Implementing Cisco Advanced Call Control On-Premises (CLACC) Free Sample Questions

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.

The voice-class sip rel1xx supported command enables support for reliable provisional responses (PRACK), which is defined in RFC 3262. When a 183 Session Progress with SDP is received, it is considered a provisional response that must be acknowledged reliably with a PRACK message. By enabling rel1xx supported, the CUBE indicates it can handle such responses, ensuring the 183 is processed and forwarded correctly, thus establishing early media. require 100rel would make PRACK mandatory, which could cause failures if the other end doesn't support it. The other two options are incorrect commands for this purpose.

The voice service voip command enters the configuration mode where SIP-specific parameters, including SRST functionality, are defined. Within this mode, the sip command is used, followed by registrar server to enable the gateway to act as a SIP registrar during a WAN failure. The call-manager-fallback command is used for SCCP (Skinny) SRST, not SIP SRST. voice class and dial-peer voice are used for call routing and policy, but they do not enable the core SIP registrar function for SRST.

This statement is True. The purpose of the Standard Local Route Group is to simplify dial plan administration. A generic route pattern is created that points to the 'Standard Local Route Group' as its destination. The actual gateway or route list used for the call is then determined by the 'Local Route Group' setting within the Device Pool of the phone that placed the call. This allows a single route pattern to be used for local calls across multiple sites.

The allow-connections sip to sip command is crucial for enabling CUBE to handle mid-call signaling changes, including call transfers initiated by a SIP REFER message. Without this global configuration command, the CUBE will not process the REFER message correctly and cannot broker the new call leg required for the transfer, causing it to fail. The other options are less likely; UCM typically accepts in-dialog REFERs, MTP is for DTMF/codec issues, and the SIP profile setting is for a different purpose.

The correct configuration requires three main components. First, the Hunt Pilot must have call queuing enabled. Second, this Hunt Pilot must point to a Hunt List which contains the Line Group of agents. Crucially, the overflow logic is configured on the Hunt Pilot itself. You must set the 'Maximum Wait Time in Queue' to 300 seconds (5 minutes) and then specify the overflow destination (a route pattern for the answering service) in the 'Destination when max wait time is exceeded' field. Overflow settings are not configured on the Line Group or Hunt List.

The most effective method is to create an IP address trusted list containing the signaling sources you explicitly trust, which in this case are the Cisco UCM servers. By applying this trusted list to the voice service voip configuration, the CUBE will only accept call setup requests from these specified IPs, effectively preventing unauthorized systems from hijacking the CUBE to make fraudulent calls. An external ACL is difficult to manage if the provider's IPs change, and translation rules can be bypassed. Binding the dial peer is good practice but the trusted list provides the core security against unauthorized origination.