Cisco 300-815 Practice Test - Questions Answers, Page 15
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What are two functions of the metadata in SIP recording? (Choose two.)
includes the details of the application media stream
contains the cryptographic context of the SRTP passthrough calls
identifies the session and media association time
carries the Agent Statistics Report
carries the communication session data (audio and video calls) that describes the call to the recording server
An engineer is configuring a Cisco UCM solution. The requirements state that some users in one location will receive calls from a number during work hours 09AM to 5PM, and another group will get the calls from the same number outside this defined timeslot. Users will also change the outgoing number when reaching out to customers based on the same time-of-day routing rules. Which feature is needed to allow for this type of configuration?
partitions
route groups
route lists
CSS
A company has users that frequently visit locations outside the office. These users need to be able to make calls and keep the company integrity by presenting a company caller ID. The users are equipped with company-paid mobile phones. The company uses Cisco UCM with Cisco Unified Border Element and Expressway-C and Expressway-E for external communications. Which technology must be configured to meet this requirement?
Cisco Unified SNR
Cisco Unified Mobility
Cisco FindMe
Cisco Roaming Mobility
Refer to the exhibit.
A collaboration engineer investigates an issue where the user calls that match the local Jabber domain are routed back to Cisco UCM. Which action resolves this problem?
Check the check box to block the pattern for the local domain.
Create another route SIP pattern for the local domain with the block pattern selected.
Create a translation pattern to block calls to the local domain.
Create a transformation pattern to block calls to the local domain.
A newly deployed site in Pennsylvania uses SIP-to-SIP legs on a Cisco Unified Border Element. The administrator wants to match the incoming dial peer based on the calling number. Which configuration is needed in the dial-peer?
answer-address
incoming called-number
destination calling
destination-pattern
An engineer configures several calling party transformation patterns in Cisco UCM so that calling numbers are shown in the +E.164 format. The route patterns match seven-digit internal numbers and prefix the numbers with 1408. The SIP trunk matches 1408XXXXXXX and prefixes the number with +. When users make outbound calls, the calling number still shows as seven digits. What is the cause of this problem?
The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied.
Calling transformations cannot be configured on both route patterns and a SIP trunk.
The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.
The engineer must also apply a calling transformation at the route group details level.
A company is using Cisco Jabber on-premises to make B2B calls on video. The calls are using Cisco Expressway-C and Expressway-E and have been configured in Cisco UCM to be able to call any URI on the internet. The Jabber client also has voice enabled and must be able to call local, regional, and international numbers. Where must Cisco UCM be configured to meet this requirement for URI dialing?
Enter '!#' in the route pattern section tied to a route group and list.
Enter '*' in the SIP route pattern.
Enter '!#' in the SIP route pattern.
Enter '.*' in the route pattern section tied to a route group and list.
Refer to the exhibit.
Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?
request Invite sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the SIP translation profile configuration
sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the voice translation profile configuration
request Invite sip-header Diversion modify '<sip:1(...)@' '<sip:9135551\1@' under the SIP profile configuration
request Invite sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the voice translation profile configuration
Refer to the exhibit.
Refer to the exhibit. A company deploys an additional instance of Cisco UCME named CUCME 2 in a branch office. Phones in the main office are register of Unified CME named CUCME 1. Phones in the branch office are registered with their extension to CUCME 2. The engineer needs to configure CUCME the branch office using an 8-kbps codec. Which configuration should be applied to CUCME 1?
A)
B)
C)
D)
Option A
Option B
Option C
Option D
What does VoIP trace in a Cisco Unified Border Element do?
It stores all call trace data in system memory for failed calls.
It stores all can trace data in system memory (or successful and failed calls
It stores all call trace data in system log files for successful and failed calls.
It stores all can trace data in system file logs for failed calls.
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