300-815 Implementing Cisco Advanced Call Control On-Premises (CLACC) Practice Questions

A Recording Profile is the CUCM configuration object that links a directory number to the call recording system. The profile specifies the recording destination (a route pattern pointing to the recording server) and the Calling Search Space to be used for recording calls. This profile is then applied directly to the Directory Number configuration page for the users who need to be recorded.

This scenario is the classic use case for UCM's native Location-Based Call Admission Control (CAC). By creating separate locations for Chicago and Denver and defining a specific bandwidth limit for audio (and video, if applicable) calls between them, UCM will track the number of active calls over the WAN. Once the allocated bandwidth is consumed, UCM will prevent any new calls from being established across that link, returning a busy signal to the user. This meets all requirements: it's managed by UCM, doesn't use RSVP, prevents new calls during congestion, and allows intra-site calls (which are assigned to the 'Unlimited' bandwidth pool within the same location) to proceed normally. QoS is a complementary technology but does not prevent oversubscription; it only prioritizes traffic during congestion.

While the c= line in the SDP indicates the address for media, it can be overridden by the IP address in the Contact header if they differ, especially in environments with NAT or a Session Border Controller. The Contact header provides the direct address for future requests within the dialog, including where to send RTP. The Via and Record-Route headers are used for signaling path routing, not for media destination.

Voice translation-profiles must be applied to the correct call leg direction. If the goal is to modify the called number as it leaves the CUBE, the profile must be applied for outgoing calls on the outbound dial-peer (e.g., voice-class translation-profile MyProfile outgoing). A very common mistake is to apply it to the incoming direction or to forget to specify the direction, which can lead to the rule not being triggered as expected.

This statement is False. SIPREC (Session Recording Protocol) is a comprehensive protocol based on SIP that manages the entire recording session. A key feature of SIPREC is that the Session Recording Client (like a CUBE) establishes a recording session with the Session Recording Server and forks both the call metadata (signaling information) and the real-time media streams (RTP) to the server. It does not rely on a separate mechanism for media forking.

Called Party Transformation Patterns are applied at the routing device level. The transformation CSS (Trans_CSS) must be applied to the configuration of the device that is routing the call, such as a gateway or a trunk. For route patterns, this is done by assigning the transformation CSS to the 'Called Party Transformation CSS' field within the route pattern itself. This ensures that before CUCM attempts to send the call to the destination specified in the route pattern, it first applies the transformation.

Time-of-day routing in UCM uses a hierarchical structure. First, you define Time Periods (e.g., 9 AM to 5 PM). Then, you group these Time Periods into a Time Schedule and associate them with specific days of the week (e.g., Monday-Friday). Finally, this Time Schedule is applied to a partition, which is then placed in a time-aware Calling Search Space to control routing.

The key to this functionality is enabling the 'Advertised to mobile phone' option for the Voicemail Calling Party Transformation CSS within the Remote Destination Profile. This feature, often called 'Reverse MWI' or similar, allows the mobile network to signal back to CUCM when a call is rejected or sent to the mobile voicemail, which in turn allows CUCM to redirect the call immediately to the corporate voicemail system. The other options are related to different aspects of Unified Mobility but do not control this specific call rejection behavior.

The correct syntax to copy the URI from the 'From' header to a new P-Asserted-Identity header is request INVITE sip-header P-Asserted-Identity add "P-Asserted-Identity: ". The special token @@ is a variable that represents the URI from the From/To header (depending on context). The other options use incorrect syntax or commands. uas-header and header-passing are not the correct keywords for adding a new header based on an existing one, and the first option attempts to add a literal string rather than copying the dynamic value.

In an EMCC environment, call routing logic (CSS) is taken from the user's home cluster configuration, specifically from their User Device Profile. This ensures consistent dialing behavior regardless of location. However, media resources like MOH, conference bridges, and transcoders are allocated from the device pool of the visiting phone to ensure media stays as local as possible and to conserve WAN bandwidth.