300-815 Sample Questions & Answers

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.

The core of TEHO is routing calls based on the destination number to the most cost-effective PSTN breakout point. In this case, the New York UCM must have a route pattern (e.g., \+44!) that explicitly matches calls destined for the UK. This route pattern must then be associated with a Route List containing a Route Group that points to the London office's PSTN gateway. This ensures the call is sent over the IP WAN to London before it enters the PSTN, thus saving on international toll charges.

In Cisco Unified Communications Manager Express (UCME), the operator command is used under the telephony-service configuration to define a specific number that should be dialed when a user dials '0'. The 'T' indicates that the subsequent digits are a variable-length number to be dialed. The full command would be something like operator 2125551212T, which dials that number when '0' is pressed.

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.