Get Started: 300-815 Exam [2025] Dumps Cisco PDF Questions
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NEW QUESTION # 102
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 613-555-1234 and numbering type "subscriber"
- B. calling number 011-1-613-555-1234 and numbering type "subscriber"
- C. calling number 011613-555-1234 and numbering type "international"
- D. calling number 613-555-1234 and numbering type "national"
Answer: D
NEW QUESTION # 103
Drag and Drop Question
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 # 104
Signal number reach call phone that not answered are leaving voicemails on the cell phone rather the corporate mailbox. Which two options will resolve this issue? (Choose two.)
- A. Decrease the T302 timer
- B. Check the Enable Unified Mobility features checkbox
- C. Check the Enable Extend and Connect checkbox
- D. Decrease the T301 timer Decrease the Answer Too Late timer
Answer: B,D
NEW QUESTION # 105
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. peer
- D. parallel
Answer: C
NEW QUESTION # 106
Refer to the exhibit.
An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?
- A. sip-ua
disable-early-media 180 - B. voice service voip sip rehxx require 100rel
- C. voice service voip sip send 180 sdp
- D. voice service voip sip no reMxx
Answer: B
NEW QUESTION # 107
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?
- A. midcall-signaling preserve-codec
- B. no update-callerid
- C. midcall-signaling passthru media-change
- D. midcall-signallng passthru
Answer: C
NEW QUESTION # 108
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
- A. Voice/Video > Session Trace Log View > Real Time Data
- B. Voice/Video > Session Trace Log View > Open From Local Disk
- C. System > Tools > Trace and Log Central
- D. Analysis Manager > Inventory > Trace File Repositories
Answer: A
NEW QUESTION # 109
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 early-media update block
- B. voice-class sip conn-reuse
- C. voice-class midcall-signaling block
- D. voice-class sip rel1xx disable
Answer: D
NEW QUESTION # 110
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 Cisco Unified CM Enterprise Parameters
- C. in the telephony service configuration
- D. in the queuing configuration
Answer: D
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 # 111
A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)
- A. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
- B. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.
- C. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call.
- D. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.
- E. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.
Answer: D,E
NEW QUESTION # 112
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 30 limit 2 recall alternate 3002
- D. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
Answer: D
Explanation:
Section: Cisco Unified CM Call Control Features
NEW QUESTION # 113
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. codec mismatch
- C. phone class of service issue
- D. missing Call Admission Control
Answer: D
NEW QUESTION # 114
ABC company has decided to implement hunt groups to help distribute calls between members. In order to implement this, administrator must configure hunt list, hunt group, and line groups on Cisco UCM. Which distribution algorithms should the administrator implement?
- A. Top Down, Circular, Broadcast
- B. Top Down, Round Robin, Distribute
- C. Top Down, Round Robin, Broadcast
- D. Sequantial, Circular, Broadcast
Answer: A
NEW QUESTION # 115
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
- A. interworking between an OOB method and RFC2833 for flow-through calls
- B. interworking between h245-alpha numeric and sip-kpml
- C. interworking between an OOB method and RFC2833 for flow-around calls
- D. interworking between h245-signal and rtp-nte
Answer: C
Explanation:
CUBE requires transcoding resources registered locally in these scenarios
...
Interworking between an OOB method and RFC2833 for flow-around calls
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/200412-DTMF-Relay-and-Interworking-on-CUBE.html
NEW QUESTION # 116
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. codec mismatch
- C. phone class of service issue
- D. missing Call Admission Control
Answer: D
NEW QUESTION # 117
Users report slrange behaviors when Ihey call external numbers from the company phone system. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. There is no indication of the possible problem, but the connection seems to be established Which command meets the requirement to capture all the data needed to debug the problem?
- A. debug mgcp media tracelevel moderate
- B. debug mgcp media tracelevel critical
- C. debug mgcp media tracelevel critical verbose
- D. debug mgcp media tracelevel critical all
Answer: A
NEW QUESTION # 118
In RTMT, how many concurrent trace collections can you schedule to download the trace files to an SFTP or FTP server on your network?
- A. three
- B. five
- C. four
- D. six
Answer: D
NEW QUESTION # 119
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?
- A. dtmf-relay cisco-rtp
- B. dtmf-relay sip-kpml
- C. dtmf-relay h245-signal
- D. dtmf-relay rtp-nte
Answer: B
NEW QUESTION # 120 
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. 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.
- B. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- C. 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.
- D. 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.
- E. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
Answer: B,D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION # 121
Users report strange behaviors when they call external numbers from the company phone system. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two TI circuits. There is no indication of the possible problem, but the connection seems to be established. Which command meets the requirement to capture all the data needed to debug the problem?
- A. debug mgcp media tracelevel critical
- B. debug mgcp media tracelevel critical verbose
- C. debug mgcp media tracelevel moderate
- D. debug mgcp media tracelevel critical all
Answer: D
NEW QUESTION # 122
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Enable Secondary Dial tone on Analog and Digital FXO Ports.
- B. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- C. Configure the command no ip address trusted authenticate under "voice service voip".
- D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
Answer: B
Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmead m/cmetoll.html#concept_E435095B0E2845C78D60332753288337
* Direct-inward-dial isdn is not supported for incoming ISDN overlap dialing call.
NEW QUESTION # 123
Which set of commands binds SIP media and signaling to interface GigabitEthernet0/0 when dial peer 1 is chosen for call routing?
- A. voice service voip
sip
bind all source-interface GigabitEthernet0/0 - B. dial-peer voice 1 voip
voice-class bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0 - C. voice service voip
bind sip source-interface GigabitEthernet0/0 - D. dial-peer voice 1 voip
voice-class source interface GigabitEthernet0/0
Answer: A
NEW QUESTION # 124
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