CLCOR Voice Gateways And Session Border Controllers — Timed 20 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

Question 11

Which TWO of the following are required to successfully register a CUBE with a SIP provider?

Accepted Answer

authentication credentials. To register a Cisco Unified Border Element (CUBE) with a SIP provider, you must configure the SIP user agent (sip-ua) with the registrar and credentials, so option E (sip-ua configuration) is required because it enables the registration mechanism and points CUBE to the provider's registrar. Option A (authentication credentials) is also required because the SIP provider typically challenges the REGISTER with 401/407 and CUBE must supply a valid username/password (via credentials under sip-ua) to authenticate successfully. Options B (voice class codec), C (dial-peer voice), and D (voice translation-rule) are not required for registration itself; they are used for media negotiation, call routing, and number manipulation respectively, which apply to call handling rather than the REGISTER transaction.

Answer

voice class codec

Answer

dial-peer voice

Answer

voice translation-rule

Question 12

Which TWO of the following are valid methods to troubleshoot SIP call failures on a CUBE?

Accepted Answer

show call active voice. Option C (show call active voice) is correct because it displays real-time details of active voice call legs on the CUBE, including SIP signaling state, call IDs, and media information, which helps identify where a SIP call setup or teardown is failing. Option E (debug ccsip messages) is correct because it enables tracing of SIP messages (INVITE, 100 Trying, 180 Ringing, 200 OK, BYE, etc.) sent and received by the CUBE, which is the primary tool for diagnosing SIP signaling failures such as incorrect headers, SDP mismatches, or response codes. Option A (debug voice translation) is not a valid command for SIP call troubleshooting; voice translation debugging relates to number translation rules, not SIP signaling. Option B (debug voice class) is not a valid Cisco IOS command for troubleshooting SIP calls. Option D (show sip-profile status) is not a valid command; SIP profiles are configured with sip-profiles but there is no such status command for troubleshooting SIP call failures.

Answer

debug voice translation

Answer

debug voice class

Answer

show sip-profile status

Question 13

Which THREE steps are required when configuring a new SIP trunk on a CUBE router?

Accepted Answer

Set session-protocol sip. SIP trunks require defining a dial-peer, the session target, and the session protocol.

Answer

Configure route group

Answer

Configure phone buttons

Question 14

Which THREE factors influence dial-peer matching for incoming calls on a Cisco IOS XE gateway?

Accepted Answer

answer-address. For inbound dial-peer matching on a Cisco IOS XE gateway, the gateway evaluates the called number against the incoming called-number command (E), which matches the full called number of the incoming call and is the most preferred inbound match criterion. It also evaluates the calling number against the answer-address command (C), which matches based on the calling party number of the inbound call. Additionally, the incoming uri command (D) matches the incoming request URI (typically for SIP calls), allowing dial-peer selection based on the URI rather than just the number. These three are the standard inbound dial-peer matching criteria on IOS XE gateways. In contrast, destination-pattern (A) is an outbound matching criterion used to match the dialed digits for outbound dial peers, and session target (B) simply specifies where the outbound call is sent (e.g., an IP address or voice port) rather than influencing inbound dial-peer selection.

Answer

destination-pattern

Answer

session target

Question 15

A CUBE is deployed between Cisco Unified Communications Manager and an ITSP. The ITSP sends SIP OPTIONS pings to the CUBE, but the CUBE does not respond, causing the ITSP to mark the trunk as down. Which two configurations are required on the CUBE to respond to these OPTIONS requests? (Choose two.)

Accepted Answer

Configure 'allow-connections sip to sip' under voice service voip.. To respond to SIP OPTIONS pings, the CUBE must have SIP-to-SIP connections allowed globally and an incoming dial peer that matches the ITSP's signaling source. The 'allow-connections sip to sip' command enables CUBE functionality, while the dial peer ensures the OPTIONS request is matched and processed. Without both, the CUBE will not reply, and the ITSP will consider the trunk down.

Answer

Configure 'sip-ua' with 'options-ping' and apply it to the ITSP-facing dial peer.

Answer

Configure a SIP profile with the 'options' command under voice service voip and apply it to the incoming dial peer.

Answer

Configure a voice class sip-profiles with 'request OPTIONS sip-header Allow-Header modify' and apply it to the incoming dial peer.

Question 16

A Cisco CUBE (Cisco Unified Border Element) is configured to route calls between two SIP trunks. The administrator wants to limit the number of concurrent SIP-to-SIP calls traversing the CUBE to 100 to prevent oversubscription. Which command should be applied to the voice service voip configuration to enforce this limit?

Accepted Answer

max-conn 100. The max-conn command under voice service voip sip is the correct way to limit the number of concurrent SIP calls on a CUBE. It provides a global cap that applies to all SIP dial peers, ensuring the CUBE does not exceed the specified number of connections. This is essential for capacity planning and preventing resource exhaustion.

Answer

max-calls 100

Answer

call threshold global 100

Answer

sip max-calls 100

Question 17

A network engineer is configuring a Cisco IOS XE gateway to register with a SIP provider. The provider requires the gateway to send SIP REGISTER messages with a specific contact user parameter. The engineer has configured the registrar and credentials. Which additional command is needed under the SIP UA configuration to include a custom user parameter in the Contact header of outgoing REGISTER messages?

Accepted Answer

contact-header user-param. To insert a custom user parameter into the Contact header of SIP REGISTER messages, the contact-header user-param command must be configured under sip-ua. This allows the gateway to meet the SIP provider's specific requirements for registration. Without it, the REGISTER may be rejected or not properly identified.

Answer

sip-contact user-param

Answer

user-param contact

Answer

contact-user-param

Question 18

A Cisco Unified Border Element (CUBE) is configured to perform SIP-to-SIP interworking. Calls from the service provider arrive with the P-Asserted-Identity header, but the enterprise SIP trunk requires the From header to contain the calling number. You need to copy the P-Asserted-Identity value into the From header for outgoing INVITEs. Which SIP profile configuration achieves this?

Accepted Answer

request INVITE sip-header From copy sip-header P-Asserted-Identity. The correct SIP profile command copies the P-Asserted-Identity header value into the From header for outgoing INVITEs. It uses the 'request INVITE' context, specifies the source as P-Asserted-Identity and the destination as From. This ensures that the enterprise SIP trunk receives the calling number in the From header as required, while preserving the original P-Asserted-Identity header for other purposes.

Answer

request INVITE sip-header P-Asserted-Identity move sip-header From

Answer

response INVITE sip-header P-Asserted-Identity copy sip-header From

Answer

request INVITE sip-header P-Asserted-Identity copy sip-header From

Question 19

A CUBE is deployed to interconnect a Cisco Unified Communications Manager cluster with a SIP service provider. The provider requires that all outbound calls use a specific SIP profile that includes early offer and session refresh. Which two configuration steps are required on the CUBE to ensure that outbound SIP INVITEs comply with these requirements? (Choose two.)

Accepted Answer

Enable session refresh by configuring 'voice-class sip session refresh' under the outbound dial-peer.. To meet the provider's requirements, the CUBE must force early offer and enable session refresh on outbound calls. The commands 'voice-class sip early-offer forced' and 'voice-class sip session refresh' are applied directly under the outbound dial-peer. These settings ensure that the initial INVITE contains SDP and that the session is periodically refreshed, complying with the provider's SIP profile requirements.

Answer

Configure 'voice-class sip profiles' with 'session refresh' and apply it to the outbound dial-peer.

Answer

Configure a voice class sip-profiles with the appropriate SIP header manipulation and apply it to the outbound dial-peer.

Answer

Configure a voice class tenant with 'early-offer forced' and apply it to the outbound dial-peer.

Question 20

A Cisco IOS XE voice gateway is configured with two dial peers: dial-peer 1 (POTS) and dial-peer 2 (VoIP). Incoming calls from the PSTN are not matching dial-peer 1 as expected. Which two dial-peer configuration elements are used to match inbound POTS calls? (Choose two.)

Accepted Answer

answer-address. Inbound POTS calls are matched using the 'incoming called-number' and 'answer-address' commands. The 'incoming called-number' matches the DNIS, while 'answer-address' matches the ANI. These are the primary criteria for associating an inbound call with a dial peer. Other commands like destination-pattern are for outbound matching, and URI-based matching is for SIP calls.

Answer

port

Answer

destination-pattern

Answer

incoming uri via