Cisco 300-815 Practice Test - Questions Answers, Page 11
List of questions
Related questions
Refer to the exhibits.
Regions have been configured for all major branches based on the available circuit bandwidth. Some calls from Region A endpoints to Region B endpoints are failing to connect. How is this issue resolved?
Update the calling search space for affected endpoints to none.
Add a media resource to transcode between available capabilities.
Update all regions to 8 kbps maximum audio bitrate.
Increase the number of available media termination points.
A new deployment is using MVA for a specific user on the sales team, but the user is having issues when dialing DTMF. Which DTMF method must be configured in resolve the issue?
gateway
out-of-band
channel
in-band
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?
Verify that IP routing is correct between the gateway and the IP phone.
Verify that the route pattern is not blocking calls to the destination number.
Verify that the dial peer of the gateway has the correct destination pattern configured.
Verify that the route pattern has the correct calling-party transformation mask
An engineer is configuring Cisco UCM lo forward parked calls back to the user who parked the call if it is not retrieved after a specified time interval. Which action must be taken to accomplish this task?
Configure device pools.
Configure service parameters
Configure enterprise softkeys.
Configure class of control.
Refer to the exhibit.
An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?
voice service voip sip send 180 sdp
voice service voip sip rehxx require 100rel
sip-ua disable-early-media 180
voice service voip sip no reMxx
DRAG DROP
Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.
Users are reporting that several inter-site calls are failing, and the message 'not enough bandwidth' is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites. How is the issue solved without increasing bandwidth utilization on the WAN links?
Disable Call Admission Control and let the calls use the amount of bandwidth they require.
Configure Call Queuing so that the user waits until there is bandwidth available
Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
Reroute all calls through the PSTN and avoid using WAN.
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next. Which distribution algorithm must be configured to accomplish this task?
top down
circular
broadcast
longest idle time
An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?
Configure Single Number Reach.
Configure an access list.
Configure a mobility identity.
Configure Mobile Voice Access.
Refer to the exhibit.
An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?
SIP binding in SIP configuration mode: voice service voip sip bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0
SIP binding In SIP configuration mode: voice service volp sip bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1
SIP binding In dial-peer configuration mode: dial-peer voice 300 voip voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1
SIP binding in dial-peer configuration mode: dial-peer voice 100 volp voice-class sip bind control source-interface GigabitEthernetO/0/0 voice-class sip bind media source-interface GigabitEthernetO/0/0
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