This page was exported from IT Certification Exam Braindumps [ http://blog.braindumpsit.com ] Export date:Sat Apr 12 19:25:17 2025 / +0000 GMT ___________________________________________________ Title: Free Cisco 300-815 Test Practice Test Questions Exam Dumps [Q72-Q92] --------------------------------------------------- Free Cisco 300-815 Test Practice Test Questions Exam Dumps Prepare Top Cisco 300-815 Exam Audio Study Guide Practice Questions Edition The Cisco 300-815 exam is a certification test that evaluates the knowledge and skills of IT professionals in implementing advanced call control and mobility services using Cisco technologies. This exam is designed for network administrators, engineers, and consultants who are responsible for designing, deploying, and managing enterprise-level collaboration solutions using Cisco Unified Communications Manager (CUCM) and its associated applications. The exam consists of multiple-choice questions that cover a wide range of topics, including Cisco Unified Communications Manager, Cisco Unity Connection, Cisco Expressway, Cisco Unified Border Element, and Cisco TelePresence Management Suite. The exam also covers topics related to dial plans, call routing, and mobility services, including device mobility, extension mobility, and mobile voice access.   QUESTION 72An 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.)  example.com@gateway.corporate.local  *@example.com  gateway.corporate.local  example.com  *.* Section: Call Control and Dial PlanningQUESTION 73What is a component of Cisco Unified Mobility?  Unified IVR  Mobile Connect  Smart Client Support  Single Number Connect QUESTION 74In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?  Dialed Number Analyzer  Real-Time Monitoring Tool  SDI trace  SDL trace Section: Call Control and Dial PlanningQUESTION 75When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)  AAR destination mask  called party transform mask  external phone number mask  +E.164 alternate number mask  enterrise alternate number mask Section: Cisco Unified CM Call Control FeaturesExplanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.htmlQUESTION 76Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?  System > Location Info  Call Routing > Route/Hunt > Local Route Group Names  System > Device Pool  Call Routing > Emergency Location > Emergency Location (ELIN) Groups Section: Call Control and Dial PlanningExplanation/Reference: https://www.uccollabing.com/configuring-standard-local-route-group-cucm/QUESTION 77A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through 8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?  813510[012345]  813510[12345]  813510[^0-5]  81XXXXX QUESTION 78Which services are needed to successfully implement Cisco Extension Mobility in a standalone Cisco Unified Communications Manager server?  Cisco Extended Functions, Cisco Extension Mobility, and Cisco AXL Web Service  Cisco CallManager, Cisco TFTP, and Cisco CallManager SNMP Service  Cisco CallManager, Cisco TFTP, and Cisco Extension Mobility  Cisco TAPS Service, Cisco TFTP, and Cisco Extension Mobility QUESTION 79Refer to the exhibit.Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?  rule 1 /^9(.*)/A1/  rulel /.*(3548S)/^1/  rulel /^9(d*)/^1/  rule 1/^9123548/^1/ QUESTION 80An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor Incoming command configured? (Choose two.)  “ephone-dn’ configuration mode  “voice register global” configuration mode  “telephony-service” configuration mode  “voice register pool” configuration mode  “dial-peer cor custom” configuration mode QUESTION 81Refer 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.)  Restart the subscriber  Activate the Cisco Extended Functions service.  Activate the Cisco CallManager service.  Activate the Cisco Dialed Number Analyzer service.  Activate the Cisco Dialed Number Analyzer Server service. QUESTION 82Where is the dtmf-relay command configured on Cisco Unified Border Element?  in the voice-class VoIP configuration  in the VoIP dial peer  in global SIP configuration  in the VoIP or POTS dial peers Reference:https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf- relay.htmlQUESTION 83In Cisco Unified Communications Manager, which tool do you use to check SIP traces?  MTP  CCSIP  RTMT  OS Administration Page Section: Call Control and Dial PlanningQUESTION 84Refer to the exhibit.A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?  Enable ‘SIP ReI1XX Options* and -Early Offer Support” on the SIP Profile Configuration Page in Cisco UCM.  Enable *Early Offer support for voice and video calls” on the SIP Profile Configuration Page in Cisco UCM.  Disable “SIP Rel1XX Options* and ‘Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.  Disable “Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM. QUESTION 85Refer 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?  answer-address 555 ……..  codec g729  session-protocol sipv2  incoming called number 555……. QUESTION 86An engineer is configuring a call park feature in Cisco Unified Communications Manager Express. Which command does the engineer use to ensure that the call is reverted to the user after 60 seconds?  R2(config-ephone-dn)#park reservation-group 60  R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002  R2(config-ephone-dn)#park reservation-group 1  R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002 QUESTION 87Where is the dtmf-relay command configured on Cisco Unified Border Element?  in the voice-class VoIP configuration  in the VoIP dial peer  in global SIP configuration  in the VoIP or POTS dial peers Section: Cisco Unified Border ElementExplanation/Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf- relay.htmlQUESTION 88What is first preference condition matched in a SIP-enabled incoming dial peer?  incoming uri  target carrier-id  answer-address  incoming called-number Reference:https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8QUESTION 89Refer 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?  answer-address 555 ……..  codec g729  session-protocol sipv2  incoming called number 555……. Section: Call Control and Dial PlanningQUESTION 90Which 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?  allow-connections sip to sip  voice service voip  voice register global  voice register dn Reference:https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/99946-cme-sip-guide.htmlQUESTION 91Which description of RTP timestamps or sequence numbers is true?  The sequence number is used to detect losses.  Timestamps increase by the time “carrying” by a packet.  Sequence numbers increase by four for each RTP packet transmitted.  The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation). Reference:https://www.cs.columbia.edu/~hgs/rtp/faq.htmlQUESTION 92Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)  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.  If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.  If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.  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.  The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.  Loading … Go to 300-815 Questions - Try 300-815 dumps pdf: https://www.braindumpsit.com/300-815_real-exam.html --------------------------------------------------- Images: https://blog.braindumpsit.com/wp-content/plugins/watu/loading.gif https://blog.braindumpsit.com/wp-content/plugins/watu/loading.gif --------------------------------------------------- --------------------------------------------------- Post date: 2023-06-16 16:53:08 Post date GMT: 2023-06-16 16:53:08 Post modified date: 2023-06-16 16:53:08 Post modified date GMT: 2023-06-16 16:53:08