CLCOR Voice Gateways And Session Border Controllers Practice Test 2 — 15 Questions

Question 1

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?

Choose an answer to begin — your selection is scored in the full session.

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 2

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 3

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 4

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 5

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

Question 6

A network engineer is configuring a Cisco CUBE to support early offer for SIP calls. Which two statements are true regarding early offer configuration on CUBE? (Choose two.)

Accepted Answer

Early offer can be configured on a per-dial-peer basis using the 'voice-class sip early-offer forced' command.. Early offer on CUBE can be forced globally with the early-offer forced command under voice service voip sip, or per dial peer using voice-class sip early-offer forced. Both methods are valid and serve different scopes. The global setting applies to all SIP calls, while the per-dial-peer setting allows selective application. These are the two true statements regarding early offer configuration.

Answer

Early offer is only supported for H.323-to-SIP interworking, not for SIP-to-SIP calls.

Answer

Early offer requires the CUBE to use delayed offer for incoming calls.

Answer

The 'early-offer forced' command must be applied on the inbound dial peer.

Question 7

You are configuring a Cisco IOS XE voice gateway to connect to a SIP provider. The provider requires that the gateway authenticate using SIP digest authentication for outbound calls. You have configured the credentials under sip-ua. Which additional command is needed to ensure that the gateway challenges incoming calls from the provider?

Accepted Answer

authentication mode digest under the inbound dial peer. To challenge incoming calls, you must enable authentication on the inbound dial peer. The authentication mode digest command under dial-peer configuration causes the gateway to respond with a 401 Unauthorized to incoming INVITEs, requiring the provider to authenticate. This is separate from the credentials used for outbound authentication, which are configured under sip-ua.

Answer

registrar ipv4:10.1.1.1

Answer

authentication username user password pass

Answer

credentials username user password pass

Question 8

A Cisco Unified Border Element (CUBE) is deployed between an enterprise Cisco Unified Communications Manager and an ITSP. The ITSP requires that all outbound calls use the SIP domain 'provider.com' in the From header. Which CUBE configuration ensures this requirement?

Accepted Answer

request INVITE sip-header From modify "sip:(.*)@.*" "sip:\1@provider.com". The correct configuration uses a SIP profile with a 'request INVITE' rule to modify the From header, replacing the domain with 'provider.com' while preserving the user part. This ensures that all outbound INVITEs carry the required domain. Other options either lack the necessary context or apply to the wrong message type.

Answer

response 200 sip-header From modify "sip:.*" "sip:user@provider.com"

Answer

voice class sip-profiles 10

Answer

sip-header From modify "sip:.*" "sip:user@provider.com"

Question 9

A voice engineer is configuring a Cisco IOS XE gateway to register with a SIP provider. The provider requires that the gateway authenticate using credentials. Which command is used to specify the authentication username and password for the SIP registrar?

Accepted Answer

authentication username password . To register a Cisco IOS XE gateway with a SIP provider that requires authentication, you must configure the credentials under the 'sip-ua' configuration mode using the 'authentication username password ' command. This provides the username and password that the gateway uses to authenticate with the registrar. The registrar command only specifies the registrar's address, not the credentials.

Answer

registrar ipv4:<ip_address>

Answer

credentials username password

Answer

sip-ua authentication

Question 10

A Cisco Unified Border Element (CUBE) is configured for SIP-to-SIP interworking. The provider sends calls with the P-Asserted-Identity header containing the calling party number. You need to copy the number from the P-Asserted-Identity header into the From header for calls sent to Unified CM. Which configuration element should you use?

Accepted Answer

voice class sip-profiles with a sip-header P-Asserted-Identity copy rule. To copy a value from one SIP header to another, you must use a voice class sip-profiles with a sip-header command that specifies the source header and the destination header. The profile is then applied to the outbound dial peer so that the From header is rewritten with the value from the P-Asserted-Identity header before the INVITE is sent to Unified CM. This is the standard method on CUBE.

Answer

voice service voip with the sip-header P-Asserted-Identity copy command

Answer

voice class uri with a pattern to match the P-Asserted-Identity header

Answer

translation-rule to map the P-Asserted-Identity to the From header

Question 11

A Cisco Unified Border Element (CUBE) is experiencing one-way audio on calls between two SIP trunks. You have verified that the signaling is correct and both sides are using the same codec. Which CUBE feature should you check first to resolve the issue?

Accepted Answer

Media flow-through. One-way audio on CUBE often stems from media flow-through misconfiguration, such as missing or inconsistent commands on dial peers. Since signaling and codec are correct, the issue is likely in how CUBE handles the RTP stream. Verifying that media flow-through is enabled on both inbound and outbound dial peers is the logical first step.

Answer

Media flow-around

Answer

Codec preference list

Answer

SIP normalization and transparency

Question 12

A CUBE is configured to connect to a SIP provider that uses a non-standard SIP port (not 5060) for signaling. The CUBE must send outbound INVITEs to this provider on port 5080. Which configuration is required on the outbound dial-peer to achieve this?

Accepted Answer

session target ipv4:192.0.2.1:5080. For a SIP dial-peer, the destination IP address and port are configured using the 'session target ipv4::' command. This directs outbound SIP signaling to the specified IP and port. In this scenario, the provider uses port 5080, so the session target must include that port. Other commands like destination-pattern or outbound-proxy serve different purposes and would not correctly direct the INVITE to the non-standard port.

Answer

voice-class sip url sip:192.0.2.1:5080

Answer

destination-pattern 5080

Answer

voice-class sip outbound-proxy ipv4:192.0.2.1:5080

Question 13

A Cisco Unified Border Element (CUBE) is deployed between Cisco Unified Communications Manager and a SIP provider. The provider sends calls with the P-Early-Media header to indicate early media. You need to ensure that CUBE forwards the P-Early-Media header to Unified CM. Which two configuration steps are required? (Choose two.)

Accepted Answer

Define a voice class sip-profiles with a rule to pass the P-Early-Media header.. To forward the P-Early-Media header, you need to create a SIP profile that allows the header to be passed and apply it to the outbound dial peer towards Unified CM. By default, CUBE may strip this header, so explicit configuration is necessary. The SIP profile ensures the header is included in the outgoing INVITE.

Answer

Enable the pass-through of unknown headers under voice service voip.

Answer

Configure the dial peer to use the sip-header pass-through command.

Answer

Set the early-media option under the SIP user agent configuration.

Question 14

A Cisco IOS XE voice gateway is configured with two dial peers: dial-peer 10 voip (destination-pattern 9T) and dial-peer 20 pots (destination-pattern 9T). An engineer wants calls matching 9T to be routed out the POTS dial peer only if the VoIP dial peer is unavailable. Which configuration change ensures this behavior?

Accepted Answer

Assign a higher preference to dial-peer 20 pots using the preference command.. Dial-peer preference controls the order of dial-peer selection when multiple dial peers match the same destination pattern. Lower preference values are tried first. To make the POTS dial peer a backup, assign it a higher preference value than the VoIP dial peer. This way, the VoIP dial peer is attempted first, and if it fails (e.g., due to no response), the gateway will then try the POTS dial peer.

Answer

Configure the huntstop command on dial-peer 20 pots.

Answer

Assign a lower preference to dial-peer 20 pots using the preference command.

Answer

Configure the huntstop command on dial-peer 10 voip.

Question 15

A Cisco Unified Border Element (CUBE) is deployed for SIP-to-SIP interworking between Cisco Unified Communications Manager (CUCM) and a SIP provider. The provider requires that all outbound calls include a specific custom header, X-Provider-ID, with a value of 12345. The CUBE must also remove any incoming X-Provider-ID header from the provider to prevent loops. Which two SIP profile configurations on CUBE achieve these requirements? (Choose two.)

Accepted Answer

request INVITE sip-header X-Provider-ID remove. To meet the provider's requirements, the CUBE must add the X-Provider-ID header to outbound INVITEs and remove any incoming X-Provider-ID header from the provider. The SIP profile commands request INVITE sip-header X-Provider-ID add header "X-Provider-ID: 12345" and request INVITE sip-header X-Provider-ID remove accomplish these tasks. These manipulations ensure proper signaling and prevent loops.

Answer

response INVITE sip-header X-Provider-ID remove

Answer

response INVITE sip-header X-Provider-ID add header "X-Provider-ID: 12345"

Answer

request INVITE sip-header X-Provider-ID copy-to X-Provider-ID