300-815 Questions PDF [2024] Use Valid New dump to Clear Exam
Passing Cisco 300-815 Exam Using 2024 Practice Tests
NEW QUESTION # 25
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. CCSIP
- C. MTP
- D. OS Administration Page
Answer: A
NEW QUESTION # 26
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses
H.323protocol (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 Terminal Capability Set
- B. H.245 Open Logical Channel Ack
- C. H.245 Open Logical Channel
- D. H.225 Connect
Answer: C
NEW QUESTION # 27
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. route group
- B. hunt list
- C. line group
- D. hunt pilot
Answer: D
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference:
Reference: https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared-line/ td-p/2658015
NEW QUESTION # 28
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?
- A. voice translation-rule 1 rule 1 /^4...S/ /9132404 0/ type any subscriber plan any Isdn
- B. voice translation-rule 1
rule 1 /^4...V /2404\0/ type any subscriber plan any isdn - C. voice translation-rule 1
rule 1 /^4...$/2404\0/ type any national plan any Isdn - D. voice translation-rule 1 rule 1 // // type any subscriber plan any isdn
Answer: B
NEW QUESTION # 29
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 rel1xx disable
- B. voice-class sip conn-reuse
- C. voice-class midcall-signaling block
- D. voice-class sip early-media update block
Answer: A
NEW QUESTION # 30
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?
- A. Use TLS Certificates must be selected.
- B. A password has never been set for ILS.
- C. Trust certificates for ILS have not been installed on the clusters
- D. The Cluster IDs have not been set to unique values
Answer: D
NEW QUESTION # 31
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. SDL trace
- D. Dialed Number Analyzer
Answer: D
NEW QUESTION # 32
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port 5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)
- A. Destination Port in SIP Information section of the SIP Trunk configuration
- B. Incoming Port in Security Information of the SIP Profile configuration.
- C. Destination Port in SIP Trunk Security Profile configuration
- D. Incoming Port in SIP Information section of the SIP Trunk configuration.
- E. Incoming Port in SIP Trunk Security Profile configuration
Answer: A,E
NEW QUESTION # 33
Refer to the exhibit.
DN 1003 was the last to ring during the most recent call. Which hunting method ensures that DN 1005 is presented with the next call when the hunt pilot is dialed?
- A. sequential
- B. call-blast
- C. parallel
- D. peer
Answer: D
NEW QUESTION # 34
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 > Open From Local Disk
- D. Voice/Video > Session Trace Log View > Real Time Data
Answer: D
Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications- manager-callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html
NEW QUESTION # 35
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. +E.164 alternate number mask
- C. external phone number mask
- D. enterrise alternate number mask
- E. called party transform mask
Answer: A,C
NEW QUESTION # 36
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port
5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration?
(Choose two.)
- A. Destination Port in SIP Information section of the SIP Trunk configuration
- B. Incoming Port in Security Information of the SIP Profile configuration.
- C. Destination Port in SIP Trunk Security Profile configuration
- D. Incoming Port in SIP Information section of the SIP Trunk configuration.
- E. Incoming Port in SIP Trunk Security Profile configuration
Answer: A,E
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 37
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. VIDEO
- B. DTMF
- C. AUDIO
- D. BFCP
- E. FAX
Answer: B,D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION # 38
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
NEW QUESTION # 39
Refer to the exhibit.
Outbound calls to the service provider cause intermittent errors due to a codec mismatch. The internal network sends early offer SDP that contains only G.711 A-law. The service provider reports that some destinations support only G.711 A-law while others support only iLBC. The service provider also allows only 20 active calls at a time Which configuration allows successful media negotiation for all calls using outbound dial peers 5002 and 5003?
- A. Option C
- B. Option D
- C. Option B
- D. Option A
Answer: B
NEW QUESTION # 40
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. T.38 fax relay
- C. transcoding
- D. SIP trunk
- E. secure SIP lines
Answer: A,B
NEW QUESTION # 41
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. +E.164 alternate number mask
- C. external phone number mask
- D. enterrise alternate number mask
- E. called party transform mask
Answer: A,C
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html
NEW QUESTION # 42
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?
- A. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
- B. Enable SIP ReMXX Options on the SIP profile.
- C. Increase Retry INVITE to 20 seconds on the SIP profile.
- D. Change Session Refresh Method on the SIP profile to INVITE.
Answer: D
NEW QUESTION # 43
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