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300-815 PDF Questions and Testing Engine With 120 Questions
Exam Details and Topics
The Cisco 300-815 exam is 90 minutes long and contains around 60 multiple-choice questions to cover. It is required to earn at least 825 points of the overall score to get the certification. The test is available in English only, and the exam voucher will cost you $300. It is administered by the Pearson VUE testing company, which has authorized training centers all over the globe.
NEW QUESTION 44
An engineer is troubleshooting Cisco Device Mobility and find that the phone has roamed to a building that is assigned to a different device pool but has not changed its device pool accordingly What action resolves the issue?
- A. Set correct Location under Current Device Mobility Settings
- B. Set the correct subnet under Device Mobility Info.
- C. Enable SRST under Current Device Mobility Settings
- D. Set Device CSS under Current Device Mobility Settings.
Answer: B
NEW QUESTION 45
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 12[2-5]
- B. 12X (urgent priority set)
- C. 1XX (urgent Priority Set)
- D. 12!
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 46
What is a component of Cisco Unified Mobility?
- A. Mobile Connect
- B. Single Number Connect
- C. Smart Client Support
- D. Unified IVR
Answer: A
NEW QUESTION 47
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?
- A. Imported Dial Plan Replication Database Object Lower Limit
- B. Global Data Service Parameter Limit
- C. ILS Max Number of Learned Objects in Database
- D. ILS Active Learned Object Upper Limit
Answer: C
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_5_1SU1/systemConfig/ cucm_b_system-configuration-guide-1251su1/cucm_b_system-configuration-guide- 1251su1_restructured_chapter_0100011.html#CUCM_TK_I7C708C2_00
NEW QUESTION 48
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?
- A. R2(config-ephone-dn)#park reservation-group 1
- B. R2(config-ephone-dn)#park reservation-group 60
- C. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
- D. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
Answer: D
NEW QUESTION 49
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
- A. BFCP
- B. AUDIO
- C. FAX
- D. DTMF
- E. VIDEO
Answer: A,D
NEW QUESTION 50
An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?
- A. Router(config-sip-ua)# no disable-early-media 180
- B. Router(conf-voi-serv)# dlisable-early-media 180
- C. Router(con(-voi-serv)# no disable-early-media 180
- D. Router(conftg-sip-ua)# disable-early-media 180
Answer: D
NEW QUESTION 51
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start 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 Open Logical Channel
- B. H.225 Connect
- C. H.245 Terminal Capability Set
- D. H.245 Open Logical Channel Ack
Answer: A
Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
NEW QUESTION 52
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\([2-9]..[2-9]......$\)/ /\0/
- B. rule 1/^\+1\([2-9]..[2-9]......$\)/ /\1/
- C. rule 1 /^\([2-9]..[2-9]......$\)/ /\1/
- D. rule 1 /^/+\([^1].*\)/ /011\1/
Answer: B
NEW QUESTION 53
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?
- A. Verify that the route pattern is not blocking calls to the destination number.
- B. Verify that the dial peer of the gateway has the correct destination pattern configured.
- C. Verify that the route pattern has the correct calling-party transformation mask
- D. Verify that IP routing is correct between the gateway and the IP phone.
Answer: B
NEW QUESTION 54
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. ptime mismatch
- B. phone class of service issue
- C. missing Call Admission Control
- D. codec mismatch
Answer: C
NEW QUESTION 55
Refer to the exhibit.
An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu. Which two steps must be performed to resolve this issue? (Choose two.)
- A. Activate the Cisco Extended Functions service.
- B. Activate the Cisco Dialed Number Analyzer Server service.
- C. Activate the Cisco CallManager service.
- D. Restart the subscriber
- E. Activate the Cisco Dialed Number Analyzer service.
Answer: B,E
NEW QUESTION 56
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. BLF Call Pickup
- B. Other Group Pickup
- C. Group Call Pickup
- D. Directed Call Pickup
Answer: B
NEW QUESTION 57
Refer to the exhibit.
Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?
- A. Adjust the service parameter T204 timer to the desired value.
- B. Adjust the service parameter T302 timet to the desired value.
- C. Check the Urgent Priority check box under 9.911 pattern.
- D. Point the emergency pattern directly to the PSTN gateway.
Answer: C
NEW QUESTION 58
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. chooses the route group that is configured under the device pool of the called-party device
- B. chooses the route group that is configured under the device pool of the calling-party device
- C. can be assigned directly to the route pattern
- D. can be associated under the route group
- E. can be associated only under the route list
Answer: B,E
NEW QUESTION 59
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. Timestamps increase by the time "carrying" by a packet.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- D. The sequence number is used to detect losses.
Answer: C
NEW QUESTION 60
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. A firewall in the media path is blocking TCP ports 16384-32768.
- B. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- C. Cisco Unified Communications Manager invoked media termination point resources.
- D. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
Answer: A
NEW QUESTION 61
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?
- A. The password is incorrect.
- B. The engineer needs to exchange the CallManager certificate.
- C. The tomcat certificate from Cluster B must be the publisher.
- D. Cluster IDs are not unique.
Answer: C
NEW QUESTION 62
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
- A. Real-Time Monitoring Tool
- B. SDI trace
- C. Dialed Number Analyzer
- D. SDL trace
Answer: C
NEW QUESTION 63
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 12[2-5]
- B. 1XX (urgent Priority Set)
- C. 12!
- D. 12X (urgent priority set)
Answer: A
NEW QUESTION 64
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?
- A. Change the destination-pattern on the outgoing dial peer to match "444333222".
- B. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
- C. Create specific matching for "222333444" on the incoming dial peer.
- D. Set up translation-profile on the incoming dial peer to match incoming traffic.
Answer: D
NEW QUESTION 65
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs must the administrator turn on? (Choose two).
- A. debug H.323 asn1
- B. debug H.225 media
- C. debug H.323 message
- D. debug H.225 asn1
- E. debug H.245 asn1
Answer: D,E
NEW QUESTION 66 
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?
- A. calling number 011613-555-1234 and numbering type "international"
- B. calling number 011-1-613-555-1234 and numbering type "subscriber"
- C. calling number 613-555-1234 and numbering type "national"
- D. calling number 613-555-1234 and numbering type "subscriber"
Answer: C
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:
NEW QUESTION 67
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.)
- A. three-way conference
- B. SIP trunk
- C. secure SIP lines
- D. T.38 fax relay
- E. transcoding
Answer: A,D
NEW QUESTION 68
Refer to the exhibit.
Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?
- A. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
- B. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- C. No DTMF is negotiated.
- D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
Answer: C
NEW QUESTION 69
......
Topics of Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam
In the 300-815 exam dumps, the following are also available:
1.0 Signaling and Media Protocols
- 1.2.b Call set up and tear down
- 1.1.d Session timers
- 1.1 Troubleshoot these elements of a SIP conversation
- 1.3 Troubleshoot media establishment
- 1.1.e UPDATE
2.0 CME/SRST Gateway Technologies
- 2.4.b Call park
- 2.4.a Hunt groups
- 2.2 Configure Cisco Unified CME dial plans
- 2.4.c Paging
- 2.4 Configure these advanced Cisco Unified CME features
3.0 Cisco Unified Border Element
- 3.1.b Voice translation rules and profiles
- 3.2 Troubleshoot these Cisco Unified Border Element dial plan elements
- 3.1.d Dial peers
- 3.2.f Signaling and media bindings
- 3.2.e Header and SDP manipulation with SIP profiles
- 3.1 Configure these Cisco Unified Border Element dial plan elements
- 3.2.a DTMF
- 3.1.e Header and SDP manipulation with SIP profiles
- 3.1.a DTMF
- 3.1.f Signaling and media bindings
4.0 Call Control and Dial Planning
- 4.1.g SIP trunking
- 4.1.c SIP route patterns
- 4.1.b Route patterns
- 4.2.a Translation patterns
- 4.2.f TEHO
- 4.1.a Translation patterns
- 4.2.e Standard local route group
- 4.1.f TEHO
- 4.2 Troubleshoot these globalized call routing elements in Cisco Unified Communications Manager
- 4.1 Configure these globalized call routing elements in Cisco Unified Communications Manager
- 4.2.b Route patterns
- 4.1.d Transformation patterns
- 4.2.d Transformation patterns
- 4.2.c SIP route patterns
- 4.1.e Standard local route group
5.0 Cisco Unified CM Call Control Features
- 5.1 Troubleshoot Call Admission Control (exclude RSVP)
- 5.6.b Meet-me
- 5.4 Configure call queuing
- 5.6 Configure supplementary functions
- 5.6.a Call park
- 5.5 Configure time of day routing
6.0 Mobility
- 6.1.b Extension Mobility 2019 Cisco Systems, Inc. This document is Cisco Public. Page 3
- 6.2.b Extension Mobility
- 6.2.c Device Mobility
- 6.1 Configure Cisco Unified Communications Manager Mobility
- 6.1.c Device Mobility
- 6.2.a Unified Mobility
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