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Cisco 300-815 (Implementing Cisco Advanced Call Control and Mobility Services) Exam is designed to test the knowledge and skills of IT professionals who are involved in implementing advanced call control and mobility services solutions using Cisco technologies. 300-815 exam is one of the requirements for obtaining the Cisco Certified DevNet Specialist - Collaboration Automation and Programmability certification.
NEW QUESTION # 63
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?
- A. dtmf-relay sip-kpml
- B. dtmf-relay cisco-rtp
- C. dtmf-relay h245-signal
- D. dtmf-relay rtp-nte
Answer: A
NEW QUESTION # 64
What is a component of Cisco Unified Mobility?
- A. Single Number Connect
- B. Smart Client Support
- C. Mobile Connect
- D. Unified IVR
Answer: C
Explanation:
Section: Mobility
Explanation/Reference:
NEW QUESTION # 65
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)
- A. The maximum number of callers allowed in queue must be 10.
- B. A unicast music on hold audio source must be configured.
- C. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
- D. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
- E. The phone button template must have the Queue Status Softkey configured.
Answer: C,D
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-
1201_chapter_01001101.html#CUCM_RF_C960BC9A_00
NEW QUESTION # 66
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button Which configuration parameter causes this problem?
- A. The voicemail pilot number is missing from the call handling on Cisco Unity Express
- B. The voicemail pilot number for call forwarding is missing from the ephone-dn
- C. The voicemail pilot number for call forwarding is missing from the ephone
- D. The voicemail pilot number is missing from the telephony service configuration on Cisco UCME
Answer: B
NEW QUESTION # 67
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
- A. route group
- B. line group
- C. hunt list
- D. hunt pilot
Answer: B
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-1201_chapter_010101.html
NEW QUESTION # 68
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
- A. Contact: header of the 200 OK response
- B. o= line of SDP content
- C. Allow: header if the 200 OK response
- D. c= line of SDP content
Answer: D
NEW QUESTION # 69
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?
- A. Verify that the route pattern has the correct calling-party transformation mask
- B. Verify that the dial peer of the gateway has the correct destination pattern configured.
- C. Verify that the route pattern is not blocking calls to the destination number.
- D. Verify that IP routing is correct between the gateway and the IP phone.
Answer: B
NEW QUESTION # 70
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. Incoming Port in Security Information of the SIP Profile configuration.
- B. Destination Port in SIP Information section of the SIP Trunk configuration
- C. Incoming Port in SIP Information section of the SIP Trunk configuration.
- D. Destination Port in SIP Trunk Security Profile configuration
- E. Incoming Port in SIP Trunk Security Profile configuration
Answer: B,E
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 71
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. incoming called number 555.......
- B. answer-address 555 ........
- C. codec g729
- D. session-protocol sipv2
Answer: D
NEW QUESTION # 72
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. MTP
- C. OS Administration Page
- D. CCSIP
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 73
Refer to the exhibit. An engineer is troubleshooting an issue where inbound Calls are failing after they transferred. The provider reports that update is not supported, and this is causing the calls to fail. Which command should resolve this issue?
- A. no update-callerId
- B. rel1xx require "100rel"
- C. no midcall-signaling passthru
- D. no contact-passig
Answer: B
NEW QUESTION # 74
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
- B. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- C. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- D. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
Answer: D
NEW QUESTION # 75
An administrator is working on an issue between the customer s Cisco Unified Border Element and the service provider. The provider only wants to see mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?
- A. midcall-signaling passthru media-change
- B. midcall-signallng passthru
- C. midcall-signaling preserve-codec
- D. no update-callerid
Answer: A
NEW QUESTION # 76
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. device pools
- B. IP phones
- C. CTI ports
- D. translation patterns
- E. CTI route points
Answer: B,C
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/dna/11_5_1/ CUCM_BK_CBA47A6E_00_cucm-dna-guide-115/CUCM_BK_CBA47A6E_00_cucm-dna-guide-
115_chapter_01.html#CUCM_TP_A5DA99E0_00
NEW QUESTION # 77 
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 613-555-1234 and numbering type "subscriber"
- C. calling number 011-1-613-555-1234 and numbering type "subscriber"
- D. calling number 011613-555-1234 and numbering type "international"
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 78
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. ptime mismatch
- B. missing Call Admission Control
- C. phone class of service issue
- D. codec mismatch
Answer: B
NEW QUESTION # 79 
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. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- B. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- C. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
- D. No DTMF is negotiated.
Answer: D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION # 80
Refer to the exhibit.
Which INVITE is sent to 10.10.100.123 as a result of this log?
A)
B)
C)
- A. Option B
- B. Option D
- C. Option C
- D. Option A
Answer: C
NEW QUESTION # 81
A user reports that when they attempt to log out from the Cisco Extension Mobility service by pressing the Services button, they cannot log out. What is the most likely cause of this issue?
- A. The user device profile is not subscribed to the Cisco Extension Mobility service.
- B. The user device profile has not been assigned to the user.
- C. The Cisco Extension Mobility service has not been configured on the phone.
- D. There might be a significant delay between the button being pressed and the Cisco Extension Mobility service recognizing it. It would be best to check network latency.
Answer: A
NEW QUESTION # 82
Refer to the exhibit.
How many maximum hops can an ILS updarte traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION # 83
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
- A. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- B. 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.
- C. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
- 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 support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
Answer: B,D
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 # 84
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Passing the Cisco 300-815 certification exam is an excellent way to demonstrate your expertise in advanced call control and mobility services. It can also help you advance your career in fields such as network engineering, network architecture, and systems engineering. With the increasing demand for professionals who can configure and troubleshoot advanced call control and mobility services, the Cisco 300-815 certification can give you a competitive edge in the job market.
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