Pass Your Cisco 300-815 Exam with Correct 120 Questions and Answers
Latest [Jan 18, 2024] 2024 Realistic Verified 300-815 Dumps
Candidates for the Cisco 300-815 exam should have a solid understanding of Cisco Collaboration technologies, including Cisco Unified Communications Manager, Cisco Unity Connection, and Cisco Unified Contact Center Express. They should also have experience configuring and troubleshooting Cisco Collaboration solutions in a real-world environment.
NEW QUESTION # 56
Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?
- A. in Cisco Unified CM Service Parameters
- B. in the queuing configuration
- C. in the telephony service configuration
- D. in Cisco Unified CM Enterprise Parameters
Answer: B
Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications- manager-callmanager/200453-Configure-CUCM-Native-Call-Queuing-Featu.html
NEW QUESTION # 57
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 1XX (urgent Priority Set)
- B. 12!
- C. 12X (urgent priority set)
- D. 12[2-5]
Answer: C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 58
Refer to the exhibit.
An engineer is troubleshooting an issue where inbound calls to Cisco UCM with early media fail to establish. While investigating the issue, the engineer finds that Cisco UCM is set to require a PRACK. but the Cisco Unified Border Element is not sending it. Which command is causing this issue?
- A. voice-class sip conn-reuse
- B. voice-class midcall-signaling block
- C. voice-class sip rel1xx disable
- D. voice-class sip early-media update block
Answer: C
NEW QUESTION # 59
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. OS Administration Page
- C. CCSIP
- D. MTP
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 60
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
- B. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- C. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- D. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
Answer: C
NEW QUESTION # 61
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. Restart the subscriber
- B. Activate the Cisco CallManager service.
- C. Activate the Cisco Dialed Number Analyzer Server service.
- D. Activate the Cisco Dialed Number Analyzer service.
- E. Activate the Cisco Extended Functions service.
Answer: C,D
NEW QUESTION # 62
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 for SIP registrar service
- C. configuration items common for all SIP phones
- D. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
Answer: A
NEW QUESTION # 63
Which IOS command creates a SIP-enabled dial peer?
- A. dial-peer voice 20 voip
- B. dial-peer voice 20 pots
- C. dial peer voice 20 sip
- D. voice dial-peer 20 sip
Answer: A
Explanation:
Section: Cisco Unified Border Element
Explanation/Reference: https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6
NEW QUESTION # 64
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.)
- A. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- B. 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.
- C. 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.
- D. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- E. 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.
Answer: A,B
NEW QUESTION # 65
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. No DTMF is negotiated.
- C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- D. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
Answer: B
NEW QUESTION # 66
Where is the dtmf-relay command configured on Cisco Unified Border Element?
- A. in the VoIP or POTS dial peers
- B. in global SIP configuration
- C. in the VoIP dial peer
- D. in the voice-class VoIP configuration
Answer: C
Explanation:
Section: Cisco Unified Border Element
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf- relay.html
NEW QUESTION # 67
Drag and drop the commands from the bottom to the blanks in the code to implement a translation rule to allow only 11 digits to be received over a SIP trunk to a SIP provider. The Cisco UCM is currently sending calls to the Cisco Unified Border Element in E.
164 format. Not all options are used.
Answer:
Explanation:
NEW QUESTION # 68
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
- A. hunt list
- B. hunt pilot
- C. route group
- D. line group
Answer: D
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-1201_chapter_010101.html
NEW QUESTION # 69
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 Ack
- B. H.245 Open Logical Channel
- C. H.225 Connect
- D. H.245 Terminal Capability Set
Answer: B
Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
NEW QUESTION # 70
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. FQDN of the servers defined in DNS
- B. username and secret key
- C. passwords
- D. TokenID
- E. TLS certificates
Answer: C,E
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/ sysConfig/11_5_1_SU1/cucm_b_system-configuration-guide-1151su1/cucm_b_system-configuration-guide-
1151su1_chapter_011001.pdf
NEW QUESTION # 71
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. answer-address
- B. incoming called-number
- C. incoming uri
- D. target carrier-id
Answer: C
Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8
NEW QUESTION # 72
An administrator sees the voice register pool 1 command in a Cisco UCME configuration. Which configuration is occurring in this section?
- A. configuration for a pool of SIP phones (similar to device pool on Cisco UCM)
- B. configuration for a single SIP phone
- C. configuration for SIP registrar service
- D. configuration items common for all SIP phones
Answer: B
NEW QUESTION # 73
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?
- A. Change the partition of the route pattern
- B. Add a sip trunk inside route group Standard Local Route Group.
- C. Modify the route pattern to add a prefix of 91
- D. Add a local route group on the device pool configuration.
Answer: D
NEW QUESTION # 74
Which services are needed to successfully implement Cisco Extension Mobility in a standalone Cisco Unified Communications Manager server?
- A. Cisco CallManager, Cisco TFTP, and Cisco Extension Mobility
- B. Cisco CallManager, Cisco TFTP, and Cisco CallManager SNMP Service
- C. Cisco Extended Functions, Cisco Extension Mobility, and Cisco AXL Web Service
- D. Cisco TAPS Service, Cisco TFTP, and Cisco Extension Mobility
Answer: A
NEW QUESTION # 75
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be assigned directly to the route pattern
- B. chooses the route group that is configured under the device pool of the calling-party device
- C. can be associated only under the route list
- D. chooses the route group that is configured under the device pool of the called-party device
- E. can be associated under the route group
Answer: B,C
NEW QUESTION # 76
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. +E.164 alternate number mask
- B. AAR destination mask
- C. external phone number mask
- D. enterrise alternate number mask
- E. called party transform mask
Answer: B,C
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html
NEW QUESTION # 77
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. secure SIP lines
- B. three-way conference
- C. SIP trunk
- D. T.38 fax relay
- E. transcoding
Answer: B,D
Explanation:
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_sip_isr4000.html
NEW QUESTION # 78
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. No DTMF is negotiated.
- C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- D. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
Answer: B
NEW QUESTION # 79
Due to a shortage of physical interfaces on a device the administrator requires that a loopback for RTP is used. Which command is required when using a loopback interface for RTP?
- A. voice-class sip resources priority mode passthrough
- B. voice-class sip bind media source-interface Loopback0
- C. voice-class sip bind control source-interface Loopback0
- D. voice-class sip early-offer forced.
Answer: B
NEW QUESTION # 80
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 Active Learned Object Upper Limit
- D. ILS Max Number of Learned Objects in Database
Answer: D
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 # 81
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Overview of the Exam Outline
The content of 300-815 exam contains six domains, and questions taken from them will pop up during the test. However, they won't appear evenly, and some sections take up a larger portion of the syllabus than others. Take a look at them below:
- Signaling and Media Protocols
This module covers about 20% of the test. It is required that you know how to troubleshoot the various elements of a SIP dialogue, including session timers, mid-call signaling, and more. You also need to equip yourself with a clear idea of troubleshooting certain protocol elements and media establishment.
- Call Control and Dial Planning
This domain has weight of about 25%. You are to have a general idea of troubleshooting and setting up the globalized call routing elements in CU-CM, such as the patterns for route, translation, transformation, and others.
- Cisco Unified CM Call Control Features
This module takes about 20% of the syllabus. The subdomains include troubleshooting of the Call Admission Control and configuring ILS, URI synchronization, GDPR, hunt groups, call queuing, time of day routing, and supplementary roles.
- CME/SRST Gateway Technologies
About 10% of the questions come from this domain. It assesses one’s knowledge of how to configure CU-CME for SIP telephone registration and dial plans. Also, you should have a solid understanding of how to enforce toll fraud prevention and configure the advanced CU-CME features and SIP SRST gateway.
- Cisco Unified Border Element
This section takes about 15% of the content. It covers the basic skills in handling the procedures involved setting up as well as troubleshooting the CU-BE dial plan components, including DTMF, voice translation rules and profiles, codec preference list, plus signaling and media bindings.
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