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300-815 Dumps

300-815 Real Exam Dumps Questions and answers 11-20

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Question No.12

Which description of RTP timestamps or sequence numbers is true?

  1. The sequence number is used to detect losses.

  2. Timestamps increase by the time quot;carryingquot; by a packet.

  3. Sequence numbers increase by four for each RTP packet transmitted.

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

Answer: D

Section: Signaling and Media Protocols

Explanation/Reference:

Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html

Question No.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?

  1. CallManager traces

  2. CTI Manager traces

  3. Cisco IP Manager Assistant

  4. Call logs

Answer: A

Section: Signaling and Media Protocols

Question No.14

image

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?

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

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

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

  4. Early Offer for G Clear Calls must be enabled.

Answer: C

Section: Signaling and Media Protocols

Question No.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.)

  1. three-way conference

  2. secure SIP lines

  3. T.38 fax relay

  4. transcoding

  5. SIP trunk

Answer: AC

Section: CME/SRST Gateway Technologies

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/srst_sip_isr4000.html

Question No.16

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

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

  2. Configure IP Address Trusted Authentication for Incoming VoIP Calls. under quot;voice service voipquot;.

  3. Configure the command no ip address trusted authenticate

  4. Enable Secondary Dial tone on Analog and Digital FXO Ports.

Answer: B

Section: CME/SRST Gateway Technologies

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmead m/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2

Question No.17

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

  1. configuration for a single SIP phone

  2. configuration items common for all SIP phones

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

  4. configuration for SIP registrar service

Answer: C

Section: CME/SRST Gateway Technologies

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/srst_setting_up_using_sip.html

Question No.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?

  1. allow-connections sip to sip

  2. voice service voip

  3. voice register global

  4. voice register dn

Answer: C

Section: CME/SRST Gateway Technologies

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/99946-cme-sip-guide.html

Question No.19

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

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

  2. interworking between h245-signal and rtp-nte

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

  4. interworking between h245-alpha numeric and sip-kpml

Answer: A

Section: Cisco Unified Border Element Explanation

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35

Question No.20

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

  1. in the voice-class VoIP configuration

  2. in the VoIP dial peer

  3. in global SIP configuration

  4. in the VoIP or POTS dial peers

Answer: B

Section: Cisco Unified Border Element Explanation

Explanation/Reference:

Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube- book/dtmf-relay.html

Question No.21

image

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 quot;222333444quot; and Cisco Unified Communications Manager is expecting the called number to be delivered as quot;444333222quot;. 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?

  1. Change the destination-pattern on the outgoing dial peer to match quot;444333222quot;.

  2. Set up translation-profile on the incoming dial peer to match incoming traffic. Create specific matching for quot;222333444quot; on the incoming dial peer.

  3. rule to match specifically number quot;222333444quot; and change it to quot;444333222quot;.

  4. Fix the voice translation-

Answer: B

Section: Cisco Unified Border Element

Explanation/Reference:

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