Implementing Cisco Advanced Call Control and Mobility Services (CLASSP) practice questions

1. Question 1 of 25

   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).

   • AH.245 Terminal Capability Set
   • BH.245 Open Logical Channel
   • CH.225 Connect
   • DH.245 Open Logical Channel Ack

   Show answer

   Correct answer: B

   How others answered

   • D100%

   Most people chose D, not B. Verify this one against the vendor documentation.

2. Question 2 of 25

   Select 2

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

   • ADTMF
   • BBFCP
   • CVIDEO
   • DFAX
   • EAUDIO

   Show answer

   Correct answer: A, D

   How others answered

   • AD33%
   • AC22%
   • CE22%
   • BC11%
   • AE11%

   Only 33% agreed on this answer — treat it as contested rather than settled.

3. Question 3 of 25

   Select 2

   

   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.)

   • APhone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
   • BPhone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
   • CAs 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.
   • DAs 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.
   • EAs 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.

   Show answer

   Correct answer: A, D

   How others answered

   • AD75%
   • AC25%

4. Question 4 of 25

   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?

   • AContact: header of the 200 OK response
   • BAllow: header if the 200 OK response
   • Co= line of SDP content
   • Dc= line of SDP content

   Show answer

   Correct answer: D

   How others answered

   • D100%

5. Question 5 of 25

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

   • AALERTING
   • BPROCEEDING
   • CCONNECT
   • DRINGING

   Show answer

   Correct answer: A

   How others answered

   • A100%

6. Question 6 of 25

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

   • AThe far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
   • BCisco Unified Communications Manager invoked media termination point resources.
   • CThe RTP traffic is arriving beyond the jitter buffer on the receiving end.
   • DA firewall in the media path is blocking TCP ports 16384-32768.

   Show answer

   Correct answer: A

   How others answered

   • A100%

7. Question 7 of 25

   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?

   • Adebug H.323 messages
   • Bdebug H.225 asn1
   • Cdebug H.246 asn 1
   • Ddebug H.225 media
   • Edebug H.323 asn 1

   Show answer

   Correct answer: B

   How others answered

   • B50%
   • D50%

   Only 50% agreed on this answer — treat it as contested rather than settled.

8. Question 8 of 25

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

   • Aincoming uri
   • Btarget carrier-id
   • Canswer-address
   • Dincoming called-number

   Show answer

   Correct answer: A

   How others answered

   • A100%

9. Question 9 of 25

   Which description of RTP timestamps or sequence numbers is true?

   • AThe sequence number is used to detect losses.
   • BTimestamps increase by the time "carrying" by a packet.
   • CSequence numbers increase by four for each RTP packet transmitted.
   • DThe timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).

   Show answer

   Correct answer: D

   How others answered

   • D100%

10. Question 10 of 25

    Select 2

    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.)

    • AGo to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
    • BAsk the firewall administrator to change the ports to TCP.
    • CAsk the firewall administrator to change the range of UDP ports to 16384-32767.
    • DGo to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
    • EGo to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

    Show answer

    Correct answer: C, D

    How others answered

    • CD100%

11. Question 11 of 25

    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?

    • ACallManager traces
    • BCTI Manager traces
    • CCisco IP Manager Assistant
    • DCall logs

    Show answer

    Correct answer: A

    How others answered

    • A100%

12. Question 12 of 25

    

    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?

    • AAllow Passthrough of Configured Line Device Caller Information must be enabled.
    • BAccept Audio Codec Preferences in Received Offer must be set to On.
    • COn the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
    • DEarly Offer for G Clear Calls must be enabled.

    Show answer

    Correct answer: C

    How others answered

    • C100%

13. Question 13 of 25

    Select 2

    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.)

    • Athree-way conference
    • Bsecure SIP lines
    • CT.38 fax relay
    • Dtranscoding
    • ESIP trunk

    Show answer

    Correct answer: A, C

    How others answered

    • AC100%

14. Question 14 of 25

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

    • AConfigure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
    • BConfigure IP Address Trusted Authentication for Incoming VoIP Calls.
    • CConfigure the command no ip address trusted authenticate under "voice service voip".
    • DEnable Secondary Dial tone on Analog and Digital FXO Ports.

    Show answer

    Correct answer: B

    How others answered

    • B100%

15. Question 15 of 25

    You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

    • Aconfiguration for a single SIP phone
    • Bconfiguration items common for all SIP phones
    • Cconfiguration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
    • Dconfiguration for SIP registrar service

    Show answer

    Correct answer: A

    How others answered

    • A100%

16. Question 16 of 25

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

    • Ainterworking between an OOB method and RFC2833 for flow-around calls
    • Binterworking between h245-signal and rtp-nte
    • Cinterworking between an OOB method and RFC2833 for flow-through calls
    • Dinterworking between h245-alpha numeric and sip-kpml

    Show answer

    Correct answer: A

    How others answered

    • A67%
    • C33%

    Only 67% agreed on this answer — treat it as contested rather than settled.

17. Question 17 of 25

    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?

    • Aallow-connections sip to sip
    • Bvoice service voip
    • Cvoice register global
    • Dvoice register dn

    Show answer

    Correct answer: B

    How others answered

    • B58%
    • C42%

    Only 58% agreed on this answer — treat it as contested rather than settled.

18. Question 18 of 25

    

    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?

    • Arule 1 /^/+\([^1].*\)/ /011\1/
    • Brule 1/^\+1\([2-9]..[2-9]"¦"¦$\)/ /\1/
    • Crule 1 /^\([2-9]..[2-9]"¦"¦$\)/ /\1/
    • Drule 1 /^\+1\([2-9]..[2-9]"¦"¦$\)/ /\0/

    Show answer

    Correct answer: B

    How others answered

    • B100%

19. Question 19 of 25

    Select 2

    

    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.)

    • AEnanle PRACK.
    • BEnable Early Offer on the Cisco Unified Border Element.
    • CEnable the supplementary-service media-renegotiate command.
    • DEnable Media Flow Around
    • EEnable Mid-Call Signaling Consumption.

    Show answer

    Correct answer: A, B

    How others answered

    • AB100%

20. Question 20 of 25

    Which IOS command creates a SIP-enabled dial peer?

    • Avoice dial-peer 20 sip
    • Bdial-peer voice 20 voip
    • Cdial-peer voice 20 pots
    • Ddial peer voice 20 sip

    Show answer

    Correct answer: B

    How others answered

    • B100%

21. Question 21 of 25

    

    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?

    • AChange the destination-pattern on the outgoing dial peer to match "444333222".
    • BSet up translation-profile on the incoming dial peer to match incoming traffic.
    • CCreate specific matching for "222333444" on the incoming dial peer.
    • DFix the voice translation-rule to match specifically number "222333444" and change it to "444333222".

    Show answer

    Correct answer: B

    How others answered

    • B50%
    • A25%
    • C13%
    • D13%

    Only 50% agreed on this answer — treat it as contested rather than settled.

22. Question 22 of 25

    Select 2

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

    • A[email protected]
    • B*@example.com
    • Cgateway.corporate.local
    • Dexample.com
    • E*.*

    Show answer

    Correct answer: D, E

    How others answered

    • DE75%
    • D25%

23. Question 23 of 25

    Select 2

    Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)

    • AThe Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.
    • BIf you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
    • CIf you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
    • DThe Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
    • EThe Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.

    Show answer

    Correct answer: A, B

    How others answered

    • AB100%

24. Question 24 of 25

    A user in location X dials an extension at location Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?

    • Amissing Call Admission Control
    • Bcodec mismatch
    • Cptime mismatch
    • Dphone class of service issue

    Show answer

    Correct answer: D

    How others answered

    • A44%
    • D33%
    • C11%
    • B11%

    Most people chose A, not D. Verify this one against the vendor documentation.

25. Question 25 of 25

    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?

    • A813510[012345]
    • B813510[12345]
    • C813510[^0-5]
    • D81XXXXX

    Show answer

    Correct answer: A

    How others answered

    • C50%
    • A50%

    Most people chose C, not A. Verify this one against the vendor documentation.

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300-815 practice, question 1 of 25