300-815 Practice Test

300-815 Practice Test practice questions, page 6 of 10, with answer review and explanations.

Question 26

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

  1. example.com@gateway.corporate.local
  2. *@example.com
  3. gateway.corporate.local
  4. example.com
  5. *.*

Question 27

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

  1. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.
  2. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
  3. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
  4. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
  5. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.

Question 28

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?

  1. 813510[012345]
  2. 813510[12345]
  3. 813510[^0-5]
  4. 81XXXXX

Question 29

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

  1. can be associated under the route group
  2. can be associated only under the route list
  3. chooses the route group that is configured under the device pool of the calling-party device
  4. chooses the route group that is configured under the device pool of the called-party device
  5. can be assigned directly to the route pattern

Question 30

Refer to the exhibit. An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?

  1. answer-address 555 "¦"¦..
  2. codec g729
  3. session-protocol sipv2
  4. incoming called number 555"¦"¦.