[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 71-80

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CCIE Collaboration (v1.1)

Question No: 71 – (Topic 1)

Which statement about what happens to a hunt member who does not answer queuing enabled hunt list call in Cisco Unified Communications Manger 9.1 is true?

  1. The hunt member remains logged in if Automatically Logout Hunt Member on No Answer is not selected in the Hunt Pilot configuration page.

  2. The hunt member is logged off if Automatically Logout Hunt Member on No Answer is selected on the Line Group Configuration page.

  3. The hunt member is logged off automatically and must manually reset the phone to log back in.

  4. The hunt member remains logged in if Automatically Logout Hunt Member on No Answer is not selected in Cisco Unified Communications Manger Service Parameters.

  5. The hunt member is logged off automatically and guest press HLOG to log back in.

Answer: B

Question No: 72 – (Topic 1)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A Cisco Unified CME administrator is configuring SNR for a line and has these requirements:

-The remote phone should receive the call after the local phones ring for 10 seconds.

-The ANI displayed on the remote phones should be the local extension number. Which two configuration commands complete these requirements? (Choose two.)

  1. snr 92875421 delay 15 timeout 10

  2. snr 92875421 delay 10 timeout 20

  3. snr calling-number local

  4. snr calling-number remote

  5. snr answer-too-soon 10

Answer: B,C

Question No: 73 – (Topic 1)

A collaboration engineer is troubleshooting an MOH problem on a Cisco IOS SIP gateway. While searching through a debug ccsip message output, which three parameters in the SIP messages can be used to determine if the call was placed on hold? (Choose three)

  1. OPTIONS WITH 301 CALLHOLD

  2. INVITE WITH a=INACTIVE

  3. INVITE WITH a=SENDONLY

  4. OPTION WITH c=INACTIVE

E. c=IN IP4 0.0.0.0

  1. BYE WITH A = CALLHOLD

    Answer: B,C,E Explanation:

    Dumps4Cert 2017 PDF and VCE

    When a call is on hold you get something like this: v=0

    o=CiscoSystemsCCM-SIP 919861 2 IN IP4 10.10.30.14

    s=SIP Call

    c=IN IP4 0.0.0.0

    t=0 0

    m=audio 30120 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20

    a=inactive

    a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

    The connection parameter shows 0.0.0.0, When the call is taking off hold you, the connection parameter should indicate the ip address where media is sent to. So it will have a real value. ( This is usual sent in the ACK.) cucm still sends a DO in the re-INVITE and the far end sends a 200 Ok with SDP. CUCM then sends ACK with SDP

    This is how the whole call hold/transfer/call resume works

    1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path

    2. CUCM sends a Delayed offer to insert MOH or to resume Media stream

      NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK

    3. CUCM establishes newcall leg with the intended transfered destination..Once this call is connected

    4. CUCM receives a new Transfer instruction from the transferring phone to connect the held party

    5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)

    6. Next CUCM sends an inactive SDP to indicate a break in media path between transfering party amp; transfered party to remove Xferring party from call

    7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call

      Question No: 74 – (Topic 1)

      Refer to the exhibit.

      Dumps4Cert 2017 PDF and VCE

      A CUBE Cluster is working in HSRP box-to-box failover model. When the phone A calls Cisco WebEx meeting server to start a conference session, no DTMF tones are recognized. Which configuration change will fix this problem when configured on both CUBEs?

      1. Voice-class sip asymmetric payload dtmf in dial-peer configuration

      2. Dtmf-relay rtp-nte digitdrop in the dial-peer configuration

      3. Media flow-around under voice service voip configuration

      4. Modem relay nse payload-type101 underglobal sip configuration

      5. Asymmetric payload full configured under global sip configuration

Answer: E

Question No: 75 – (Topic 1)

Where can a Cisco Unified CM administrator define Call Detail Records data collection

interval?

  1. Cisco Unified CM Administration Service Parameters

  2. Cisco Unified CM Administration Enterprise Parameters

  3. Cisco Unified Serviceability

  4. Cisco Unified Reporting

  5. Call Detail Records data collection internal is not a configurable parameter.

Answer: B

Question No: 76 – (Topic 1)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A cisco collaboration engineer is troubleshooting a gateway and gatekeeper problem and sees this output from a debug command. Which two configuration can cause this problem? (Choose two)

  1. The same zone prefix is configured in two different gatekeepers

  2. The same H323-ID is configured in two different gateways

  3. The same gw-type-prefix is configured in two different zone subnets IDs

  4. The same zone subnet ID is configured in two different gatekeepers

  5. The same E164-ID is configured in two different gateways

Answer: B,E

Explanation: This output from the debug h225 asn1 command shows a registration reject reason of duplicateAlias.

RAS INCOMING PDU ::=

value RasMessage ::= registrationReject :

{

requestSeqNum 24

protocolIdentifier { 0 0 8 2250 0 3 } rejectReason duplicateAlias:

{

}

gatekeeperIdentifier {quot;gkquot;}

}

This is usually the result of the gateway registering a duplicate of an E164-ID or H323-ID: Another gateway has already been registered to the gatekeeper. If it is a duplicated E164- ID, change the destination pattern configured under a POTS dial-peer associated with an FXS port. If it is a duplicated H323-ID, change the gateway#39;s H.323 ID under the H.323 VoIP interface.

http://www.cisco.com/c/en/us/support/docs/voice/h323/22378-gk-reg-issues.html#rr1

Question No: 77 – (Topic 1)

Refer to the Exhibit.

Dumps4Cert 2017 PDF and VCE

An Engineer is troubleshooting transcoding issue in a remote branch office after a WAN outage. All IP phones can register to a CME in SRST 2900 ISR Router however, the users reported that calls disconnect after pressing the answer soft key. Which three configuration are necessary for successful media resource failover? (Choose three)

  1. SCCP ccm group 1 Associate ccm 3 priority 1

  2. SCCP ccm 10.1.1.2 identifier 3 version 7.0

  3. Telephony-service Sdspfarm units 1

    Sdspfarm tag 1 XD-REMOTE

  4. Sccp ccm 10.1.1.2 identifier 1 version 7.0

  5. Associate profile 3 register XD-REMOTE2

  6. Sccp ccm group 1 Associate ccm 3 priority 3

Answer: B,C,F

Question No: 78 – (Topic 1)

An engineer is configuring QoS for a 100 Mb WAN link. An ISP SLA was signed to support 70% of the link. Which QoS command allows the engineer to use 70% of the link while maintaining a steady flow?

A. traffic-shape rate 100000000 70000000 70000000

  1. police cir 70000000 confirm-action transmit exceed-action drop

  2. police 70000000 13125000 confirm-action transmit exceed-action drop D. traffic-shape rate 70000000 8750000 8750000

Answer: D

Question No: 79 – (Topic 1)

Which of the two responses from a SIP device, which is the only remote destination on a Cisco Unified

Communications Manager SIP trunk with OPTIONS ping enabled cause the trunk to be marked as quot;Out Of Servicequot;? (Choose two.)

  1. 484 Address Incomplete

  2. 408 Request Timeout

  3. 503 Service Unavailable

  4. 504 Server Timeout

  5. 404 Not Found

  6. 505 Version Not Supported

Answer: B,C

Question No: 80 – (Topic 1)

Which method does a Cisco unified 9971 Phone use to send keep-alive messages to Cisco unified

Communications manager?

  1. Sccp StationServerReq

  2. SIP Notify with events set to keep-alive

  3. SIP register with expires set to zero

  4. Sccp stationregister

  5. SIP options

Answer: C

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