Cisco CCNP Collaboration 300-815 Questions & Answers

• Question 131:

  Refer 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?

  A. Disable "SIP Rel1XX Options" and "Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.

  B. Enable "SIP Rel1XX Options" and "Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.

  C. Disable "Send send-receive SDP in mid-call INVITE" on the SIP Profile Configuration Page in Cisco UCM.

  D. Enable "Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.

  Correct Answer: D

• Question 132:

  Refer to the exhibit.

  

  An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?

  A. voice service voip sip no rel1xx

  B. sip-ua disable-early-media 180

  C. voice service voip sip rel1xx require 100rel

  D. voice service voip sip send 180 sdp

  Correct Answer: C

• Question 133:

  Refer to the exhibit.

  55697959.007 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - before DAapplyCdpnXform() preXformCdpn=11112222 preTag=SUBSCRIBER prePos=11112222 crCdpnMask=33334444 crPrefixDigit=

  crDDI=2

  55697959.008 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - after DAapplyCdpnXform() xformCdpn=33334444 xformTag=SUBSCRIBER xformPos=11112222 55697959.009 |12:20:50.913 |AppInfo |

  RouteListCdrc::transformed cdpn (without unconsumpt digits) = 33334444, unconsumed digit=

  Which INVITE is sent to 10.10.100.123 as a result of this log?

  A. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41 [95992364,NET] INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "1000" ;tag=32412716~41f7 To: Date: Thu, 01 Apr 2021 17:20:50 GMT Call-ID: [email protected] Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0

  B. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41 [95992364,NET] INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "11112222" ;tag=32412716~41f7 To: Date: Thu, 01 Apr 2021 17:20:50 GMT Call-ID: [email protected] Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0

  C. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41 [95992364,NET] INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "1000" ;tag=32412716~41f7 To: Date: Thu, 01 Apr 2021 17:20:50 GMT Call-ID: [email protected] Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0

  D. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41 [95992364,NET] INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "11112222" ;tag=32412716~41f7 To: Date: Thu, 01 Apr 2021 17:20:50 GMT Call-ID: [email protected] Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0

  Correct Answer: C

• Question 134:

  Refer to the exhibit.

  

  Regions have been configured for all major branches based on the available circuit bandwidth. Some calls from Region A endpoints to Region B endpoints are failing to connect. How is the issue resolved?

  A. Update the calling search space for affected endpoints to none.

  B. Update all regions to 8 kbps maximum audio bitrate.

  C. Increase the number of available media termination points.

  D. Add a media resource to transcode between available capabilities.

  Correct Answer: D

  Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200788-Call-Recording-Basic-Configuration-and-T.html

• Question 135:

  Refer to the exhibit.

  

  An administrator has configured a SIP trunk between two Cisco UCM clusters. For calls that should use the trunk, the calls fail with a fast busy. The administrator checks the Cisco CallManager SDL traces and found that the cluster to which the calling device is registered never sends an INVITE to the destination cluster. The administrator also verifies that all nodes from both clusters are powered on, and the CallManager service is running. How is this issue resolved?

  A. The administrator must associate the route pattern with a calling search space the device can dial.

  B. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.

  C. The administrator must allow connectivity so that TCP connections do not fail between the nodes.

  D. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.

  Correct Answer: C

• Question 136:

  Refer to the exhibit.

  

  An engineer is troubleshooting an issue where inbound calls are failing after they are 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 midcall-signaling passthru

  B. no update-callerid

  C. no contact-passing

  D. rel1xx require "100rel"

  Correct Answer: B

• Question 137:

  The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?

  A. Change Session Refresh Method on the SIP profile to INVITE.

  B. Increase Retry INVITE to 20 seconds on the SIP profile.

  C. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.

  D. Enable SIP Rel1XX Options on the SIP profile.

  Correct Answer: A

  Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/213843-troubleshoot-session-refresh-on-cube.html

• Question 138:

  Refer to the exhibit.

  

  While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provide, but there is no response. The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?

  A. Require: 100rel

  B. Content-Type: application/sdp

  C. Contact:

  D. Content-Disposition: session;handling=required

  Correct Answer: A

• Question 139:

  Refer to the exhibit.

  

  A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

  A. Allow Passthrough of Configured Line Device Caller Information must be enabled.

  B. Accept Audio Codec Preferences in Received Offer must be set to On.

  C. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

  D. Early Offer for G Clear Calls must be enabled.

  Correct Answer: C

• Question 140:

  A support engineer is troubleshooting a voice network. When conducting a search for call setup details related to calling search space issues, which trace files should be investigated?

  A. CallManager traces

  B. CTI Manager traces

  C. Cisco IP Manager Assistant

  D. Call logs

  Correct Answer: A