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About the Exam

To be eligible, you have to be 18 years or older. If your age group falls between 13-17, you can take part in the test provided your parents or legal guardians support you. However, you're not eligible if you're below 13 years of age.

The exam can be taken in English and requires one to pay a registration fee of $300. The questions are expected to be provided in different forms, including MSQs, fill-in-the-blanks, testlets, and others. The time limit for this test is 1,5 hours.

300-815 is administered by Pearson VUE and can be taken at its testing centers. You can also opt for online format, and any location is fine for taking the test as long as it's quiet and private. In both cases, you will be under close supervision throughout the exam duration. The online format involves high-level supervision through the OnVUE software, so you should have it along with a webcam. Remember to provide an ID that is government-issued. If you decide to go for 300-815 at the center, you should have two valid and self-signed identification documents.

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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.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q117-Q122):

NEW QUESTION # 117
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

A. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.B. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.C. The phone button template must have the Queue Status Softkey configured.D. The maximum number of callers allowed in queue must be 10.E. A unicast music on hold audio source must be configured.

Answer: A,B


NEW QUESTION # 118
Drag and drop the commands from the bottom to the blanks in the code to implement a translation rule to allow only 11 digits to be received over a SIP trunk to a SIP provider. The Cisco UCM is currently sending calls to the Cisco Unified Border Element in E.
164 format. Not all options are used.

Answer:

Explanation:


NEW QUESTION # 119
Refer to the exhibit.

Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?

A. Point the emergency pattern directly to the PSTN gateway.B. Adjust the service parameter T302 timet to the desired value.C. Adjust the service parameter T204 timer to the desired value.D. Check the Urgent Priority check box under 9.911 pattern.

Answer: D


NEW QUESTION # 120
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

A. Allow: header if the 200 OK responseB. Contact: header of the 200 OK responseC. o= line of SDP contentD. c= line of SDP content

Answer: C

Explanation:
Section: Signaling and Media Protocols


NEW QUESTION # 121
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. T.38 fax relayB. SIP trunkC. secure SIP linesD. three-way conferenceE. transcoding

Answer: A,D

Explanation:
Section: CME/SRST Gateway Technologies
Explanation/Reference: 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 # 122
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