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Exam Code: 300-075
Exam Name: CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
Tactics to 300 075 ciptv2
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Q1. Refer to the exhibit.
The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number.
Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit.
Which partition should be configured in the AAR CSS applied at the phones?
A. PSTN partition
B. LD partition
C. The HQ AAR CSS must include a partition assigned to route pattern 91408XXXXXXX. The BR1 AAR CSS must include a partition assigned to route pattern 91650XXXXXXX.
D. AAR CSS must contain translation pattern 9.1[2-9]XX[2-9]XXXXXX for each site that must be globalized. Otherwise the called numbers will not be localized at the egress gateway.
Q2. What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
Q3. Which method can be used to address variable-length dial plans?
A. Overlap sending and receiving.
B. Add a prefix for all calls that are longer than 10-digits long
C. Use nested translation patterns to eliminate inter-digit timeout
D. Use the @macro on the route pattern
E. Use MGCP gateways, which support variable-length dial plans
Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link:
Q4. Which command is needed to utilize local dial peers on an MGCP-controlled ISR during an SRST failover?
A. ccm-manager fallback-mgcp
D. isdn overlap-receiving
Q5. Refer to the exhibit.
The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number. Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. Which statement is true?
A. The AAR group system must be configured on the device configuration of the phones.
B. The AAR group system must be configured on the line configuration of the phones.
C. The single AAR group system cannot be used. A second AAR group must be configured in order to have source and destination AAR groups.
D. The AAR group system must be configured under the AAR service parameters.
Q6. Which statement about enrollment in the IP telephony PKI is true? (Source. Understanding Cisco IP Telephony Authentication and Encryption Fundamentals)
A. CAPF enrollment supports the use of authentication strings.
B. The CAPF itself has to enroll with the Cisco CTL client.
C. LSCs are issued by the Cisco CTL client or by the CAPF.
D. MICs are issued by the CAPF itself or by an external CA.
Incorrect Answer: B, C, D
The CAPF enrollment process is as follows:
1. The IP phone generates its public and private key pairs.
2. The IP phone downloads the certicate of the CAPF and uses it to establish a TLS session with the CAPF.
3. The IP phone enrolls with the CAPF, sending its identity, its public key, and an optional authentication string.
4. The CAPF issues a certicate for the IP phone signed with its private key.
5. The CAPF sends the signed certicate to the IP phone.
Q7. In a distributed call processing network with locations-based CAC, calls are routed to and from intercluster trunks. Which trunk type is implemented in this network?
A. intercluster trunk with gatekeeper control
B. intercluster trunk without gatekeeper control
C. SIP trunk
D. h225 trunk
Q8. What is the difference between an H.323 gateway and a SIP gateway?
A. An H.323 gateway requires that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. The H.323 gateway can be added in Cisco Unified Communications Manager under gateway type as H.323 Gateway. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk.
C. A SIP gateway requires a call agent for PSTN calls to be placed and received. An H.323 gateway does not require a call agent for PSTN calls to be placed and received.
D. An H.323 gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The H.323 gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The SIP gateway must be configured in Cisco Unified Communications Manager using the domain name.
Q9. Which statement is true when device mobility mode is enabled or disabled in the Phone
A. The device mobility mode phone settings take precedence over the service parameter settings.
B. The service parameter settings take precedence over the device mobility mode phone settings.
C. The combined service parameter settings and the device mobility mode phone settings will be used.
D. The default settings will be used due to the conflicts.
Q10. Which of the following are two functions that ensure that the telephony capabilities stay operational in the remote location Cisco Unified SRST router? (Choose two)
A. Automatically detecting a failure in the network.
B. Initiating a process to provide call-processing backup redundancy.
C. Notifying the administrator of an issue for manual intervention.
D. Proactively repairing issues in the voice network.
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