Voice Gateways And Session Border Controllers practice questions

Dial-peer none

Why it fails: Dial-peer none does not exist.

Dial-peer 1

Why it fails: This is not the system default.

Dial-peer 100

Why it fails: This is not a default dial-peer.

dtmf-relay all

Why it fails: This command does not exist.

dtmf-relay rfc2833

Why it fails: This only supports one method.

dtmf-relay h245-alphanumeric

Why it fails: This is used for H.323, not SIP.

dtmf-relay

Why it fails: This affects tone signaling, not session duration.

voice-class codec

Why it fails: This governs media negotiation, not session duration.

timer connect

Why it fails: This timer handles call connection timeout, not media refreshes.

dial-peer voice

Why it fails: Dial-peers route calls but do not modify individual headers.

voice class codec

Why it fails: This manages media codecs.

voice translation-rule

Why it fails: Translation rules are for digit manipulation, not SIP headers.

session-protocol sipv2

Why it fails: This is not a valid command syntax for dial-peer protocol configuration.

signaling-protocol sip

Why it fails: This is not a valid command syntax.

voice-class protocol sip

Why it fails: This command is used for voice-class definitions, not dial-peer protocol binding.

Partitioned route patterns

Why it fails: This is a CUCM feature, not IOS XE gateway.

Dial-peer groups

Why it fails: Dial-peer groups are for hunting, not tenancy.

SIP headers

Why it fails: Headers do not provide network isolation.

Voice class tenant

Why it fails: This is not a valid command.

no signaling h323

Why it fails: This is not the correct syntax.

disable h323

Why it fails: This is not a valid command.

shutdown h323

Why it fails: This is not a valid command.

debug calls

Why it fails: This is for real-time tracing, not a status summary.

show call history

Why it fails: This shows past calls, not active ones.

show voice status

Why it fails: This is not a valid command.

tls-srtp

Why it fails: This is not the correct command syntax.

encryption srtp

Why it fails: This is not the correct command syntax.

secure media

Why it fails: This is not the correct command syntax.

dial-peer voice

Why it fails: This is the object where profiles are applied, not the command to configure the profile itself.

sip-ua

Why it fails: Used for registration, not profile definition.

voice class codec

Why it fails: This is for media negotiation, not registration.

dial-peer voice

Why it fails: Dial-peers handle routing, not registration.

voice translation-rule

Why it fails: This is for digit manipulation.

debug voice translation

Why it fails: Only useful if translation rules are the suspect.

debug voice class

Why it fails: This is not a valid debug command.

show sip-profile status

Why it fails: This does not provide runtime troubleshooting.

Configure route group

Why it fails: This is a CUCM configuration.

Configure phone buttons

Why it fails: Not part of gateway trunking.

destination-pattern

Why it fails: Used for outgoing, not incoming matches.

session target

Why it fails: Used for outgoing, not incoming matches.

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

Why it fails: The 'options-ping' command under sip-ua is used to send OPTIONS pings to a remote target for monitoring, not to respond to inbound OPTIONS. It would not help the CUBE answer the ITSP's OPTIONS. The CUBE needs to be enabled for SIP-to-SIP connections and have a dial peer that matches the incoming OPTIONS.

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

Why it fails: The 'options' command under voice service voip is not used to enable OPTIONS responses. It is used to configure OPTIONS ping parameters for outbound monitoring. Applying it to an incoming dial peer would not make the CUBE answer inbound OPTIONS. The correct approach involves the 'allow-connections' and 'sip-profiles' or specific 'options' handling for inbound.

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

Why it fails: SIP profiles are used for header manipulation, not for enabling OPTIONS responses. Modifying the Allow header does not cause the CUBE to answer an OPTIONS request. The CUBE must be configured to accept and respond to OPTIONS messages through global SIP configuration and proper dial-peer matching.

max-calls 100

Why it fails: There is no command named max-calls in Cisco IOS XE voice configuration. The correct command to limit concurrent calls is max-conn. Using an invalid command would result in a configuration error and would not enforce the desired limit, making it an incorrect choice for this scenario.

call threshold global 100

Why it fails: The call threshold global command is used to set thresholds for generating SNMP traps or triggering actions based on call volume, not to enforce a hard limit on concurrent calls. It monitors call activity but does not restrict the number of calls. Therefore, it does not satisfy the requirement to limit concurrent calls to 100.

sip max-calls 100

Why it fails: The command sip max-calls does not exist in Cisco IOS XE. While the sip keyword is used in many commands, this specific syntax is invalid. The correct command to limit concurrent SIP connections is max-conn under voice service voip sip. Thus, this option is incorrect and would not achieve the goal.

sip-contact user-param

Why it fails: The command sip-contact user-param does not exist in Cisco IOS XE. While sip-contact might appear in other contexts, it is not used to add user parameters to the Contact header. The correct command is contact-header user-param. Therefore, this option is invalid and would not fulfill the requirement.

user-param contact

Why it fails: The command user-param contact is not valid in Cisco IOS XE SIP UA configuration. The correct syntax places contact-header first, followed by the user-param keyword. This option would result in a syntax error and would not modify the Contact header as needed, making it incorrect for this scenario.

contact-user-param

Why it fails: There is no command contact-user-param in Cisco IOS XE SIP UA configuration. While custom parameters can be added to the Contact header, the correct approach is to use the contact-header command with the user-param keyword. This option is invalid and would not achieve the desired customization.

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

Why it fails: The 'move' action would move the P-Asserted-Identity header to the From header, deleting the original P-Asserted-Identity header. While this might populate the From header, it would remove the P-Asserted-Identity header, which may be required by other elements. The requirement is to copy the value, not move it, to preserve the original header.

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

Why it fails: This command applies to response messages, not requests. The scenario requires modification of outgoing INVITE requests, so a response manipulation would not affect the initial call setup. Additionally, the source and destination are reversed, so even if applied to requests, it would not produce the desired outcome.

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

Why it fails: This command attempts to copy the P-Asserted-Identity header into the From header, but the syntax is incorrect. The 'copy' action requires a source and destination, and the correct order is to copy from the source to the destination. Here, the source and destination are reversed, so it would not achieve the desired result and may cause errors.

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

Why it fails: SIP profiles are for header manipulation, not for enabling session refresh. The 'session refresh' parameter is not a valid SIP profile command; it is configured directly under the dial-peer or via a voice class. Therefore, this approach would not enable session refresh and is incorrect for the scenario.

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

Why it fails: SIP profiles are used for header manipulation, such as adding or modifying headers, but they do not control early offer or session refresh behavior. Applying a SIP profile to the outbound dial-peer would not enforce early offer or session refresh; those are controlled by other parameters, so this step alone is insufficient.

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

Why it fails: A voice class tenant is used for multi-tenant CUBE configurations, where different tenants have separate SIP settings. While it can contain early-offer settings, it is not the direct method to force early offer on a specific dial-peer. The scenario does not mention multi-tenancy, so using a tenant is unnecessarily complex and not required.

port

Why it fails: The 'port' command under a voice-port or dial peer specifies the physical voice port to use for outbound calls. It does not participate in inbound dial-peer matching. Inbound POTS calls are matched based on called number, calling number, or other criteria, not by the port configuration on the dial peer.

destination-pattern

Why it fails: The 'destination-pattern' command is used to match outbound calls, not inbound calls. For outbound dial peers, it specifies the destination number pattern to which the call is routed. Inbound matching uses different commands such as 'incoming called-number' or 'answer-address'. Therefore, this does not help match inbound POTS calls.

incoming uri via

Why it fails: The 'incoming uri via' command is used for matching inbound SIP calls based on the Via header. It is not applicable to POTS calls, which do not use SIP URIs. POTS calls are matched using traditional dial-peer matching commands like 'incoming called-number' and 'answer-address'.