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Questions Cisco 300-815 Exam | 300-815 Valid Exam Testking

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Introduction to Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam

Implementing Cisco Advanced Call Control and Mobility Services v1.0 CLACCM 300-815 is a 90-minute exam associated with the CCNP Collaboration Certification. This exam tests a candidate's knowledge of advanced call control and mobility services, including signaling and media protocols, CME/SRST gateway technologies, Cisco Unified Board Element, call control and dial planning, Cisco Unified CM Call Control, and mobility. The course, Implementing Cisco Advanced Call Control and Mobility Services, helps candidates to prepare for this exam. In addition, you use the Dynamic Host Configuration Protocol (DHCP) to fix faults, network security, and VLAN settings, as well as many other common network concerns. The chance to acquire new concepts and to improve understanding via a range of integrated lab settings is an essential element of this course. You may resolve your problems by using 300-815 exam dumps and different laboratory conditions in a wide range of case studies.

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Cisco 300-815 exam is a part of the CCNP Collaboration certification and is designed to test the skills and knowledge of IT professionals in implementing Cisco advanced call control and mobility services. 300-815 exam is intended for professionals who work with Cisco Unified Communications Manager (CUCM), Cisco Unity Connection, and Cisco IM and Presence.

To prepare for the Cisco 300-815 Exam, candidates can take advantage of various study materials, such as official Cisco training courses, practice exams, and study guides. These resources can help candidates gain a deeper understanding of the exam topics and improve their chances of passing the exam on the first attempt.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q146-Q151):

NEW QUESTION # 146
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)

A. transcoding
B. SIP trunk
C. secure SIP lines
D. three-way conference
E. T.38 fax relay

Answer: D,E

Explanation:
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
Hold/Resume
Call Forward
Call Transfer
Three-way Conference (with BIB conferencing only)
Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/ srst_sip_isr4000.html

NEW QUESTION # 147
Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?

A. in Cisco Unified CM Enterprise Parameters
B. in the queuing configuration
C. in Cisco Unified CM Service Parameters
D. in the telephony service configuration

Answer: B

NEW QUESTION # 148
Refer to the exhibit.

Refer to the exhibit. Calls from users to the PSTN in an organization get disconnected with a 408 Request Timeout when the called party is unavailable to pick up the call. Which solution must be used to resolve this challenge?

A. Choose "Send PRACK" if 1xx contains SDP in the SIP profile.
B. Configure midcall-signaling passthru media-change
C. Choose "Send PRACK for all 1xx Messages" in the SIP profile.
D. Configure midcall-signaling preserve-codec.

Answer: B

NEW QUESTION # 149
What is first preference condition matched in a SIP-enabled incoming dial peer?

A. answer-address
B. target carrier-id
C. incoming uri
D. incoming called-number

Answer: C

Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8

NEW QUESTION # 150
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses
H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls.
Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

A. H.245 Open Logical Channel
B. H.245 Terminal Capability Set
C. H.225 Connect
D. H.245 Open Logical Channel Ack

Answer: D

Explanation:
H245 Open Logical Channel contains only RTCP port, H245 Open Logical Channel ACk will have both RTP and RTP Port.
https://community.cisco.com/t5/collaboration-voice-and-video/h323-call-flow/ta-p/3160014

NEW QUESTION # 151
......

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