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| Section | Weight | Objectives |
|---|---|---|
| Signaling and Media Protocols | 20% | - Troubleshoot these elements of a SIP conversation
-Troubleshoot these H.323 protocol elements
|
| Cisco Unified CM Call Control Features | 20% | -Troubleshoot Call Admission Control (exclude RSVP) -Configure ILS, URI synchronization, and GDPR -Configure hunt groups -Configure call queuing -Configure time of day routing -Configure supplementary functions
|
| Mobility | 10% | -Configure Cisco Unified Communications Manager Mobility
-Troubleshoot Cisco Unified Communications Manager Mobility
|
| Cisco Unified Border Element | 15% | -Configure these Cisco Unified Border Element dial plan elements
-Troubleshoot these Cisco Unified Border Element dial plan elements
|
| CME/SRST Gateway Technologies | 10% | -Configure Cisco Unified Communications Manager Express for SIP phone registration -Configure Cisco Unified CME dial plans -Implement toll fraud prevention -Configure these advanced Cisco Unified CME features
-Configure SIP SRST gateway |
| Call Control and Dial Planning | 25% | - Configure these globalized call routing elements in Cisco Unified CommunicationsManager
-Troubleshoot these globalized call routing elements in Cisco Unified CommunicationsManager
|
Earning a passing score in such a Cisco exam will require the candidate to have a thorough understanding of all the test parts that are organized in the following exam domain:
To get through the fifth section of the exam that takes up 20%, the applicant should know the concept of Call Admission Control and must be able to troubleshoot it. However, this doesn't include RSVP technology. This section also includes various technologies such as ILS, URI, and GDPR, and the applicant should be able to skillfully configure them. Furthermore, the entrant's knowledge of hunt groups, call queuing, and the time of day routing and their configuration will be tested under such an objective. Finally, the applicant should be capable of configuring various supplementary functions such as call park, meet-me, and call pick-up.
Signaling and media protocols are the first exam domain that covers a total of 20% of all the questions. The candidate's knowledge and understanding of the elements of SIP conversation will be tested in this section. The entrant is expected to know about the concepts and skillfully describe early media, PRACK, the mid-call signaling like call transfer, session timers, and UPDATE. The applicant should also be capable of troubleshooting various H.323 protocol elements that include DTMF, call set up, and tear down. The comprehension of the media establishment and its troubleshooting will also be beneficial to get through in this portion.
The fourth portion of this exam covers 25% of the content and consists of questions regarding the Cisco Unified Communications Manager. The candidate should be capable of configuring various globalized call routing components of the Cisco Unified Communications Manager. These components include translation patterns, route patterns, transformation patterns, TEHO, SIP trunking, and standard local routing group. Moreover, the candidates must be able to troubleshoot all these globalized elements for call routing.
The third section of the exam includes 15% of all questions regarding the Cisco Unified Border Element technology. Thus, the entrant should be capable of configuring various Cisco Unified Border Element dial plan parts such as DTMF, voice translations rules & profiles, the list of codec preferences, dial peers, headers, and the manipulation of SDP with diverse SIP profiles. This domain also deals with troubleshooting all the aforementioned Cisco Unified Border Element dial plan components.
The final domain deals with Unified Communications Manager Mobility by Cisco and makes up 10% of the exam content. Therefore, the entrant should know how to configure unified mobility, extension mobility, and device mobility. Moreover, it is also essential for the candidate to be familiar with successfully troubleshooting the aforementioned mobility technologies.
The second objective takes up 10% of all test questions and deals with the Cisco Unified Communications Manager Express. What is more, the entrant needs to be aware of how to configure the aforementioned technologies for SIP phone registration. In addition to that, the applicant should be capable of configuring Cisco Unified CME dial plans along with implementing the prevention of toll fraud. Furthermore, this domain tests the candidate's capability in configuring various advanced Cisco Unified CME features that include Hunt groups, Call park, and Paging. At last, under such a category, you'll get to know more about the gateway for SIP SRST and its configuration.
The content of 300-815 exam contains six domains, and questions taken from them will pop up during the test. However, they won't appear evenly, and some sections take up a larger portion of the syllabus than others. Take a look at them below:
This domain has weight of about 25%. You are to have a general idea of troubleshooting and setting up the globalized call routing elements in CU-CM, such as the patterns for route, translation, transformation, and others.
About 10% of the questions come from this domain. It assesses one’s knowledge of how to configure CU-CME for SIP telephone registration and dial plans. Also, you should have a solid understanding of how to enforce toll fraud prevention and configure the advanced CU-CME features and SIP SRST gateway.
About 10% of the exam questions come from this topic. To tackle these items, you must understand the concepts of setting up CU-CM mobility, including unified mobility, extension mobility, and device mobility, and troubleshooting them.
This module takes about 20% of the syllabus. The subdomains include troubleshooting of the Call Admission Control and configuring ILS, URI synchronization, GDPR, hunt groups, call queuing, time of day routing, and supplementary roles.
This section takes about 15% of the content. It covers the basic skills in handling the procedures involved setting up as well as troubleshooting the CU-BE dial plan components, including DTMF, voice translation rules and profiles, codec preference list, plus signaling and media bindings.
This module covers about 20% of the test. It is required that you know how to troubleshoot the various elements of a SIP dialogue, including session timers, mid-call signaling, and more. You also need to equip yourself with a clear idea of troubleshooting certain protocol elements and media establishment.
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