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Cisco 300-815 Exam | Latest 300-815 Real Test - High-Efficient Exam Topics Pdf for your 300-815 Preparing

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The Implementing Cisco Advanced Call Control and Mobility Services exam is a 90-minute test that consists of 60-70 questions. 300-815 exam is conducted in English and is available in different formats, including online proctored and in-person testing. 300-815 exam requires the candidates to have a good understanding of Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, Cisco Unity Connection, and Cisco Unified Communications Manager IM&P.

Cisco 300-815 Exam is an excellent way for candidates to showcase their knowledge and skills in implementing advanced call control and mobility services. Implementing Cisco Advanced Call Control and Mobility Services certification exam is highly respected in the industry and can help candidates advance their career prospects. With proper preparation and training, candidates can successfully pass the exam and earn their certification.

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Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q27-Q32):

NEW QUESTION # 27
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?

A. Enable Secondary Dial tone on Analog and Digital FXO Ports.
B. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
C. Configure the command no ip address trusted authenticate under "voice service voip".
D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.

Answer: B

Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/ cmeadm/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2

NEW QUESTION # 28
Refer to the exhibit.

Calls incoming from the provider are not working through newly set up Cisco Unified Border Element.
Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number "222333444" and Cisco Unified Communications Manager is expecting the called number to be delivered as "444333222". The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?

A. Create specific matching for "222333444" on the incoming dial peer.
B. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
C. Set up translation-profile on the incoming dial peer to match incoming traffic.
D. Change the destination-pattern on the outgoing dial peer to match "444333222".

Answer: C

NEW QUESTION # 29
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

A. configuration for a single SIP phone
B. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
C. configuration items common for all SIP phones
D. configuration for SIP registrar service

Answer: A

Explanation:
To enter voice register pool configuration mode and create a pool configuration for a SIP IP phone in Cisco Unified CME or for a set of SIP phones in Cisco Unified SIP SRST, use the voice register pool command in global configuration mode.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/cme_ v1ht.html#wp2339729225

NEW QUESTION # 30
Refer to the exhibit. An administrator just implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily.
Which action resolves the issue?

A. Change DTMF Signaling Method to "RFC 4733".
B. Change DTMF Signaling Method to "No Preference".
C. Disable SIP Rel1XX Options.
D. Change to a range that does not result in MTP.

Answer: D

NEW QUESTION # 31
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses
H.323protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real- Time Transport Protocol traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

A. H.225 Connect
B. H.245 Open Logical Channel
C. H.245 Terminal Capability Set
D. H.245 Open Logical Channel Ack

Answer: B

NEW QUESTION # 32
......

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