Refer to the exhibit. An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?

bind signaling and media to the loopback interface

use of port 5061 for SIP secure

access list allowing the firewall IP

ip prefix-list to filter the unwanted IP address

bind signaling and media to the loopback interface

What are two configureation features of the Client matter code setting in the route pattern configuration? (Choose two.)

Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.

The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.

The client Matter Code feature supports overlap sending since the Cisco UCM can determine when to prompt the user for the code.

Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.

The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.

Selecting the Allow Overlap Sending setting allows a user to select the Require Client Matter Code setting.

An administrator sees the voice register pool 1 command in a Cisco UCME configuration. Which configuration is occurring in this section?

configuration for a single SIP phone

configuration for a pool of SIP phones (similar to device pool on Cisco UCM)

configuration for a single SIP phone

configuration for SIP registrar service

configuration items common for all SIP phones

An engineer has two cisco UCM Clusters and wants to integrate them using ILS with TLS certificates. Cluster A (pub and 1 subscriber) will be the hub, and Cluster B (pub and 1 subscriber) will be the spoke. Both Clusters have self-signed certificates. The engineer has exchanged Publisher A and subscriber B Tomcat certificates, but the connection fails. What is the cause of the failure?

The tomcat certificate from Cluster B must be the publisher.

The engineer needs to exchange the CallManager certificate.

Refer to the exhibit. Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system. Each building has is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?

Add a local route group on the device pool configuration.

Change the partition of the route pattern

Add a sip trunk inside route group Standard Local Route Group.

Modify the route pattern to add a prefix of 91

Add a local route group on the device pool configuration.

ABC company has decided to implement hunt groups to help distribute calls between members. In order to implement this, administrator must configure hunt list, hunt group, and line groups on Cisco UCM. Which distribution algorithms should the administrator implement?

Top Down, Round Robin, Broadcast

Top Down, Round Robin, Distribute

Sequantial, Circular, Broadcast

Signal number reach call phone that not answered are leaving voicemails on the cell phone rather the corporate mailbox. Which two options will resolve this issue? (Choose two.)

Check the Enable Unified Mobility features checkbox

Decrease the T301 timer Decrease the Answer Too Late timer

Check the Enable Extend and Connect checkbox

Check the Enable Unified Mobility features checkbox

Decrease the T301 timer Decrease the Answer Too Late timer

A customer routes PSTN calls to ITSP through a SIP trunk on Cisco UCM that forwards and receives calls to and from ITSP. ITSP is set to send an E.164 number when the customer's extension is four digits. Which action should be taken to route the incoming calls to four-digit extensions?

Set the Significant Digits to 4 on the SIP trunk.

Configure a voice translation rule to map the E.164 number to four digits and assign it to the incoming dial-peer on Cisco Unified Border Element.

Set the Significant Digits to 4 on the SIP trunk.

Configure a voice translation profile to map the E.164 number to four digits and assign it to the incoming dial-peer on Cisco Unified Border Element.

Set the Significant Digits to 8 on the SIP trunk.

Refer to the exhibit. An administrator has configured a SIP trunk between two Cisco UCM clusters. For calls that should use the trunk, the calls fail with a fast busy. The administrator checks the Cisco CallManager SDL traces and found that the cluster to which the calling device is registered never sends an INVITE to the destination cluster. The administrator also verifies that all nodes from both clusters are powered on, and the CallManager service is running. How is this issue resolved?

The administrator must allow connectivity so that TCP connections do not fail between the nodes.

The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.

The administrator must allow connectivity so that TCP connections do not fail between the nodes.

The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.

The administrator must associate the route pattern with a calling search space the device can dial.

Refer to the exhibit. For long-distance calls, users must prefix their dialed number with ''91.'' The translation pattern was created to strip the 91 as the PSTN expects a 10- digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?

Disable Use Calling Party's External Phone Number Mask on the route pattern.

Change the partition of the translation pattern from none to pstn_pt.

Enable Force Authorization Code on the route pattern.

Disable Use Calling Party's External Phone Number Mask on the route pattern.

Enable Use Calling Party's External Phone Number Mask on the translation pattern.

The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?

Change Session Refresh Method on the SIP profile to INVITE.

Enable SIP ReMXX Options on the SIP profile.

Enable Send send-receive SDP in mid-call INVITE on the SIP profile.

Change Session Refresh Method on the SIP profile to INVITE.

Increase Retry INVITE to 20 seconds on the SIP profile.

A user's phone is already configured for Single Number Reach, and the user wants a feature to move an active call from a mobile phone to a desk phone and vice-versa. As an administrator, which additional configuration should be made to fulfill the user's request?

Add the mobility key to the softkey template that the desk phone is using.

Confirm that the desk phone is subscribed to Cisco Extension Mobility.

Check to make sure that the Resume softkey option appears on the desk phone.

Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.

Add the mobility key to the softkey template that the desk phone is using.

Cisco UCM has 100,000 entries in the database learned through the ILS Service. Parameter ILS Max Number of Learned Objects in Database value is set to 100,000. What will happen to learned data when the service parameter value is reduced to 50,000?

Cisco UCM does not write additional ILS learned objects to the database and keeps the existing database entries.

Cisco UCM does not write additional ILS learned objects to the database and will delete the first 50,000 entries learned to keep it to the service parameter value.

Cisco UCM does not write additional ILS learned objects to the database and will delete the last 50,000 entries learned to keep it to the service parameter value.

Cisco UCM does not write additional ILS learned objects to the database and keeps the existing database entries.

Cisco UCM will overwrite an entry for newly learned data and keep the parameter value at 100,000.

Refer to the exhibit. ILS has been configured between two hubs using this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?

The Cluster IDs have not been set to unique values

A password has never been set for ILS.

Use TLS Certificates must be selected.

Trust certificates for ILS have not been installed on the clusters

The Cluster IDs have not been set to unique values

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 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.

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.

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.

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

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.

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?

Add a media resource to transcode between available capabilities.

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 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 the dial peer of the gateway has the correct destination pattern configured.

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

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 rehxx require 100rel

voice service voip sip send 180 sdp

voice service voip sip rehxx require 100rel

voice service voip sip no reMxx

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?

Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.

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.

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 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

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

A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site. Which two actions will fulfill the requirement? (Choose two.)

Create one route group for each site and one global route list for PSTN calls that point to the local route group.

Assign one route group as a local route group in the device pool of the corresponding site.

Create one route group for each site and one global route list for PSTN calls that point to the local route group.

Create a route group which has all the gateways and associate it to the device pool of every site.

Create one global route list for PSTN calls that points to one global PSTN route group.

Create a hunt group and assign it to each side route pattern

Assign one route group as a local route group in the device pool of the corresponding site.

Refer to the exhibit. A company needs to ensure that all calls are normalized to E164 format. Which configuration will ensure that the resulting digit string 14085554001 is created and will be routed to the E.164 routing schema?

Called Party Transformation Mask of + 14085554XXX

Called Party Transformation Mask of 1408555[35)XXX

Calling Party Transformation Mask of +1408555XXXX

Calling Party Transformation Mask of +14085554XXX

An administrator is asked to configure egress call routing by applying globalization and localization on Cisco UCM. How should this be accomplished?

Globalize the calling and called numbers to E. 164 format and localize the called number in the gateway.

Localize the calling and called numbers to PSTN format and globalize the calling and called numbers in the gateway.

Globalize the calling and called numbers to PSTN format and localize the calling number in the gateway.

Localize the calling and called numbers to E. 164 format and globalize the called number in the gateway.

Globalize the calling and called numbers to E. 164 format and localize the called number in the gateway.

Refer to the exhibit. A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?

Enable *Early Offer support for voice and video calls' on the SIP Profile Configuration Page in Cisco UCM.

Enable 'SIP ReI1XX Options* and -Early Offer Support' on the SIP Profile Configuration Page in Cisco UCM.

Enable *Early Offer support for voice and video calls' on the SIP Profile Configuration Page in Cisco UCM.

Disable 'SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.

Disable 'Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.

An administrator is trying to apply configuration changes on Cisco CME. When the users registered on Cisco CME to dial a local number to a PSTN call, the Cisco CME sends an incorrect number of digits. What translation rule fixes the issue and sends the correct number of digits?

voice translation-rule 1 rule 1 /^4...V /2404\0/ type any subscriber plan any isdn

voice translation-rule 1 rule 1 /^4...$/2404\0/ type any national plan any Isdn

voice translation-rule 1 rule 1 // // type any subscriber plan any isdn

voice translation-rule 1 rule 1 /^4...S/ /9132404 0/ type any subscriber plan any Isdn

voice translation-rule 1 rule 1 /^4...V /2404\0/ type any subscriber plan any isdn

Refer to the exhibit. A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?

The destination pattern is incorrect for the dialed number.

No codecs are configured on the dial peers

The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.

The destination pattern is incorrect for the dialed number.

VAD was not disabled on the outgoing dial poor.

Cisco 300-815 Practice Test 3 - Free Online Exam Questions 81-120

Refer to the exhibit.

Cisco 300-815 image Question 81 114985 10102024232831000000

While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provider, but there is no response The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?

Become a Premium Member for full access

Unlock Premium Member

Cisco 300-815 Practice Test 3

Question

Cisco 300-815 Practice Tests

Practice Tests