Implementing Cisco Unified Communications IP Telephony - Part 2

HBF IT EDUCATION SERVICES
In Bangalore

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Important information

  • Course
  • Bangalore
  • Duration:
    1 Day
Description

Important information
Venues

Where and when

Starts Location
On request
Bangalore
#70/1, 2nd Floor, Dr.Brownamma Towers 2nd Cross, Mission Road, CSI Compund, Opp: HCG Hospital, 560027, Karnataka, India
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Course programme

Meant for * Channel Associates and Resellers * Customers * EmployeesCertificateThis course is part of the following Certifications: *CCIE Voice *CCVPRequirements:You should attend Cisco IP Telephony Part 1 or have similar skills prior to applying for this course Cisco IP Telephony Part 2.Employment of CIPT1Below mentioned knowledge and skills is must for the admission to this course. * Operational understanding of converge voice and records network. * Operational understanding of the Media Gateway Control Protocol, Session Initiation Protocol, and H.323 and their implementation on Cisco Internetwork Operating System gateways* Skill to install and handle Cisco routers & switches* Ability to install and handle CUCM in a single-site atmosphereThis Course aims at:On the completion of this course students will be having the skills of: * Describe multisided employment of issues and solutions, and describe and configure required dial plan elements* Implement call processing flexibility in remote sites with the use of Survivable Remote Site Telephony, Media Gateway Control Protocol fallback, and CUCM Express in Survivable Remote Site Telephony mode* Employ call admission control to check excess subscription of the Internet Protocol Wide area network* Employ characteristic and application that are important for multisite employments* Secure a Cisco Unified Communications Internet Protocol Telephony deploymentSyllabus:Employing Cisco Unified Communications Internet Protocol Telephony Part 2 (CIPT2) v6.0 makes you to configure, a CUCM solution in a multisided atmosphere. This course pays attention on on CUCM Release 6.0, the call routing and signaling component for the Cisco Unified Communications solution. It also includes H.323 and Media Gateway Control Protocol gateway employment, the use of a Cisco Unified Border Element, and installetion of SRST, various mobility functions, and voice safety. This course involves lab activity in which you will use a dial plan for a multisided atmosphere, install survivability for remote sites during WAN malfunction and employ solution to lessen band-width necessities in the Internet Protocol WAN. You will also activate CAC and AAR, a characteristic that permits re-routing of calls over the PSTN in case of no approachable bandwidth. There are labs for the employment device mobility, extension mobility, Cisco Unified Mobility, and voice security.

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