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  1. Home
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  3. 300-815

Cisco 300-815 Exam (page: 3)
Cisco Implementing Advanced Call Control and Mobility Services (CLASSM)
Updated on: 31-Mar-2026

Viewing Page 3 of 32
300-815 PDF 245 Questions



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?

  1. no midcall-signaling passthru
  2. no update-callerid
  3. no contact-passing
  4. rel1xx require "100rel"

Answer(s): B





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?

  1. The administrator must associate the route pattern with a calling search space the device can dial.
  2. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.
  3. The administrator must allow connectivity so that TCP connections do not fail between the nodes.
  4. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.

Answer(s): C





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?

  1. Update the calling search space for affected endpoints to none.
  2. Update all regions to 8 kbps maximum audio bitrate.
  3. Increase the number of available media termination points.
  4. Add a media resource to transcode between available capabilities.

Answer(s): 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



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=

Refer to the exhibit.
Which INVITE is sent to 10.10.100.123 as a result of this log?

  1. 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:33334444@10.10.100.123:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "1000" <sip:1000@10.122.200.50>;tag=32412716~41f7
    To: <sip:33334444@10.10.100.123>
    Date: Thu, 01 Apr 2021 17:20:50 GMT
    Call-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50
    Supported: timer,resource-priority,replaces
    Min-SE: 1800
    User-Agent: Cisco-CUCM12.0
  2. 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:33334444@10.10.100.123:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "11112222" <sip:11112222@10.122.200.50>;tag=32412716~41f7
    To: <sip:11112222@10.10.100.123>
    Date: Thu, 01 Apr 2021 17:20:50 GMT
    Call-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50
    Supported: timer,resource-priority,replaces
    Min-SE: 1800
    User-Agent: Cisco-CUCM12.0
  3. 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:11112222@10.10.100.123:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "1000" <sip:1000@10.122.200.50>;tag=32412716~41f7
    To: <sip:11112222@10.10.100.123>
    Date: Thu, 01 Apr 2021 17:20:50 GMT
    Call-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50
    Supported: timer,resource-priority,replaces
    Min-SE: 1800
    User-Agent: Cisco-CUCM12.0
  4. 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:11112222@10.10.100.123:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3ae From: "11112222" <sip:11112222@10.122.200.50>;tag=32412716~41f7
    To: <sip:11112222@10.10.100.123>
    Date: Thu, 01 Apr 2021 17:20:50 GMT
    Call-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50
    Supported: timer,resource-priority,replaces
    Min-SE: 1800

    User-Agent: Cisco-CUCM12.0

Answer(s): C





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?

  1. Disable "SIP Rel1XX Options" and "Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.
  2. Enable "SIP Rel1XX Options" and "Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.
  3. Disable "Send send-receive SDP in mid-call INVITE" on the SIP Profile Configuration Page in Cisco UCM.
  4. Enable "Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.

Answer(s): D





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?

  1. voice service voip sip no rel1xx
  2. sip-ua disable-early-media 180
  3. voice service voip sip rel1xx require 100rel
  4. voice service voip sip send 180 sdp

Answer(s): C



What are two functions of the metadata in SIP recording? (Choose two.)

  1. includes the details of the application media stream
  2. contains the cryptographic context of the SRTP passthrough calls
  3. identifies the session and media association time
  4. carries the Agent Statistics Report
  5. carries the communication session data (audio and video calls) that describes the call to the recording server

Answer(s): C,E


Reference:

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/m_voi-cube- sip-rec-recorder.pdf





Refer to the exhibit. Calls from users to the PSTN in an organization get disconnected with a 408 Request Timeout when the called party is unavailable to pick up the call.
Which solution must be used to resolve this challenge?

  1. Configure midcall-signaling preserve-codec.
  2. Choose "Send PRACK" if 1xx contains SDP in the SIP profile.
  3. Configure midcall-signaling passthru media-change.
  4. Choose "Send PRACK for all 1xx Messages" in the SIP profile.

Answer(s): D



Viewing Page 3 of 32



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