300-815 | All About Certified 300-815 Exams

Proper study guides for Up to the immediate present Cisco Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) certified begins with Cisco 300-815 preparation products which designed to deliver the Highest Quality 300-815 questions by making you pass the 300-815 test at your first time. Try the free 300-815 demo right now.

Check 300-815 free dumps before getting the full version:

NEW QUESTION 1
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?

  • A. System > Location Info
  • B. Call Routing > Route/Hunt > Local Route Group Names
  • C. System > Device Pool
  • D. Call Routing > Emergency Location > Emergency Location (ELIN) Groups

Answer: B

NEW QUESTION 2
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

  • A. DTMF
  • B. BFCP
  • C. VIDEO
  • D. FAX
  • E. AUDIO

Answer: AB

NEW QUESTION 3
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. three-way conference
  • B. secure SIP lines
  • C. T.38 fax relay
  • D. transcoding
  • E. SIP trunk

Answer: AC

NEW QUESTION 4
Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?

  • A. allow-connections sip to sip
  • B. voice service voip
  • C. voice register global
  • D. voice register dn

Answer: C

NEW QUESTION 5
300-815 dumps exhibit
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_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
  • B. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
  • C. 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.
  • D. 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.
  • E. 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.

Answer: AC

NEW QUESTION 6
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

  • A. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
  • B. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
  • C. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
  • D. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.

Answer: A

NEW QUESTION 7
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

  • A. Analysis Manager > Inventory > Trace File Repositories
  • B. System > Tools > Trace and Log Central
  • C. Voice/Video > Session Trace Log View > Real Time Data
  • D. Voice/Video > Session Trace Log View > Open From Local Disk

Answer: C

NEW QUESTION 8
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 Cisco Extension Mobility service has not been configured on the phone.
  • B. There might be a significant delay between the button being pressed and the Cisco Extension Mobility service recognizing i
  • C. It would be best to check network latency.
  • D. The user device profile has not been assigned to the user.
  • E. The user device profile is not subscribed to the Cisco Extension Mobility service.

Answer: D

NEW QUESTION 9
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)

  • A. example.com@gateway.corporate.local
  • B. *@example.com
  • C. gateway.corporate.local
  • D. example.com
  • E. *.*

Answer: BE

NEW QUESTION 10
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. Allow: header if the 200 OK response
  • C. o= line of SDP content
  • D. c= line of SDP content

Answer: C

NEW QUESTION 11
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?

  • A. debug H.323 messages
  • B. debug H.225 asn1
  • C. debug H.246 asn 1
  • D. debug H.225 media
  • E. debug H.323 asn 1

Answer: B

NEW QUESTION 12
300-815 dumps exhibit
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. There is SIP Delayed Offe
  • C. DTMF is supported only in Early Offer.
  • D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
  • E. No DTMF is negotiated.

Answer: D

NEW QUESTION 13
Which IOS command creates a SIP- enabled dial peer?

  • A. voice dial-peer 20 sip
  • B. dial-peer voice 20 voip
  • C. dial-peer voice 20 pots
  • D. dial peer voice 20 sip

Answer: B

NEW QUESTION 14
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

  • A. can be associated under the route group
  • B. can be associated only under the route list
  • C. chooses the route group that is configured under the device pool of the calling-party device
  • D. chooses the route group that is configured under the device pool of the called-party device
  • E. can be assigned directly to the route pattern

Answer: BD

NEW QUESTION 15
What is a component of Cisco Unified Mobility?

  • A. Unified IVR
  • B. Mobile Connect
  • C. Smart Client Support
  • D. Single Number Connect

Answer: B

NEW QUESTION 16
What are the elements for Device Mobility configuration?

  • A. physical location, device pool, and Device Mobility group
  • B. device pool, Device Mobility group, and region
  • C. physical locatio
  • D. Device Mobility group, and region
  • E. device pool, Device Mobility group, and Cisco IP phone

Answer: A

NEW QUESTION 17
300-815 dumps exhibit
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 “subscriber”
  • B. calling number 011-1-613-555-1234 and numbering type “subscriber”
  • C. calling number 011613-555-1234 and numbering type “international”
  • D. calling number 613-555-1234 and numbering type “national”

Answer: D

NEW QUESTION 18
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 PSTN format and the called number in 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 E.164 format and the called number in localized (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: B

NEW QUESTION 19
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 SIP Information section of the SIP Trunk configuration.
  • B. Incoming Port in Security Information of the SIP Profile configuration.
  • C. Destination Port in SIP Information section of the SIP Trunk configuration
  • D. Incoming Port in SIP Trunk Security Profile configuration
  • E. Destination Port in SIP Trunk Security Profile configuration

Answer: CD

NEW QUESTION 20
300-815 dumps exhibit
Refer to the exhibit. Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?

  • A. Adjust the service parameter T302 timet to the desired value.
  • B. Adjust the service parameter T204 timer to the desired value.
  • C. Check the Urgent Priority check box under 9.911 pattern.
  • D. Point the emergency pattern directly to the PSTN gateway.

Answer: C

NEW QUESTION 21
......

P.S. Surepassexam now are offering 100% pass ensure 300-815 dumps! All 300-815 exam questions have been updated with correct answers: https://www.surepassexam.com/300-815-exam-dumps.html (0 New Questions)