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Free 2025 CCNP Collaboration 300-815 dumps are available by DumpsFree [Q39-Q59]

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Welcome to download the newest DumpsFree 300-815 PDF dumps: https://www.dumpsfree.com/300-815-valid-exam.html ( 176 Q&As)


Successfully passing the Cisco 300-815 exam can help network professionals enhance their career prospects and demonstrate their expertise in implementing advanced call control and mobility services in a Cisco network. With this certification, candidates can showcase their knowledge and skills to employers and organizations, and potentially advance their career in the field of networking and collaboration.


To prepare for the Cisco 300-815 exam, candidates must have a solid understanding of Cisco Unified Communications Manager and related services, as well as experience in implementing and troubleshooting these services. They must also be familiar with Cisco IOS and be comfortable with configuring and troubleshooting network devices. Successful completion of the Cisco 300-815 exam demonstrates an individual's expertise in advanced call control and mobility services, making them a valuable asset to any organization.

 

NEW QUESTION # 39
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...$/2404\0/ type any national 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...S/ /9132404 0/ type any subscriber plan any Isdn
  • D. voice translation-rule 1 rule 1 // // type any subscriber plan any isdn

Answer: B


NEW QUESTION # 40
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

  • A. A unicast music on hold audio source must be configured.
  • B. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
  • C. The maximum number of callers allowed in queue must be 10.
  • D. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
  • E. The phone button template must have the Queue Status Softkey configured.

Answer: B,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_01001101.html#CUCM_RF_C960BC9A_00


NEW QUESTION # 41
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 only under the route list
  • E. can be associated under the route group

Answer: A,D


NEW QUESTION # 42
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. Cluster IDs are not unique.
  • B. The engineer needs to exchange the CallManager certificate.
  • C. The password is incorrect.
  • D. The tomcat certificate from Cluster B must be the publisher.

Answer: D


NEW QUESTION # 43
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. parallel
  • B. sequential
  • C. peer
  • D. call-blast

Answer: C

Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmead m/cmecover.html#concept_84738A1DC1EE4FCD81E16B79CA9CCB47


NEW QUESTION # 44
An administrator is implementing a new dial-plan on Cisco Unified Border Element. The administrator must ensure that incoming dial-peers are matched based on the IP address from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?

  • A. dial-peer voice 1 voip
    incoming url request
  • B. dial-peer voice 1 voip
    incoming called-number
  • C. dial-peer voice 1 voip
    incoming url via
  • D. dial-peer voice 1 voip
    incoming url to

Answer: C


NEW QUESTION # 45
Refer to the exhibit. An administrator just implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily.
Which action resolves the issue?

  • A. Change to a range that does not result in MTP.
  • B. Change DTMF Signaling Method to "RFC 4733".
  • C. Disable SIP Rel1XX Options.
  • D. Change DTMF Signaling Method to "No Preference".

Answer: A


NEW QUESTION # 46
An administrator manages an environment with mobile users who commonly move between locations and take IP phones to remote sites. When mobile users are at one of the sites, users report issues with call quality, such as the audio cutting in and out. In addition, the affected area has a low bandwidth connection compared to other sites, so the administrator has put these devices in a region with an inter-region restriction of 8 kbps per call. Which feature must the administrator configure to mitigate the issue?

  • A. Extension Mobility Cross Cluster
  • B. Extension Mobility
  • C. Location Bandwidth Manager
  • D. Device Mobility

Answer: C


NEW QUESTION # 47
Where is the dtmf-relay command configured on Cisco Unified Border Element?

  • A. in the VoIP or POTS dial peers
  • B. in the VoIP dial peer
  • C. in global SIP configuration
  • D. in the voice-class VoIP configuration

Answer: B


NEW QUESTION # 48
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. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
  • B. Early Offer for G Clear Calls must be enabled.
  • C. Accept Audio Codec Preferences in Received Offer must be set to On.
  • D. Allow Passthrough of Configured Line Device Caller Information must be enabled.

Answer: A

Explanation:
https://community.cisco.com/t5/ip-telephony-and-phones/ring-back-while-183-session- progress/m-p/2813529#M315884


NEW QUESTION # 49
What are the elements for Device Mobility configuration?

  • A. device pool, Device Mobility group, and Cisco IP phone
  • B. device pool, Device Mobility group, and region
  • C. physical location, device pool, and Device Mobility group
  • D. physical location. Device Mobility group, and region

Answer: C

Explanation:
https://www.ciscopress.com/articles/article.asp?p=1249228&seqNum=4


NEW QUESTION # 50
Refer to the exhibit. An engineer is trying to set up a new deployment using the SIP provider with TCP for signaling. After troubleshooting, the customer notices that the matching incoming dial peer and outgoing dial peer did not generate the INVITE to the SIP provider. Why is the call failing?

  • A. The ITSP is not sending the ACK.
  • B. The ITSP is sending CLOSE state.
  • C. The ITSP is sending FIN.
  • D. The ITSP is not answering the SYN.

Answer: D


NEW QUESTION # 51
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?

  • A. System > Device Pool
  • B. System > Location Info
  • C. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
  • D. Call Routing > Route/Hunt > Local Route Group Names

Answer: D


NEW QUESTION # 52
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. no update-callerid
  • B. midcall-signaling passthru media-change
  • C. midcall-signallng passthru
  • D. midcall-signaling preserve-codec

Answer: B


NEW QUESTION # 53
Users report silence on the line when they try to connect to external voice numbers. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. The MGCP server seems to be responding. Which action must the engineer take to begin to investigate the media for this type of issue?

  • A. Run the debug H.323 gateway command on the Cisco router.
  • B. Run the debug mgcp media command on the Cisco router.
  • C. Run RTMP on the Cisco router.
  • D. Verify that the MGCP service is running under serviceability in Cisco UCM.

Answer: B


NEW QUESTION # 54
An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?

  • A. Configure an access list.
  • B. Configure a mobility identity.
  • C. Configure Mobile Voice Access.
  • D. Configure Single Number Reach.

Answer: A


NEW QUESTION # 55
Refer to the exhibit.

A no-way audio situation was experienced after the calling party on an outbound call placed the call on hold and then resumed from hold. Which action will resolve the issue?

  • A. Upgrade the Cisco Unified Border Element device.
  • B. Restart the SIP trunk on Cisco UCM.
  • C. Upgrade the Cisco Unified Border Element to 15.2.
  • D. Assign an MTP to the SIP trunk.

Answer: C


NEW QUESTION # 56
Refer to the exhibit. For long-distance calls, users must prefix their dialed number with "91". The translation pattern was created to strip the 91 as the PSTN expects a 10-digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?

  • A. Disable Use Calling Party's External Phone Number Mask on the route pattern.
  • B. Change the partition of the translation pattern from none to pstn_pt.
  • C. Enable Force Authorization Code on the route pattern.
  • D. Enable Use Calling Party's External Phone Number Mask on the translation pattern.

Answer: A


NEW QUESTION # 57
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

  • A. A unicast music on hold audio source must be configured.
  • B. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
  • C. The maximum number of callers allowed in queue must be 10.
  • D. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
  • E. The phone button template must have the Queue Status Softkey configured.

Answer: B,D

Explanation:
Section: Cisco Unified CM Call Control Features
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_01001101.html#CUCM_RF_C960BC9A_00


NEW QUESTION # 58
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. 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.
  • B. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.
  • 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. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
  • 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: B,E


NEW QUESTION # 59
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