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Cisco 300-815 Exam (page: 4)
Cisco Implementing Advanced Call Control and Mobility Services (CLASSM)
Updated on: 31-Mar-2026

Viewing Page 4 of 32
300-815 PDF 245 Questions

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?

  1. Run RTMP on the Cisco router.
  2. Verify that the MGCP service is running under serviceability in Cisco UCM.
  3. Run the debug mgcp media command on the Cisco router.
  4. Run the debug H.323 gateway command on the Cisco router.

Answer(s): C



What is a function of the metadata carried in SIP sessions between the recording client and the recording server?

  1. It forks RTP media to the recorder.
  2. It provides advanced capabilities, such as speech analytics.
  3. It sets up a new SIP session
  4. It identifies the participant change due to transfers during the call.

Answer(s): D





Refer to the exhibit. An administrator just upgraded to Cisco UCM to version 14 and started SIP implementation with some new SIP trunks. During the testing, an error was reported when making a call.
Which action resolves the issue?

  1. Make sure the destination number is configured correctly and the device is registered.
  2. Disable SIP Rel1XX Options.
  3. Change DTMF Signaling Method to "RFC 4833".
  4. Change to a valid range.

Answer(s): A

Explanation:

The SIP response code 404 Not Found indicates that the called number was not found on the destination system -- typically meaning the destination device is not registered, or the dialed number is incorrect or not configured on the remote endpoint.
In this log, the INVITE was sent to 10.122.44.183, but the response is 404 Not Found, which means that IP

doesn't recognize or have a match for the called SIP URI or number.
To resolve this, verify that the dialed number exists on the remote system and that the corresponding device or trunk is registered and reachable.





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?

  1. Change DTMF Signaling Method to "No Preference".
  2. Disable SIP Rel1XX Options.
  3. Change to a range that does not result in MTP.
  4. Change DTMF Signaling Method to "RFC 4733".

Answer(s): D

Explanation:

When RFC 2833 is selected as the DTMF method, but the SIP trunk or endpoint does not support it properly, Cisco UCM may invoke an MTP (Media Termination Point) to interwork DTMF signaling. To avoid unnecessary MTP resource usage, it's recommended to use RFC 4733, which is the updated and widely accepted standard for out-of-band DTMF over RTP.
Switching to RFC 4733 ensures better compatibility and avoids forcing UCM to insert an MTP, which saves resources and improves call handling efficiency.





Refer to the exhibit. An administrator received reports that users do not hear a ringback when they dial specific external numbers. For example, the Cisco Unified Border Element receives a delayed offer invite from Cisco UCM and responds with the 183 session progress in the exhibit. Assume that the PSTN is responsible for generating ringback in these problematic scenarios.
Which action resolves the issue?

  1. On Cisco Unified Border Element:
    voice service voip sip rel1xx disable 100rel
  2. Edit the configuration for SIP Rel1XX Options in the appropriate SIP profile on Cisco UCM to Send PRACK if 1xx Contains SDP.
  3. On Cisco Unified Border Element:
    voice service voip sip rel1xx require 100rel
  4. Enable early media on the appropriate SIP trunk on Cisco UCM.

Answer(s): B



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?

  1. debug mgcp media tracelevel critical
  2. debug mgcp media tracelevel critical all
  3. debug mgcp media tracelevel critical verbose
  4. debug mgcp media tracelevel moderate

Answer(s): C

Explanation:

The command debug mgcp media tracelevel critical verbose provides comprehensive media-related debug information for MGCP-controlled gateways, including RTP stream setup, codec negotiation, and media path issues. This level of detail is essential when the call connects but behaves abnormally, as reported. Using the verbose keyword ensures all relevant information is captured for accurate troubleshooting.



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.)

  1. 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
  2. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
  3. 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.
  4. 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.
  5. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed.

Answer(s): B,D



A voice administrator enabled secure media between multiple Cisco UCM clusters that are connected via a SIP trunk.
Which configuration parameter in the SIP trunk must be enabled to allow encrypted media traffic to traverse the SIP trunk?

  1. Check the Allow iX Application Media check box in the SIP Profile.
  2. Check the SRTP Allowed check box in the SIP Trunk configuration.
  3. Check the RTP Allowed check box in the SIP Trunk configuration.
  4. Uncheck the RTP Allowed check box in the SIP Trunk configuration.

Answer(s): B

Explanation:

To allow encrypted media (Secure RTP or SRTP) to traverse a SIP trunk between Cisco UCM clusters, you must enable the SRTP Allowed option in the SIP Trunk configuration. This permits the trunk to negotiate and carry SRTP streams, ensuring media encryption between clusters.



Viewing Page 4 of 32



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