Course 2023 300-815 Test Prep Training Practice Exam Download [Q64-Q82] : Free Exam Dumps Collection : http://free.examcollectionpass.com

NO.64 A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)

three-way conference

secure SIP lines

T.38 fax relay

transcoding

SIP trunk

NO.65 How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?

At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.

At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.

At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.

At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.

NO.66 A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)

three-way conference

secure SIP lines

T.38 fax relay

transcoding

SIP trunk

NO.67

Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user
C. Which two scenarios are correct? (Choose two.)

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

NO.68 Refer to the exhibit.

Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized
E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?

rule 1 /^/+([^1].*)/ /0111/

rule 1/^+1([2-9]..[2-9]……$)/ /1/

rule 1 /^([2-9]..[2-9]……$)/ /1/

rule 1 /^+1([2-9]..[2-9]……$)/ //

NO.69 Refer to the exhibit.

A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

Allow Passthrough of Configured Line Device Caller Information must be enabled.

Accept Audio Codec Preferences in Received Offer must be set to On.

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

Early Offer for G Clear Calls must be enabled.

NO.70 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?

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.

NO.71 Which description of RTP timestamps or sequence numbers is true?

The sequence number is used to detect losses.

Timestamps increase by the time “carrying” by a packet.

Sequence numbers increase by four for each RTP packet transmitted.

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout

NO.72 Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?

Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.

Configure IP Address Trusted Authentication for Incoming VoIP Calls.

Configure the command no ip address trusted authenticate under “voice service voip”.

Enable Secondary Dial tone on Analog and Digital FXO Ports.

NO.73

Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

Allow Passthrough of Configured Line Device Caller Information must be enabled.

Accept Audio Codec Preferences in Received Offer must be set to On.

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

Early Offer for G Clear Calls must be enabled.

NO.74 Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally.
What are two possible solutions? (Choose two.)

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Ask the firewall administrator to change the ports to TCP.

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

NO.75 An engineer is configuring a call park feature in Cisco Unified Communications Manager Express. Which command does the engineer use to ensure that the call is reverted to the user after 60 seconds?

R2(config-ephone-dn)#park reservation-group 60

R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002

R2(config-ephone-dn)#park reservation-group 1

R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002

NO.76 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?

midcall-signallng passthru

midcall-signaling preserve-codec

no update-callerid

midcall-signaling passthru media-change

NO.77 An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?

ILS Max Number of Learned Objects in Database

ILS Active Learned Object Upper Limit

Global Data Service Parameter Limit

Imported Dial Plan Replication Database Object Lower Limit

NO.78 Refer to the exhibit.

Within the North American Numbering Plan, gateways located in Ottawa, Canada and marked as “YOW” are assigned to the Calling Party Transformation CSS NANP_CgPTP, which contains partition NANP_calling_xforms. What is the calling-party number and the numbering type if the calling user
+1613-555-1234 dials the number?

calling number 613-555-1234 and numbering type “subscriber”

calling number 011-1-613-555-1234 and numbering type “subscriber”

calling number 011613-555-1234 and numbering type “international”

calling number 613-555-1234 and numbering type “national”

NO.79 Refer to the exhibit.

Calls incoming from the provider are not working through newly set up Cisco Unified Border Element.
Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number “222333444” and Cisco Unified Communications Manager is expecting the called number to be delivered as “444333222”. The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?

Change the destination-pattern on the outgoing dial peer to match “444333222”.

Set up translation-profile on the incoming dial peer to match incoming traffic.

Create specific matching for “222333444” on the incoming dial peer.

Fix the voice translation-rule to match specifically number “222333444” and change it to “444333222”.

NO.80 Which services are needed to successfully implement Cisco Extension Mobility in a standalone Cisco Unified Communications Manager server?

Cisco Extended Functions, Cisco Extension Mobility, and Cisco AXL Web Service

Cisco CallManager, Cisco TFTP, and Cisco CallManager SNMP Service

Cisco CallManager, Cisco TFTP, and Cisco Extension Mobility

Cisco TAPS Service, Cisco TFTP, and Cisco Extension Mobility

NO.81 A user in location X dials an extension at location Y.
The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?

missing Call Admission Control

codec mismatch

ptime mismatch

phone class of service issue

NO.82 After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323

Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad

Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod

Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily