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Certification Cisco 300-815 Questions | 300-815 Test Testking

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Cisco 300-815 Exam Syllabus Topics:

Topic
Details

Topic 1

Configure These Globalized Call Routing Elements In Cisco Unified Communications Manager

Topic 2

Troubleshoot Cisco Unified Communications Manager Mobility
Troubleshoot These H.323 Protocol Elements

Topic 3

Troubleshoot Call Admission Control (Exclude RSVP)
Call Control And Dial Planning

Topic 4

Mid-Call Signaling (Hold
Resume, Call Transfer, Conferencing)
Configure SIP SRST Gateway

Topic 5

Configure These Cisco Unified Border Element Dial Plan Elements
Configure Cisco Unified CME Dial Plans

Topic 6

Configure Cisco Unified Communications Manager Express For SIP Phone Registration
Cisco Unified Border Element

Topic 7

Configure These Advanced Cisco Unified CME Features
Voice Translation Rules And Profiles

Topic 8

Configure Cisco Unified Communications Manager Mobility
Signaling And Media Protocols

Topic 9

Header And SDP Manipulation With SIP Profiles
Voice Translation Rules And Profiles

Topic 10

Configure ILS, URI Synchronization, And GDPR
CME
SRST Gateway Technologies

Topic 11

Troubleshoot These Cisco Unified Border Element Dial Plan Elements
Implement Toll Fraud Prevention

Topic 12

Troubleshoot These Globalized Call Routing Elements In Cisco Unified Communications Manager

Topic 13

Configure Supplementary Functions
Troubleshoot These Elements Of A SIP Conversation

>> Certification Cisco 300-815 Questions <<

300-815 Test Preparation: Implementing Cisco Advanced Call Control and Mobility Services & 300-815 Exam Lab Questions

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Candidates who pass the Cisco 300-815 Exam will be able to demonstrate their knowledge and skills in advanced call control and mobility services. They will be able to design, implement, and troubleshoot these services on Cisco collaboration solutions. They will also be able to configure and manage call routing, dial plans, media resources, and call admission control.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q181-Q186):

NEW QUESTION # 181
Users of a Cisco UCME report that calls lose audio after they are forwarded to some users. Calling point-to-point seems to function fine, but depending on the type of phone, some calls lose audio when forwarded. Looking at the trace header that shows:
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER What is missing and is a reason that the SRTP stream stops working?

A. UPDATE
B. FORWARD
C. EARLY-OFFER
D. REDIRECT

Answer: A

NEW QUESTION # 182
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?

A. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
B. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
C. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
D. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.

Answer: C

Explanation:
Section: Call Control and Dial Planning

NEW QUESTION # 183
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

A. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
B. configuration for a single SIP phone
C. configuration for SIP registrar service
D. configuration items common for all SIP phones

Answer: B

NEW QUESTION # 184
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. voice register global
B. voice register dn
C. voice service voip
D. allow-connections sip to sip

Answer: A

Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/99946-cme-sip-guide.html

NEW QUESTION # 185
In Cisco UCM, which tool is used to check SIP traces?

A. MTP
B. OS Administration Page
C. RTMT
D. CCSIP

Answer: C

NEW QUESTION # 186
......

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Walt Scott Walt Scott

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