Cisco 300-815 Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) Online Training
Cisco 300-815 Online Training
The questions for 300-815 were last updated at Oct 23,2025.
- Exam Code: 300-815
- Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
- Certification Provider: Cisco
- Latest update: Oct 23,2025
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
Which description of RTP timestamps or sequence numbers is true?
- A . The sequence number is used to detect losses.
- B . Timestamps increase by the time “carrying” by a packet.
- C . Sequence numbers increase by four for each RTP packet transmitted.
- D . The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
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
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.
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
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A . Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
- B . Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- C . Configure the command no ip address trusted authenticate under “voice service voip”.
- D . Enable Secondary Dial tone on Analog and Digital FXO Ports.
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration.
Which configuration is occurring in this section?
- A . configuration for a single SIP phone
- B . configuration items common for all SIP phones
- C . configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
- D . configuration for SIP registrar service
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
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
- A . interworking between an OOB method and RFC2833 for flow-around calls
- B . interworking between h245-signal and rtp-nte
- C . interworking between an OOB method and RFC2833 for flow-through calls
- D . interworking between h245-alpha numeric and sip-kpml
Where is the dtmf-relay command configured on Cisco Unified Border Element?
- A . in the voice-class VoIP configuration
- B . in the VoIP dial peer
- C . in global SIP configuration
- D . in the VoIP or POTS dial peers
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