CCNP Collaboration Certification 300-815 CLACCM Exam Dumps Online

300-815 Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) certification exam is for CCNP Collaboration certification, which tests your knowledge of advanced call control and mobility services. It is one of the four concentration exams:

• 300-810 CLICA Implementing Cisco Collaboration Applications (CLICA)
• 300-815 CLACCM Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
• 300-820 CLCEI Implementing Cisco Collaboration Cloud and Edge Solutions (CLCEI)
• 300-835 CLAUTO Implementing Automation for Cisco Collaboration Solutions (CLAUI)

If you are interested in 300-815 CLACCM exam, you need to understand all the exam details and information first. Also, we highly recommend to get the most valid CCNP Collaboration Certification 300-815 CLACCM Exam Dumps Online to prepare for the exam well.

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1. Refer to the exhibit.





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.)

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.

2. Refer to the exhibit.





Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?

The negotiated RTP port is outside of the range described by RFC, so in band DTMFs do not work.

There is SIP Delayed Offer. DTMF is supported only in Early Offer.

The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

No DTMF is negotiated.

3. The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.

You gather the H.225 and H.245 messages for one of the one-way audio calls.

Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

245 Terminal Capability Set

245 Open Logical Channel

225 Connect

245 Open Logical Channel Ack

4. Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

DTMF

BFCP

VIDEO

FAX

AUDIO

5. When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?

ALERTING

PROCEEDING

CONNECT

RINGING

6. 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?

Contact: header of the 200 OK response

Allow: header if the 200 OK response

o= line of SDP content

c= line of SDP content

7. Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

Cisco Unified Communications Manager invoked media termination point resources.

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

A firewall in the media path is blocking TCP ports 16384-32768.

8. An administrator is troubleshooting call failures on an H.323 gateway via the CLI.

To see signaling for media and call setup, which debug must the Administrator turn on?

debug

323 messages

debug

225 asn1

debug

246 asn 1

debug

225 media

debug

323 asn 1

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

incoming uri

target carrier-id

answer-address

incoming called-number

10. 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.)

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

Ask the firewall administrator to change the ports to TC

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

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

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

11. Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

Analysis Manager > Inventory > Trace File Repositories

System > Tools > Trace and Log Central

Voice/Video > Session Trace Log View > Real Time Data

Voice/Video > Session Trace Log View > Open From Local Disk

12. Which description of RTP timestamps or sequence numbers is true?

The sequence number is used to detect losses.

Timestamps increase by the time “carrying” by a packet.

Sequence numbers increase by four for each RTP packet transmitted.

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).

13. A support engineer is troubleshooting a voice network.

When conducting a search for call setup details related to calling search space issues, which trace files should be investigated?

CallManager traces

CTI Manager traces

Cisco IP Manager Assistant

Call logs

14. Refer to the exhibit.





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?

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

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

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

Early Offer for G Clear Calls must be enabled.

15. A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide.

To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)

three-way conference

secure SIP lines

38 fax relay

transcoding

SIP trunk

16. Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?

Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.

Configure IP Address Trusted Authentication for Incoming VoIP Calls.

Configure the command no ip address trusted authenticate under “voice service voip”.

Enable Secondary Dial tone on Analog and Digital FXO Ports.

17. You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration.

Which configuration is occurring in this section?

configuration for a single SIP phone

configuration items common for all SIP phones

configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)

configuration for SIP registrar service

18. Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?

allow-connections sip to sip

voice service voip

voice register global

voice register dn

19. For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

interworking between an OOB method and RFC2833 for flow-around calls

interworking between h245-signal and rtp-nte

interworking between an OOB method and RFC2833 for flow-through calls

interworking between h245-alpha numeric and sip-kpml

20. Where is the dtmf-relay command configured on Cisco Unified Border Element?

in the voice-class VoIP configuration

in the VoIP dial peer

in global SIP configuration

in the VoIP or POTS dial peers

21. Refer to the exhibit.





Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number “222333444” and Cisco Unified Communications Manager is expecting the called number to be delivered as “444333222”. The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit.

Which action must the administrator take to fix the issue?

Change the destination-pattern on the outgoing dial peer to match “444333222”.

Set up translation-profile on the incoming dial peer to match incoming traffic.

Create specific matching for “222333444” on the incoming dial peer.

Fix the voice translation-rule to match specifically number “222333444” and change it to “444333222”.

22. Refer to the exhibit.





Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit.

Which rule modified DNIS in the format that the provider is expecting?

rule 1 /^/+([^1].*)//0111/

rule 1/^+1([2-9]..[2-9]……$)//1/

rule 1 /^([2-9]..[2-9]……$)//1/

rule 1 /^+1([2-9]..[2-9]……$)///

23. Refer to the exhibit.





An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer’s carrier has a requirement that dialing long distance requires an access code to be entered.

Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)

Enanle PRAC

Enable Early Offer on the Cisco Unified Border Element.

Enable the supplementary-service media-renegotiate command.

Enable Media Flow Around

Enable Mid-Call Signaling Consumption.

24. Which IOS command creates a SIP-enabled dial peer?

voice dial-peer 20 sip

dial-peer voice 20 voip

dial-peer voice 20 pots

dial peer voice 20 sip

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