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Cisco 300-815 exam is designed to assess the knowledge and skills of IT professionals who are responsible for implementing Cisco advanced call control and mobility services. 300-815 exam is part of the Cisco Certified Specialist - Collaboration Core certification and is intended for those who have experience with Cisco collaboration solutions and technologies. Passing 300-815 Exam demonstrates the candidate's ability to design, configure, and implement advanced call control and mobility services using Cisco technologies.

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Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q139-Q144):

NEW QUESTION # 139
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

A. interworking between an OOB method and RFC2833 for flow-through calls
B. interworking between h245-alpha numeric and sip-kpml
C. interworking between h245-signal and rtp-nte
D. interworking between an OOB method and RFC2833 for flow-around calls

Answer: D

NEW QUESTION # 140
Refer to the exhibit.

Refer to the exhibit An engineer deploys a Cisco Unified Border Element for SIP trunk interconnection to an ITSP The engineer performs test calls but there is no audio when the called party picks up the phone. What must the engineer do to resolve the issue?

A. Decrease the SIP max-forwards value to 16.
B. Disable SIP User Agent bind status (media).
C. Enable Check media source packets.
D. Disable SIP early-media for 180 responses with SDP

Answer: A

NEW QUESTION # 141
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?

A. ALERTING
B. CONNECT
C. PROCEEDING
D. RINGING

Answer: B

Explanation:
Section: Signaling and Media Protocols

NEW QUESTION # 142
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. Enable Force Authorization Code on the route pattern.
B. Disable Use Calling Party's External Phone Number Mask on the route pattern.
C. Enable Use Calling Party's External Phone Number Mask on the translation pattern.
D. Change the partition of the translation pattern from none to pstn_pt.

Answer: B

NEW QUESTION # 143
Which description of RTP timestamps or sequence numbers is true?

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

Answer: D

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

NEW QUESTION # 144
......

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