A.

can be associated under the route group

B.

can be associated only under the route list

C.

chooses the route group that is configured under the device pool of the calling-party device

D.

chooses the route group that is configured under the device pool of the called-party device

E.

can be assigned directly to the route pattern

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?

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.

Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)

A.

TokenID

B.

username and secret key

C.

TLS certificates

D.

passwords

E.

FQDN of the servers defined in DNS

Refer to the exhibit.

An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?

A.

SIP binding in SIP configuration mode:

voice service voip sip

bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0

B.

SIP binding In SIP configuration mode:

voice service volp

sip

bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1

C.

SIP binding In dial-peer configuration mode:

dial-peer voice 300 voip

voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1

D.

SIP binding in dial-peer configuration mode:

dial-peer voice 100 volp

voice-class sip bind control source-interface GigabitEthernetO/0/0

voice-class sip bind media source-interface GigabitEthernetO/0/0

Refer to the exhibit. This message is sent to the device being placed on hold for the Music On Hold audio setup. The held party reports receiving dead air rather than music when the call was put on hold. The software Music On Hold server on Cisco UCM is used in this scenario. Assume that the audio leg between the Music On Hold server and the held device uses G.711, and the relevant region relationship is configured for 64 kbps. What is the cause of the issue?

A.

The bandwidth configured for this region relationship is too low and must be increased to 96 kbps or higher.

B.

The device that is placed on hold does not support G.711, and a transcoder could not be allocated for the call.

C.

Cisco UCM is sending a=inactive to the held device.

D.

The Music On Hold server does not support G.711 and a transcoder could not be allocated for the call.

An organization configures a SIP trunk in Cisco UCM to connect to another system. These requirements must be met:

1. Use a specific IP address for SIP signaling.

2. Encrypt the signaling traffic.

3. Restrict which devices can use the SIP trunk. 4 Simplify SIP signaling.

Drag and drop the Cisco UCM configuration steps from the left onto the order on the right to achieve these goals.

Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.

A customer reports that outbound calls from Cisco UCM to Cisco Unified Border Element are not getting early media. During troubleshooting, the customer finds that Cisco UCM is sending a delayed offer SIP INVITE. Cisco Unified Border Element responds with a 183 Session Progress with SDP message. However, the customer finds that Cisco UCM does not respond to the 183 messages. Which action must the customer perform to resolve the problem?

A.

On Cisco Unified Border Element, configure midcall-signaling passthru.

B.

On Cisco UCM, change the "Early Offer support for voice and video calls" field on the SIP profile that is applied to the SIP trunk to Disabled.

C.

On Cisco Unified Border Element, configure early-offer forced.

D.

On Cisco UCM. change the "SIP ReHXX Options" field on the SIP profile that is applied to the SIP trunk to Send PRACK if 1xx Contains SDP.

An engineer configures several calling party transformation patterns in Cisco UCM so that calling numbers are shown in the +E.164 format. The route patterns match seven-digit internal numbers and prefix the numbers with 1408. The SIP trunk matches 1408XXXXXXX and prefixes the number with +. When users make outbound calls, the calling number still shows as seven digits. What is the cause of this problem?

A.

The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied.

B.

Calling transformations cannot be configured on both route patterns and a SIP trunk.

C.

The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.

D.

The engineer must also apply a calling transformation at the route group details level.

Refer to the exhibit.

A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?

A.

Enable 'SIP ReI1XX Options* and -Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.

B.

Enable *Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.

C.

Disable "SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.

D.

Disable "Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.

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?

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

What does VoIP trace in a Cisco Unified Border Element do?

A.

It stores all call trace data in system memory for failed calls.

B.

It stores all can trace data in system memory (or successful and failed calls

C.

It stores all call trace data in system log files for successful and failed calls.

D.

It stores all can trace data in system file logs for failed calls.

Refer to the exhibit. Calls from users to the PSTN in an organization get disconnected with a 408 Request Timeout when the called party is unavailable to pick up the call. Which solution must be used to resolve this challenge?

A.

Configure midcall-signaling preserve-codec.

B.

Choose "Send PRACK" if 1xx contains SDP in the SIP profile.

C.

Configure midcall-signaling passthru media-change

D.

Choose "Send PRACK for all 1xx Messages" in the SIP profile.

Which set of commands binds SIP media and signaling to interface GigabitEthernet0/0 when dial peer 1 is chosen for call routing?

A.

dial-peer voice 1 voip

voice-class source interface GigabitEthernet0/0

B.

voice service voip

bind sip source-interface GigabitEthernet0/0

C.

voice service voip

sip

bind all source-interface GigabitEthernet0/0

D.

dial-peer voice 1 voip

voice-class bind control source-interface GigabitEthernet0/0

voice-class sip bind media source-interface GigabitEthernet0/0

Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system. Each building has is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?

A.

Change the partition of the route pattern

B.

Add a sip trunk inside route group Standard Local Route Group.

C.

Modify the route pattern to add a prefix of 91

D.

Add a local route group on the device pool configuration.

Refer to the exhibit. An administrator is trying to test outbound calls toward the ITSP but cannot complete the call and receives a SIP error. ITSP is consulted, and the issue is that the ANI that is being sent is not the DID provided 8005532447. Which configuration change sends the correct ANI on the INVITE sent to ITSP to fix the error?

A.

voice class sip-profiles 2

request INVITE sip-header To modify “sip:(.*)@” sip:8005532447@

B.

voice translation-rule 3

rule 1/.*/ /8005532447/

C.

voice class sip-profiles 1

request INVITE sip-header Diversion modify “sip:(.*)@” “sip:8005532447@”

D.

voice translation-rule 4

rule 1/^.*\(8005532447\)/ /\1/

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Total 221 questions
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