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Exam Code: 300-075
Exam Name: CIPTV2 Implementing Cisco IP Telephony and Video, Part 2
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Q11. Which remote-site redundancy technology fails over to POTS dial peers from the Cisco Unified Communications Manager dial plan during a WAN failure?
A. MGCP fallback
B. H.323 fallback
C. SCCP fallback
D. SIP fallback
Q12. Refer to the exhibit.
If an IP phone in San Jose roams to New York, which two IP phone settings will be modified by Device Mobility so that the phone can place and receive calls in New York? (Choose two.)
A. The physical locations are not different, so the configuration of the phone is not modified.
B. The physical locations are different, so the roaming-sensitive parameters of the roaming device pool are applied.
C. The device mobility groups are the same, so the Device Mobility-related settings are applied in addition to the roaming-sensitive parameters.
D. The Device Mobility information is associated with one or more device pools other than the home device pool of the phone, so one of the associated device pools is chosen based on a round-robin load-sharing algorithm.
E. The Device Mobility information is associated with the home device pool of the phone, so the phone is considered to be in its home location. Device Mobility will reconfigure the roaming-sensitive settings of the phone.
Q13. Which technologies provide remote-site redundancy for Cisco IP Phones during a WAN failure?
A. SRST and MGCP fallback
B. SRST and TEHO
C. TEHO and MGCP fallback
D. SRST and AAR
Q14. On which Cisco Unified Communications Manager configuration parameter does the CODEC that a Cisco IP Phone uses for a call depend?
A. enterprise parameters
B. media resources
C. physical location
Q15. You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2-Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?
Q16. Refer to the following exhibit.
The MGCP gateway has the following configurations:
called party transformation CSS HQ_cld_pty CSS (partition=HQ cld_pty.Pt) call.ng party transformation CSS HQ_clng_pty CSS (partition=HQ_clng_pty Pt)
All translation patterns have the check box "Use Calling Party's External Phone Number Mask" enabled.
When the IP phone at extension 3001 places a call to 9011 49403021 56001# what is the resulting called and calling number that is sent to the PSTN?
A. The called number is 01 1 49403021 56001. The calling number will be 5553001 and number type set to subscriber.
B. The called number is 011 49403021 56001. The calling number will be 5215553001 and number type set to national.
C. The called number is 4940302156001 with number type set to international. The calling number will be 5215553001 and number type set to national.
D. The called number is +49403021 56001 with number type set to international. The calling number will be 5215553001 and number type set to subscriber.
Incorrect Answer: B, C, D Check the check box "Use Calling Party's External Phone Number Mask" if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls. You may also configure an External Phone Number Mask on all phone devices. Link: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00805 b6f33.shtml
Q17. Which two configurations can you perform to allow Cisco Unified Communications Manager SIP trunks to send an offer in the INVITE? (Choose two.)
A. Enable the Media Termination Point Required option on the SIP trunk.
B. Enable the Early Offer Support for Voice and Video Calls option on the SIP profile.
C. Select the Display IE Delivery check box in the gateway configuration.
D. Select the Enable Inbound FastStart check box on the Cisco Unified Communications Manager servers.
E. Select the SRTP Allowed check box on the SIP trunk.
F. Execute the isdn switch-type primary-ni command globally.
Q18. Which option describes a function of SIP preconditions?
A. SIP preconditions enable end-to-end RSVP over an SIP trunk.
B. SIP preconditions enable RSVP between Cisco IP Phones.
C. SIP preconditions can be enabled in a gatekeeper.
D. SIP preconditions enable end-to-end RSVP for calls through the PSTN.
Q19. In a cluster-wide deployment, what is the maximum number of Service Advertisement Framework forwarders to which the Cisco Unified Communications Manager can connect?
F. as many as are configured
Q20. This is the configuration on the voice gateway:
max-dn 60 preference 0
srst mode auto-provision all
srst dn line-mode dual
srst dn template 3 srst ephone description
srst fallback auto-provision phone
srst ephone template 5
Which ephone-dn would be expected upon activation of SRST?
A. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones
B. ephone-dn 1 dual-linenumber 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones
C. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 5This DN is learned from srst fallback ephones
D. ephone-dn 1number 7001description 7001name 7001ephone-dn-template 3This DN is learned from srst fallback ephones
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