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Cisco 350-801 Exam is a certification exam that is designed to test the knowledge and skills of candidates in implementing and operating Cisco Collaboration Core Technologies. 350-801 exam is one of the most widely recognized and respected certifications in the IT industry, and it is highly sought after by professionals who wish to advance their careers in the field of collaboration technologies.
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Cisco 350-801 Exam is a challenging exam that requires candidates to have a solid understanding of Cisco collaboration technologies. 350-801 exam consists of 60 to 70 multiple-choice questions and has a time limit of 90 minutes. To pass the exam, candidates must achieve a score of at least 825 out of 1000.
The 350-801 certification exam assesses the applicants’ competency in a wide range of various topics. The skills covered in this test are combined in several objectives that are highlighted below:
Answering the questions from this objective, the students must demonstrate that they have the ability to troubleshoot such elements as tear down and call set up, and SDP of a SIP conversation. They also need to show that they are capable of troubleshooting a SIP call between specified two endpoints, troubleshoot DTMF Internetworking and Relay on CUBE and DTMF Relay, and identify the most suitable collaboration codecs for a specific scenario. The topic also covers the learners’ expertise in troubleshooting collaboration endpoints and deploying SIP endpoints, including Bulk Administration Tool, self-provisioning, and manual.
This subject evaluates one’s skills in customizing Cisco Unity Connection mailbox, explaining Cisco Unity Connection call handlers and deploying Cisco Jabber on-premises. In addition, the area measures the candidates’ ability to describe the class models used in providing QoS on a network.
This subject encompasses the candidates’ knowledge of configuring a number of gateway elements, such as DTMF, dial peers, codec preference list, and voice translation profiles and rules, as well as familiarity with configuring ISDN PRI/BRI. This area also involves the skills associated with troubleshooting ISDN/BRI, verifying and configuring MGCP, and identifying the most suitable media resources for a specific scenario (software and hardware).
NEW QUESTION # 185
An administrator must implement a new Cisco Unity Connection cluster integrated with LDAP for users and SIP with Cisco UCM. Assuming a phone system and LDAP directory has been configured, how must the administrator add a new user mailbox with message waiting indication functionality to this setup?
Answer: B
Explanation:
To add a new user mailbox with Message Waiting Indicator (MWI) functionality in a Cisco Unity Connection cluster integrated with LDAP for user management and SIP with Cisco Unified Communications Manager (UCM), follow these steps:
1. Import the New User from LDAP:
- In Cisco Unity Connection Administration, navigate to Users > Import Users.
- Select the appropriate LDAP directory and search for the user.
- Import the user into Unity Connection, which will create a corresponding mailbox.
2. Configure the Mailbox:
- After importing, configure mailbox settings as needed, such as mailbox size, message aging policies, and notification settings.
3. Set Up MWI On and Off Directory Numbers (DNs) in Cisco UCM:
- In Cisco UCM Administration, navigate to Voice Mail > Message Waiting.
- Configure the MWI On and Off DNs that Unity Connection will use to signal MWI status changes to the user's phone.
- Ensure that these DNs are correctly associated with the Unity Connection SIP trunk in UCM.
By following these steps, the new user will have a mailbox in Unity Connection, and MWI functionality will be properly configured to indicate new voice messages on their phone.
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/15/administration/guide/b_15c ucsag/b_15cucsag_chapter_011.html
NEW QUESTION # 186
Where does an administrator establish the trust boundary on a LAN?
Answer: A
Explanation:
On a LAN, the trust boundary is configured on the access switch.
NEW QUESTION # 187
Refer to the exhibit.
Which codec should an engineer select for a call made between "Dallas-REG' & "Austin-REG'?
Answer: C
Explanation:
The codec preference list for the "Dallas-REG" region is "Factory Default low loss".
This list includes the following codecs in order of preference:
G. 729
G. 711
OPUS
MP4A-LATM
The codec preference list for the "Austin-REG" region is "Factory Default low loss". This list includes the following codecs in order of preference:
G. 729
G. 711
OPUS
MP4A-LATM
Since both regions have the same codec preference list, the codec that will be used for a call made between "Dallas-REG" and "Austin-REG" is G.729.
G. 729 is a narrowband speech codec that was developed by the ITU-T in 1988. It is a low-bitrate codec that provides good quality speech at a bitrate of 8 kbps. G.729 is widely used in VoIP applications and is the default codec for many VoIP systems.
G. 711 is a wideband speech codec that was developed by the ITU-T in 1972. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 64 kbps. G.711 is not as widely used as G.729 due to its high bitrate requirements.
OPUS is a lossy audio codec that was developed by the IETF in 2012. It is a low-bitrate codec that provides good quality speech at a bitrate of 6 kbps. OPUS is widely used in VoIP applications and is the default codec for many VoIP systems.
MP4A-LATM is a lossy audio codec that was developed by the IETF in 1999. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 24 kbps. MP4A-LATM is not as widely used as G.729 or OPUS due to its high bitrate requirements.
NEW QUESTION # 188
Which type of greeting in the Call Handler configuration in Cisco Unity Connection overwrites all other greetings?
Answer: A
Explanation:
Section: Collaboration Applications
Explanation:
Alternate
Can be used for a variety of special situations, such as vacations or a leave. An alternate greeting overrides all other greetings.
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/ guide/10xcucsagx/10xcucsag080.html
NEW QUESTION # 189 
Refer to the exhibit There is a call flow between Phone A and Phone B Phone B (holder) places Phone A (holder) on hold Which MRGL and Audio Source are played to Phone A?
Answer: A
NEW QUESTION # 190
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