Cisco 300-815 Online Practice Questions and Exam Preparation

Cisco 300-815 Online Questions & Answers

• Question 191:

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

  A. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
  B. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
  C. Configure the command no ip address trusted authenticate under "voice service voip".
  D. Enable Secondary Dial tone on Analog and Digital FXO Ports.
  B. Configure IP Address Trusted Authentication for Incoming VoIP Calls.

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

• Question 192:

  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
  C. voice service voip sip bind all source-interface GigabitEthernet0/0

• Question 193:

  If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?

  A. 12!
  B. 12X (urgent priority set)
  C. 1XX (urgent Priority Set)
  D. 12[2-5]
  D. 12[2-5]

• Question 194:

  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
  D. midcall-signaling passthru media-change

• Question 195:

  When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?

  A. dtmf-relay h245-signal
  B. dtmf-relay rtp-nte
  C. dtmf-relay cisco-rtp
  D. dtmf-relay sip-kpml
  D. dtmf-relay sip-kpml

• Question 196:

  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?

  A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
  B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
  C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
  D. No DTMF is negotiated.
  D. No DTMF is negotiated.

• Question 197:

  Refer to the exhibit.

  

  

  A collaboration engineer is troubleshooting an issue where the PSTN calls of a Cisco UCM IP phone user are not reaching the PSTN gateway. Which action resolves the issue?

  A. Change the calling search space of the user's line or device.
  B. Change the "Call Classification" to "OnNet" on the route pattern.
  C. Ensure that the user's phone is assigned to a device pool with the correct local route settings.
  D. Deselect "Block this pattern" on the route pattern.
  C. Ensure that the user's phone is assigned to a device pool with the correct local route settings.

• Question 198:

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

  A. in the voice-class VoIP configuration
  B. in the VoIP dial peer
  C. in global SIP configuration
  D. in the VoIP or POTS dial peers
  B. in the VoIP dial peer

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

• Question 199:

  A user requests a feature to send an active call to the mobile phone number on the physical phone. As an administrator, what should be configured in the Cisco UCM to accomplish this?

  A. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.
  B. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.
  C. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template
  D. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template.
  B. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.

• Question 200:

  Due to a shortage of physical interfaces on a device the administrator requires that a loopback for RTP is used. Which command is required when using a loopback interface for RTP?

  A. voice-class sip resources priority mode passthrough
  B. voice-class sip bind control source-interface Loopback0
  C. voice-class sip early-offer forced.
  D. voice-class sip bind media source-interface Loopback0
  D. voice-class sip bind media source-interface Loopback0