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Implementing Cisco Advanced Call Control and Mobility Services (CLASSM) Questions 86–90 (page: 19)

Updated On: 10-Feb-2026



Refer to the exhibit. An engineer troubleshoots why a new SIP trunk does not come up on Cisco UCM.
Which action resolves this issue?

  1. The device cube.abc.com must have the SIP stack turned up.
  2. The DNS server that Cisco UCM is pointing to needs an A Record added for cube.abc.com.
  3. The ACL that is configured on cube.abc.com must include the Cisco UCM IP address.
  4. The destination port must be set to 5061.

Answer(s): B

Explanation:

The SIP trunk configuration uses a Fully Qualified Domain Name (FQDN): cube.abc.com, but the trunk status is down with status reason local=3, which indicates a DNS resolution failure. To resolve this, the DNS server that Cisco UCM uses must have an A record for cube.abc.com so that it can resolve the IP address of the CUBE and establish the SIP connection.





Refer to the exhibit. Users are experiencing no-way audio after an incoming PSTN call is answered using the pickup softkey.
Which action resolves this issue?

  1. Open UDP ports 16384 - 32767 on the firewall.
  2. Configure "rtp-port range 16384-48198" on the Cisco Unified Border Element.
  3. Block SIP update using SIP profiles on the Cisco Unified Border Element.
  4. Remove the access list from the router.

Answer(s): C

Explanation:

The SIP call flow shows multiple SIP UPDATE messages after the call is picked up. In some scenarios -- especially with PSTN calls and certain ITSPs -- these midcall SIP UPDATEs can lead to mismatched media negotiation or cause the CUBE to incorrectly update media parameters, resulting in no-way audio.
Blocking SIP UPDATE messages using a SIP profile on the CUBE prevents this renegotiation and preserves working media parameters, resolving the audio issue.



A user has a problem calling a specific PSTN number. An engineer investigates the issue on a Cisco UCM server and a Cisco Unified Border Element router, and discovers that the ITSP rejects the outbound SIP INVITE with a 422 Session Timer Too Small SIP message.
What must the engineer do to resolve the issue?

  1. Increase the Min-SE SIP header timer on the appropriate Cisco Unified Border Element dial peer.
  2. Increase the SIP Session Expires Timer service parameter in Cisco UCM.
  3. Decrease the Min-SE SIP header timer on the appropriate Cisco Unified Border Element dial peer.
  4. Decrease the SIP Session Expires Timer service parameter in Cisco UCM.

Answer(s): A

Explanation:

A 422 Session Timer Too Small error means the Min-SE (Minimum Session Expiration) value in the SIP INVITE is below the acceptable threshold of the ITSP. To resolve this, you must increase the Min-SE timer on the dial peer in the Cisco Unified Border Element, which controls the SIP signaling. This ensures that the session timer negotiation meets the ITSP's minimum requirement.



A company is using third-party phones on the internal network at a location behind a Cisco Unified Border Element. These phones seem to work with only two types of codecs, which are high-complexity codecs.
Which configuration ensures that calls are completed with the correct and working media each time?

  1. Configure a codec priority list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the dial-peer voice.
  2. Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the VoIP peer.
  3. Configure a codec preference list on the Cisco Unified Border Element with the VoIP class codec command, set the codec preference list codecs, and add it to the dial peer.
  4. Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec preference list codecs, and add it to the dial peer.

Answer(s): D

Explanation:

To ensure that calls always use one of the supported high-complexity codecs, you configure a codec preference list using the voice class codec command on the CUBE. You then specify the preferred codecs in order of priority and apply this codec class to the dial peer. This guarantees consistent codec negotiation for successful call setup with third-party phones.





Refer to the exhibit. An engineer configured Call Admission Control on Cisco Unified Border Element to reject SIP calls when the aggregate bandwidth that is required for the calls exceeds the threshold. However, users complain of dropped calls when they place calls on hold during peak periods.
Which solution resolves this challenge?

  1. Bind media to interface GigabitEthernet0/0.200.
  2. Enable the error-code-override feature on the dial peer.
  3. Enable the midcall-exceed feature on the dial peer.
  4. Bind media to the loopback interface.

Answer(s): C

Explanation:

Enabling the midcall-exceed feature allows Cisco Unified Border Element Call Admission Control to permit midcall bandwidth changes, such as when a call is placed on hold and media characteristics change. Without this feature, CUBE recalculates bandwidth during midcall events and may reject the call if the aggregate bandwidth exceeds the configured threshold, causing calls to drop during peak periods. The midcall-exceed feature prevents call drops by allowing these midcall bandwidth increases to proceed.



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