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New 300-815 Test Tips, Interactive 300-815 Practice Exam
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2026 Latest TestValid 300-815 PDF Dumps and 300-815 Exam Engine Free Share: https://drive.google.com/open?id=1mmr1KHzDx7LklJ1ENV6qo4icc0emgf0V
Earning the Implementing Cisco Advanced Call Control and Mobility Services (300-815) exam credential is undoubtedly a big achievement. No matter how hard the Implementing Cisco Advanced Call Control and Mobility Services (300-815) test of this certification is, it serves the important purpose to validate skills in the Cisco industry. Once you crack the Implementing Cisco Advanced Call Control and Mobility Services (300-815) exam, a whole new career scope opens up for you. Candidates for the Implementing Cisco Advanced Call Control and Mobility Services (300-815) exam dumps usually don't have enough time to study for the test. To prepare successfully in a short time, you need a trusted platform of real and updated Implementing Cisco Advanced Call Control and Mobility Services (300-815) exam dumps.
Cisco 300-815 Exam consists of 60-70 multiple-choice and multiple-answer questions that are designed to test your knowledge and skills in implementing and troubleshooting advanced call control and mobility services. 300-815 exam duration is 90 minutes, and candidates must achieve a score of at least 825 out of 1000 points to pass. 300-815 exam can be taken in English and Japanese languages and is available globally through Pearson VUE testing centers.
Cisco 300-815 exam covers a wide range of topics related to advanced call control and mobility services. Candidates will be tested on their ability to configure and troubleshoot CUCM features such as call routing, multi-site deployment, media resources, and SIP trunks. They will also be evaluated on their knowledge of Cisco Unity Connection features such as voicemail, unified messaging, and AutoAttendant. Additionally, the exam covers Cisco Unified IM&P features such as presence, chat, and group chat.
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Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q135-Q140):NEW QUESTION # 135
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 # 136
Users report slrange behaviors when Ihey call external numbers from the company phone system. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. There is no indication of the possible problem, but the connection seems to be established Which command meets the requirement to capture all the data needed to debug the problem?
- A. debug mgcp media tracelevel critical verbose
- B. debug mgcp media tracelevel moderate
- C. debug mgcp media tracelevel critical
- D. debug mgcp media tracelevel critical all
Answer: B
NEW QUESTION # 137
Refer to the exhibit.
In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)
- A. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- B. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.
- C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
- D. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.
- E. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
Answer: C,E
NEW QUESTION # 138
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
- A. interworking between h245-alpha numeric and sip-kpml
- B. interworking between an OOB method and RFC2833 for flow-around calls
- C. interworking between an OOB method and RFC2833 for flow-through calls
- D. interworking between h245-signal and rtp-nte
Answer: B
NEW QUESTION # 139
Refer to the exhibit.
Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?
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