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Cisco 300-815 Exam is designed for professionals who want to validate their knowledge and expertise in implementing Cisco Advanced Call Control and Mobility Services. Implementing Cisco Advanced Call Control and Mobility Services certification exam is part of the Cisco Certified Network Professional (CCNP) Collaboration program and is intended for individuals who are responsible for designing, configuring, and deploying advanced collaboration solutions in enterprise environments.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q95-Q100):

NEW QUESTION # 95
Which description of RTP timestamps or sequence numbers is true?

A. Timestamps increase by the time "carrying" by a packet.
B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
C. The sequence number is used to detect losses.
D. Sequence numbers increase by four for each RTP packet transmitted.

Answer: B

Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html

NEW QUESTION # 96
Refer to the exhibit. For long-distance calls, users must prefix their dialed number with "91". The translation pattern was created to strip the 91 as the PSTN expects a 10-digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?

A. Disable Use Calling Party's External Phone Number Mask on the route pattern.
B. Change the partition of the translation pattern from none to pstn_pt.
C. Enable Force Authorization Code on the route pattern.
D. Enable Use Calling Party's External Phone Number Mask on the translation pattern.

Answer: A

NEW QUESTION # 97
Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?

A. allow-connections sip to sip
B. voice register global
C. voice register dn
D. voice service voip

Answer: D

Explanation:
You have to enable the registrar service before you can provide the address in voice register global voice service voip allow-connections sip to sip
!--- Enable SIP to SIP calls.
sip
registrar server expires max 1200 min 300
!--- Enable Cisco IOS SIP registrar.

NEW QUESTION # 98
An administrator is asked to configure egress call routing by applying globalization and localization on Cisco UCM.
How should this be accomplished?

A. Localize the calling and called numbers to E.164 format and globalize the called number in the gateway.
B. Globalize the calling and called numbers to E.164 format and localize the called number in the gateway.
C. Localize the calling and called numbers to PSTN format and globalize the calling and called numbers in the gateway.
D. Globalize the calling and called numbers to PSTN format and localize the calling number in the gateway.

Answer: B

NEW QUESTION # 99
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF.
Which configuration should be added to the outgoing dial peer to resolve this issue?

A. dtmf-relay sip-kpml
B. dtmf-relay cisco-rtp
C. dtmf-relay h245-signal
D. dtmf-relay rtp-nte

Answer: A

NEW QUESTION # 100
......

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Mike Fisher Mike Fisher

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