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

  • Exam Code: 300-815
  • Exam Name: Implementing Cisco Advanced Call Control and Mobility Services
  • Updated: Nov 02, 2026   Q&As: 243 Questions and Answers

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About Cisco 300-815 Exam

Different versions for different study habits — PDF, test engine, online engine. Choose yours: the Cisco Implementing Cisco Advanced Call Control and Mobility Services material at DumpsTests, 243 practice questions for the 300-815 exam in 2026.

Cisco 300-815 Exam Syllabus Topics:
SectionObjectives
Cisco Expressway and Remote Access- Mobile and Remote Access (MRA)
  • 1. Expressway-C and Expressway-E architecture
    • 2. Traversal and firewall considerations
      Cisco Unified Communications Manager (CUCM) Call Control- Dial plan design and implementation
      • 1. Numbering plans and routing patterns
        • 2. Translation patterns and normalization
          - Call routing and call admission control
          • 1. CAC (Call Admission Control) concepts
            • 2. Call routing logic in CUCM
              Advanced Call Control Features- Quality of Service (QoS)
              • 1. Voice traffic prioritization and DSCP marking
                - Media resources and call treatment
                • 1. Music on Hold (MoH) and conferencing resources
                  VoIP Protocols and SIP Signaling- SIP trunking and interworking
                  • 1. SIP normalization and manipulation
                    • 2. SIP message flows and troubleshooting
                      - H.323 and legacy protocol interaction
                      • 1. Interoperability considerations
                        Mobility Services- Extension Mobility
                        • 1. User device profiles and roaming
                          - Cisco Unified Mobility features
                          • 1. Single Number Reach (SNR)
                            • 2. Mobile Connect and remote destinations

                              Cisco 300-815 Exam: Partner Answers

                              Yes:

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                              The Cisco Implementing Cisco Advanced Call Control and Mobility Services blueprint spans 5 domains — including VoIP Protocols and SIP Signaling, Mobility Services, Cisco Expressway and Remote Access. Know your strength and shortcoming per domain; the complete outline above lists every subtopic.

                              90 minutes for 55–65 (varies) questions. The DumpsTests test engine simulates the actual test's atmosphere, so you get used to real conditions in advance.

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                              USD 300 per attempt, 750-850 (Cisco scaled scoring, passing typically around 825/1000) to pass. A failed attempt is a loss of time and money — prepare steadily with the 243 practice questions for the 300-815 exam at DumpsTests.

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                              The Cisco Implementing Cisco Advanced Call Control and Mobility Services is Cisco's certification exam for CCNP Collaboration, at the Professional level. Passing it is a new start — toward better salary, decent work, and promotion chances. Related credentials include CCNP Collaboration, CCIE Collaboration, Cisco Certified Network Professional (CCNP), Implementing Cisco Enterprise Network Core Technologies (ENCOR 350-401).

                              No formal prerequisite, but Cisco recommends CCNA-level knowledge or completion of CCNP Collaboration core exam (350-401 ENCOR). Eligibility rules change over time, so verify the current requirements on the official page (official 300-815 exam page) before registering.

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                              Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions:
                              Question #1

                              A user requests a feature to send an active call to the mobile phone number on the physical phone. As an administrator, what should be configured in the Cisco UCM to accomplish this?

                              • A. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template
                              • B. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template.
                              • C. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.
                              • D. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.
                              Reveal Solution  Discussion  0

                              Correct Answer: D  🗳️

                              Question #2

                              Refer to the exhibit.

                              Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?

                              • A. No DTMF is negotiated.
                              • B. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
                              • C. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
                              • D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
                              Reveal Solution  Discussion  0

                              Correct Answer: A  🗳️

                              Question #3

                              Refer to the exhibit.

                              Refer to the exhibit. A company is using Microsoft Teams with Cisco Unified Border Element integration, but the administrator sees a one-way audio issue with Microsoft Teams. The administrator must modify the SIP profile to send the proper information on the SDP for IP address for media to match the internal and external interface. Which set of commands resolves the issue?

                              • A. voice class sip-profiles 1
                                request INVITE sdp-header Session-Owner modify "27\.0\.0" "27.3.0"
                              • B. voice class sip-profiles 1
                                request INVITE sdp-header Connection add "2\.76\.1" "2.78.1"
                              • C. voice class sip-profiles 1
                                request INVITE sdp-header Audio-Attribute modify "2\.76\.1" "2.71.1"
                              • D. voice class sip-profiles 1
                                request INVITE sdp-header Connection-Info modify "2\.76\.1" "2.78.1"
                              Reveal Solution  Discussion  0

                              Correct Answer: B  🗳️

                              Question #4

                              Refer to the exhibit.

                              A collaboration engineer is troubleshooting an issue where the PSTN calls of a Cisco UCM IP phone user are not reaching the PSTN gateway. Which action resolves the issue?

                              • A. Ensure that the user's phone is assigned to a device pool with the correct local route group settings.
                              • B. Change the calling search space of The user's line or device.
                              • C. Change the "Call Classification" to "OnNet" on the route pattern.
                              • D. Deselect "Block this pattern" on the route pattern.
                              Reveal Solution  Discussion  0

                              Correct Answer: A  🗳️

                              Question #5

                              Refer to the exhibit.

                              A no-way audio situation was experienced after the calling party on an outbound call placed the call on hold and then resumed from hold. Which action will resolve the issue?

                              • A. Assign an MTP to the SIP trunk.
                              • B. Upgrade the Cisco Unified Border Element device.
                              • C. Restart the SIP trunk on Cisco UCM.
                              • D. Upgrade the Cisco Unified Border Element to 15.2.
                              Reveal Solution  Discussion  0

                              Correct Answer: D  🗳️

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