CLCOR Voice Gateways And Session Border Controllers — Timed 10 Questions

Question 1

What is the default dial-peer used if no other dial-peer matches a call on a Cisco IOS XE gateway?

Accepted Answer

Dial-peer 0. Cisco IOS XE gateways use dial-peer 0 as the default dial-peer that matches any call not explicitly matched by a configured dial-peer. Dial-peer 0 is a permanent, built-in dial-peer that cannot be deleted or modified, and it serves as the catch-all for inbound and outbound call routing when no other dial-peer matches. This ensures that calls are not silently dropped due to missing dial-peer configuration.

Answer

Dial-peer none

Answer

Dial-peer 1

Answer

Dial-peer 100

Question 2

A CUBE is rejecting calls because of a mismatch in supported DTMF methods. Which command forces CUBE to offer both RFC2833 and OOB DTMF?

Accepted Answer

dtmf-relay rtp-nte sip-notify. dtmf-relay rtp-nte digit-drop is a common configuration, but to support multiple, the dial-peer must explicitly define the relay types.

Answer

dtmf-relay all

Answer

dtmf-relay rfc2833

Answer

dtmf-relay h245-alphanumeric

Question 3

A CUBE is experiencing media cut-off issues after 30 seconds. Which parameter in the dial-peer configuration is the most likely cause if the re-invite is not being processed correctly?

Accepted Answer

session-timer. The timer-hold command controls how long the gateway waits for re-invites, but the issue is typically related to the session-timer which governs the SIP session refresh.

Answer

dtmf-relay

Answer

voice-class codec

Answer

timer connect

Question 4

You are configuring CUBE for SIP-to-SIP header manipulation. You need to remove the Diversion header from outgoing INVITE requests. Which component is required to achieve this?

Accepted Answer

voice class sip-profiles. Removing a SIP header such as Diversion from outgoing INVITEs is done with a voice class sip-profiles configuration, which contains Lua rules that match and remove headers. The profile is then applied to the outbound dial-peer. Dial-peers, codec classes, and translation-rules do not manipulate SIP headers.

Answer

dial-peer voice

Answer

voice class codec

Answer

voice translation-rule

Question 5

You are configuring a dial-peer to route calls to a PSTN gateway using SIP. Which command is required to enable the use of the SIP protocol on the dial-peer?

Accepted Answer

session-protocol sip. The session-protocol sip command is the foundational command required to define the signaling protocol for a dial-peer.

Answer

session-protocol sipv2

Answer

signaling-protocol sip

Answer

voice-class protocol sip

Question 6

To allow a CUBE to handle calls for multiple tenants, which feature should be configured?

Accepted Answer

VRF. VRFs (Virtual Routing and Forwarding) are used in CUBE for multi-tenancy and network isolation.

Answer

Partitioned route patterns

Answer

Dial-peer groups

Answer

SIP headers

Answer

Voice class tenant

Question 7

You are migrating from H.323 to SIP. Which command on the gateway ensures that H.323 is disabled for new calls?

Accepted Answer

no h323-gateway. Removing the h323-gateway command or not defining h323 dial-peers prevents H.323 usage.

Answer

no signaling h323

Answer

disable h323

Answer

shutdown h323

Question 8

Which command displays the current status and statistics of all active calls on a Cisco voice gateway?

Accepted Answer

show call active voice. show voice call status provides a comprehensive list of active calls and their status.

Answer

debug calls

Answer

show call history

Answer

show voice status

Question 9

You are deploying CUBE and need to ensure secure media. Which protocol should you enable on the dial-peer?

Accepted Answer

secure rtp. SRTP (Secure Real-time Transport Protocol) is required for encrypted media, configured via secure rtp.

Answer

tls-srtp

Answer

encryption srtp

Answer

secure media

Question 10

Which THREE commands are used to configure a SIP profile in CUBE?

Accepted Answer

voice-class sip-profiles apply. Option C, `voice class sip-profiles`, is correct because it is the global configuration command that creates and names a SIP profile container, entering voice-class configuration mode where the profile rules are defined. Option D, `request/response`, is correct because within the SIP profile it specifies whether the manipulation rules apply to inbound requests or responses (and their direction), which is the syntax used to build the profile's rule set. Option B, `voice-class sip-profiles apply`, is correct because it attaches the previously created SIP profile to a dial peer (or to the sip-ua) so the header manipulation is actually enforced on that call leg. Option A, `dial-peer voice`, is not correct here because it merely creates a dial peer and does not itself configure a SIP profile, though it is the context in which the profile is later applied. Option E, `sip-ua`, is not correct because it enters SIP user-agent configuration mode for global SIP parameters and does not define or build a SIP profile.

Answer

dial-peer voice

Answer

sip-ua