300-815 CLACCM Implementing Cisco Advanced Call Control and Mobility Services
Duration: 90 minutes
Languages: English
Exam overview
Associated certifications: CCNP Collaboration
Cisco Certified Specialist – Collaboration Call Control & Mobility Implementation
This exam tests your knowledge of advanced call control and mobility services, including:
Signaling and media protocols
CME/SRST gateway technologies
Cisco Unified Board Element
Call control and dial planning
Cisco Unified CM Call Control and
Mobility
Implementing Cisco Advanced Call Control and Mobility Services v1.0 (300-815)
Exam Description: Implementing Cisco Advanced Call Control and Mobility Services v1.0 (CLACCM 300-815) is a 90-minute exam associated with the CCNP Collaboration Certification. This exam tests a candidate’s knowledge of advanced call control and mobility services, including signaling and media protocols, CME/SRST gateway technologies, Cisco Unified Board Element, call control and dial planning, Cisco Unified CM Call Control, and mobility. The course, Implementing Cisco Advanced Call Control and Mobility Services, helps candidates to prepare for this exam.
The following topics are general guidelines for the content likely to be included on the exam. However, other related topics may also appear on any specific delivery of the exam. To better reflect the contents of the exam and for clarity purposes, the guidelines below may change at any time without notice.
20% 1.0 Signaling and Media Protocols
1.1 Troubleshoot these elements of a SIP conversation
1.1.a Early media
1.1.b PRACK
1.1.c Mid-call signaling (hold/resume, call transfer, conferencing)
1.1.d Session timers
1.1.e UPDATE
1.2 Troubleshoot these H.323 protocol elements
1.2.a DTMF
1.2.b Call set up and tear down
1.3 Troubleshoot media establishment
10% 2.0 CME/SRST Gateway Technologies
2.1 Configure Cisco Unified Communications Manager Express for SIP phone registration
2.2 Configure Cisco Unified CME dial plans
2.3 Implement toll fraud prevention
2.4 Configure these advanced Cisco Unified CME features
2.4.a Hunt groups
2.4.b Call park
2.4.c Paging
2.5 Configure SIP SRST gateway
15% 3.0 Cisco Unified Border Element
3.1 Configure these Cisco Unified Border Element dial plan elements
3.1.a DTMF
3.1.b Voice translation rules and profiles
3.1.c Codec preference list
3.1.d Dial peers
3.1.e Header and SDP manipulation with SIP profiles
3.1.f Signaling and media bindings
3.2 Troubleshoot these Cisco Unified Border Element dial plan elements
3.2.a DTMF
3.2.b Voice translation rules and profiles
3.2.c Codec preference list
3.2.d Dial peers
3.2.e Header and SDP manipulation with SIP profiles
3.2.f Signaling and media bindings
25% 4.0 Call Control and Dial Planning
4.1 Configure these globalized call routing elements in Cisco Unified Communications Manager
4.1.a Translation patterns
4.1.b Route patterns
4.1.c SIP route patterns
4.1.d Transformation patterns
4.1.e Standard local route group
4.1.f TEHO
4.1.g SIP trunking
4.2 Troubleshoot these globalized call routing elements in Cisco Unified Communications Manager
4.2.a Translation patterns
4.2.b Route patterns
4.2.c SIP route patterns
4.2.d Transformation patterns
4.2.e Standard local route group
4.2.f TEHO
4.2.g SIP trunking
20% 5.0 Cisco Unified CM Call Control Features
5.1 Troubleshoot Call Admission Control (exclude RSVP)
5.2 Configure ILS, URI synchronization, and GDPR
5.3 Configure hunt groups
5.4 Configure call queuing
5.5 Configure time of day routing
5.6 Configure supplementary functions
5.6.a Call park
5.6.b Meet-me
5.6.c Call pick-up
10% 6.0 Mobility
6.1 Configure Cisco Unified Communications Manager Mobility
6.1.a Unified Mobility
6.1.b Extension Mobility
6.1.c Device Mobility
6.2 Troubleshoot Cisco Unified Communications Manager Mobility
6.2.a Unified Mobility
6.2.b Extension Mobility
6.2.c Device Mobility
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QUESTION 1
An administrator is troubleshooting a one-way audio issue for a call that uses H.323 protocol in slow-start mode. The administrator requests that the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call is provided. The H.225 and H.245 messages for one of the one-way audio calls are gathered and the call flow has not invoked any media resources. Where is the RTP IP and port information for both sides found?
A. H.245 Terminal Capability Set
B. H.245 Open Logical Channel
C. H.225 Connect
D. H.245 Open Logical Channel Ack
Answer: B
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: A,B
QUESTION 3
When an administrator troubleshoots H.323 call setup, which message gives an alert that the called party is being notified about the call?
A. ALERTING
B. PROCEEDING
C. CONNECT
D. RINGING
Answer: C
QUESTION 4
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
QUESTION 5
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
B. Cisco UCM invoked media termination point resources.
C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
D. A firewall in the media path is blocking TCP ports 16384-32768.
Answer: D
