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How to study the Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam
Cisco provides numerous solutions on their website for the Cisco Service Provider VPN Services (Cisco 300-815) Exam. Cisco offers training in classrooms, through which Cisco’s approved learning partners provide instructor-led courses worldwide. Cisco provides e-learning solutions for the preparation of examinations via self-paced online courses. Students are strongly encouraged to join the Cisco Certification Community where they may connect and learn together from students all around the globe. 300-815 practice exams are required to have a correct grasp of the test structure and kinds of questions.
As preparation for this test, we propose a mixture of practical experience, training courses and self studies in the areas mentioned in the Exam Outline part of the Exam Guide. Check 300-815 dumps for subjects included in the Exam Outline portion of this manual, examine the online documentation, tip sheets and user manuals, and study specifics related to these subjects. For extra study material, go to the links at the end of this text.
Cisco CLASSM 300-815 Practice Test Questions, Cisco CLASSM 300-815 Exam Practice Test Questions
If you are looking for the best way to validate your knowledge and skills in regards to the implementation of the advanced call control and mobility services, you can opt for the Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services exam and obtain two certifications associated with it. Thus, you will be eligible for earning one of the Cisco Certified Specialist certificates (Collaboration Call Control & Mobility Implementation). You can also get the CCNP Collaboration certification if you pass this test after clearing the Cisco 350-801 exam.
How to Prepare for Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam
Preparation Guide for Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam
Introduction
Cisco Systems, Inc. is an American global technology company based in San Jose, California, near the heart of Silicon Valley. Cisco designs the manufacture and sale of hardware, software, telecom equipment and other high-tech networking services and products. CISCO, via its many acquired companies, such as OpenDNS, WebEx, Jabber and Jasper, specialized in unique technology markets such as the Internet of Things, domain security and energy management. Cisco technology creates a world of potential.
With Cisco’s Training and Certification, you may enhance your career with a new learning portfolio that offers both developers and network engineers chances. Cisco’s training and certification programmed provides a straight route to your technological professional objectives. IT technology led to the evolution of Cisco’s training and certification programmers to educate instructors, engineers and software developers for success in the industry’s most significant jobs. Explore the power of the Cisco Learning Network’s dynamic culture to start your certification and lifetime learning objectives. Obtain valuable IT training resources for all Cisco certifications. Access IT certification research tools, CCNA practice tests, IT salary and IT job findings. The purpose of this exam guide is to assist you assess if you can successfully finish Cisco IP network troubleshooting and support 300-815 exam test. This guide provides information on the test target audience, required preparation and documentation, as well as a complete list of examination aims, to help you get a degree. We strongly advised to study the 300-815 dumps and 300-815 practice exams for increased chances of passing the exam.
NEW QUESTION 28
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Configure the command no ip address trusted authenticate under "voice service voip".
- B. Enable Secondary Dial tone on Analog and Digital FXO Ports.
- C. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
Answer: C
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 29 
Refer to the exhibit. Within the North American Numbering Plan, gateways located in Ottawa, Canada and marked as "YOW" are assigned to the Calling Party Transformation CSS NANP_CgPTP, which contains partition NANP_calling_xforms. What is the calling-party number and the numbering type if the calling user
+1613-555-1234 dials the number?
- A. calling number 613-555-1234 and numbering type "national"
- B. calling number 011613-555-1234 and numbering type "international"
- C. calling number 011-1-613-555-1234 and numbering type "subscriber"
- D. calling number 613-555-1234 and numbering type "subscriber"
Answer: A
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:
NEW QUESTION 30
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. answer-address
- B. incoming uri
- C. target carrier-id
- D. incoming called-number
Answer: B
NEW QUESTION 31
Which description of RTP timestamps or sequence numbers is true?
- A. Timestamps increase by the time "carrying" by a packet.
- B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- C. The sequence number is used to detect losses.
- D. Sequence numbers increase by four for each RTP packet transmitted.
Answer: B
Explanation:
Reference:
https://www.cs.columbia.edu/~hgs/rtp/faq.html
NEW QUESTION 32
Refer to the exhibit.
An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?
- A. answer-address 555 ........
- B. incoming called number 555.......
- C. session-protocol sipv2
- D. codec g729
Answer: C
NEW QUESTION 33
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port
5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)
- A. Destination Port in SIP Trunk Security Profile configuration
- B. Destination Port in SIP Information section of the SIP Trunk configuration
- C. Incoming Port in SIP Trunk Security Profile configuration
- D. Incoming Port in Security Information of the SIP Profile configuration.
- E. Incoming Port in SIP Information section of the SIP Trunk configuration.
Answer: B,C
NEW QUESTION 34
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
- A. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
- B. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
- C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- D. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
- E. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.
Answer: D,E
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/ CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features- services-guide-100_chapter_010000.pdf
NEW QUESTION 35
CollabCorp is a global company with two clusters, emea.collab corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make inter-cluster calls via URI. but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)
- A. SIP route pattern
- B. calling search space and partition
- C. directory URI partition
- D. SIP trunk
- E. intercluster trunk
Answer: A,D
NEW QUESTION 36
Refer to the exhibit.
An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu. Which two steps must be performed to resolve this issue? (Choose two.)
- A. Activate the Cisco Dialed Number Analyzer Server service.
- B. Activate the Cisco CallManager service.
- C. Activate the Cisco Extended Functions service.
- D. Restart the subscriber
- E. Activate the Cisco Dialed Number Analyzer service.
Answer: A,E
NEW QUESTION 37
Which two types of authentication are supported for the configuration of Intercluster Lookup Service?
(Choose two.)
- A. username and secret key
- B. TLS certificates
- C. TokenID
- D. FQDN of the servers defined in DNS
- E. passwords
Answer: B,E
NEW QUESTION 38
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 relay
- B. SIP trunk
- C. transcoding
- D. secure SIP lines
- E. three-way conference
Answer: A,E
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 39
A user in location X dials an extension at location Y.
The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?
- A. codec mismatch
- B. ptime mismatch
- C. missing Call Admission Control
- D. phone class of service issue
Answer: C
NEW QUESTION 40
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Directed Call Pickup
- B. Other Group Pickup
- C. Group Call Pickup
- D. BLF Call Pickup
Answer: B
NEW QUESTION 41
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
- A. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323 - B. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod - C. Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily - D. Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad
Answer: A
Explanation:
Section: Call Control and Dial Planning
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_call_handling.html
NEW QUESTION 42
When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?
- A. Cisco Extension Mobility is not enabled in the Phone Configuration Window (Device > Phone)
- B. The URL configured for Cisco Extension Mobility is not correct.
- C. Cisco Extension Mobility Service is not running.
- D. The phone is not subscribed to Cisco Extension Mobility Service.
Answer: D
NEW QUESTION 43
Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.
Answer:
Explanation:
NEW QUESTION 44
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
- A. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323 - B. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod - C. Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily - D. Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad
Answer: A
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_call_handling.html
NEW QUESTION 45
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
- A. interworking between an OOB method and RFC2833 for flow-around calls
- B. interworking between h245-signal and rtp-nte
- C. interworking between an OOB method and RFC2833 for flow-through calls
- D. interworking between h245-alpha numeric and sip-kpml
Answer: A
NEW QUESTION 46
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Directed Call Pickup
- B. Other Group Pickup
- C. Group Call Pickup
- D. BLF Call Pickup
Answer: B
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmsys/ a07cpick.html#wp1022865
NEW QUESTION 47
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next. Which distribution algorithm must be configured to accomplish this task?
- A. circular
- B. broadcast
- C. top down
- D. longest idle time
Answer: C
NEW QUESTION 48
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)
- A. Ask the firewall administrator to change the ports to TCP.
- B. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- D. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- E. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
Answer: C,D
NEW QUESTION 49
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