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Cisco 300-815 Practice testing questions from Youtube

latest updated Cisco 300-815 exam questions and answers

QUESTION 1
What is the first preference condition matched in a SIP-enabled incoming dial peer?
A. incoming uri
B. target carrier-id
C. answer-address
D. incoming called-number
Correct Answer: A
Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-DepthExplanation-of-Cisco-IOS-and-IO.html#anc8

 

QUESTION 2
What is the relationship between partition, time schedule, and the time period in Time-of-Day routing in Cisco Unified
Communications Manager?
A. A partition can have multiple time schedules assigned. A time schedule contains one or more time periods.
B. A partition can have a one-time schedule assigned. A time schedule contains one or more time periods.
C. A partition can have multiple time schedules assigned. A time schedule contains only one time period.
D. A partition can have a one-time schedule assigned. A time schedule contains only one time period.
Correct Answer: A

 

QUESTION 3
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call
through the PSTN? (Choose two.)
A. AAR destination mask
B. called party transform mask
C. external phone number mask
D. +E.164 alternate number mask
E. enterprise alternate number mask
Correct Answer: AC
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html

QUESTION 4
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?
A. missing Call Admission Control
B. codec mismatch
C. ptime mismatch
D. phone class of service issue
Correct Answer: B

 

QUESTION 5
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slowstart mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol
traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port
information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
A. H.245 Terminal Capability Set
B. H.245 Open Logical Channel
C. H.225 Connect
D. H.245 Open Logical Channel Ack
Correct Answer: B
Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html

 

QUESTION 6
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration.
Which configuration is occurring in this section?
A. configuration for a single SIP phone
B. configuration items common for all SIP phones
C. configuration for a pool of SIP phones (similar to devise pool on Cisco Unified Communications Manager)
D. configuration for SIP registrar service
Correct Answer: C
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/srst_setting_up_using_sip.html

 

QUESTION 7[2021.2] lead4pass 300-815 exam questions q7

Refer to the exhibit. Users report that when they dial Cisco Unity Connection from an external network, they cannot
enter any digits. Assuming only in-band DTMF is supported, what is the reason for this malfunction?
A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
D. No DTMF is negotiated.
Correct Answer: D

 

QUESTION 8[2021.2] lead4pass 300-815 exam questions q8

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?
A. Allow Passthrough of Configured Line Device Caller Information must be enabled.
B. Accept Audio Codec Preferences in Received Offer must be set to On.
C. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx
Messages.
D. Early Offer for G Clear Calls must be enabled.
Correct Answer: C

 

QUESTION 9[2021.2] lead4pass 300-815 exam questions q9

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?
A. rule 1 /^/+\([^1].*\)/ /011\1/
B. rule 1/^\+1\([2-9]..[2-9]……$\)/ /\1/
C. rule 1 /^\([2-9]..[2-9]……$\)/ /\1/
D. rule 1 /^\+1\([2-9]..[2-9]……$\)/ /\0/
Correct Answer: B

 

QUESTION 10
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call
setup, which debug must the Administrator turn on?
A. debug H.323 messages
B. debug H.225 asn1
C. debug H.246 asn 1
D. debug H.225 media
E. debug H.323 asn 1
Correct Answer: B

 

QUESTION 11
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
A. route group
B. line group
C. hunt list
D. hunt pilot
Correct Answer: B
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/cucm_b_systemconfiguration-guide-1201/cucm_b_system-configuration-guide-1201_chapter_010101.html

 

QUESTION 12
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration?
(Choose two.)
A. 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.
B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
D. 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.
E. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
Correct Answer: AB
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F
_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-servicesguide100_chapter_010000.pdf

 

QUESTION 13
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real-time?
A. Analysis Manager > Inventory > Trace File Repositories
B. System > Tools > Trace and Log Central
C. Voice/Video > Session Trace Log View > Real Time Data
D. Voice/Video > Session Trace Log View > Open From Local Disk
Correct Answer: C
Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-managercallmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html

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