300-815 CLACC: Implementing Cisco Advanced Call Control On-PremisesDemo

Cisco Unified CM routes SIP URIs by matching the host portion against configured SIP route patterns. example.com is an exact match for that host, while *.* matches a two-label domain such as example.com. Domain-route patterns permit domain characters and wildcards but not @; the user portion is not part of this match. Cisco IPv6 Deployment Guide—SIP IPv6 Route Patterns

The SDP c= field carries connection-address information for the media stream. If a deep packet-inspection device overwrites that information incorrectly in the signaling path, the originating endpoint can send RTP to an incorrect media destination, preventing receipt at the far endpoint.

Cisco Unified Communications Manager applies closest-match logic to overlapping digit patterns. 12[2-5] is the most specific match for 123, matching only 122 through 125. Urgent Priority does not give a broader matching pattern precedence; it only causes immediate routing when that urgent pattern is already the best match.

Cisco SIP IP phone media uses RTP over UDP, and the media-port range is configured in the SIP profile assigned to the phone. The default range is 16384–32767, which does not match a firewall rule allowing only UDP 20000–22000. The issue is resolved by allowing UDP 16384–32767 through the firewall or by configuring the assigned SIP profile to use UDP 20000–22000.

A numbered voice register pool in Cisco Unified CME configures an individual SIP phone. It holds the phone-specific identity and settings, such as its MAC address, phone type, and directory-number assignments; shared settings are configured globally or through templates.

Call Admission Control limits concurrent calls on a WAN link to the available bandwidth. QoS prioritizes voice traffic but cannot prevent oversubscription when too many calls are admitted; the resulting packet loss or variable delay causes choppy or clipped audio.

Cisco Unified CM adds the Standard Local Route Group to a route list. When a call uses that route-list entry, Unified CM selects the actual route group configured in the device pool of the calling device, enabling site-specific egress routing from a common dial plan.

The Standard Local Route Group is configured in a device pool. Devices assigned to that device pool use its local route group when routing calls through a route list that includes the Standard Local Route Group.

Cisco Unified CM SIP line-side transfer uses the SIP REFER method, with the transferring endpoint acting as the transferor and identifying the transfer target in the Refer-To header. Activating Transfer invokes network hold; therefore, Phone B receives MOH selected from Phone A’s Network Hold MOH Audio Source setting.

Calling search spaces control which partitions a calling device can search when completing a call. Cisco Unified Communications Manager records the associated digit-analysis and call-routing details in CallManager traces, making those traces the appropriate source for diagnosing calling search space problems.

A SIP 183 Session Progress message is a provisional 1xx response. PRACK acknowledges reliable provisional SIP responses, and configuring SIP Rel1XX Options to Send PRACK for all 1xx Messages makes Cisco Unified Communications Manager send that acknowledgement. Cisco documents this setting specifically for configuring reliable provisional responses between CUCM and CUBE.

The SDP c= line contains the connection information, including the network address used for the media stream. Checking it can reveal an incorrect or unreachable RTP address that causes one-way audio. RFC 8866: SDP: Session Description Protocol

Cisco Unified CME uses park-slot timeout <seconds> limit <count> recall to return a parked call to the phone that parked it after the configured number of timeout intervals. Two 30-second intervals equal 60 seconds, so timeout 30 limit 2 recall provides the required recall timing; alternate 3002 supplies a fallback destination if the recall target is busy.

IPv6 Unified SRST supports three-way conferencing (with BIB conferencing only) and PSTN fax support using T.38 fax relay. Secure SIP lines, SIP trunks, and transcoding are unsupported in this IPv6 deployment.

Cisco Unified Border Element requires locally registered transcoding resources when it interworks an out-of-band DTMF method with RFC 2833 on a flow-around call. Flow-through calls can interwork the other DTMF relay methods without a transcoder.

For external-to-internal calls, the calling-party number is normalized to E.164, while the called-party number is represented as the internal directory number needed to route the call to the destination. Cisco’s dial-plan guidance describes normalizing inbound PSTN calls at the gateway and manipulating the called E.164 number into the internal number required to reach the destination phone.

The inbound CUBE dial peer must match the provider’s DNIS before its destination dial-peer group can route the call. Configuring specific inbound matching for 222333444 selects dial peer 999; its existing outgoing translation profile then rewrites the called number to 444333222 before the call is sent to Cisco Unified CM. A destination dial-peer group statically chooses peers in that group, so the outbound peer’s destination-pattern is required syntactically but is not used for dial-peer matching in this configuration.

RTMT uses Voice/Video > Session Trace Log View > Real Time Data to display session-trace information for calls in real time. The local-disk option is for viewing previously saved trace logs rather than live call data.

IP address trusted authentication validates the remote address of incoming SIP or H.323 VoIP calls against the trusted IP address list, blocking unauthorized calls that could be used for toll fraud. In Unified CME, ip address trusted authenticate enables this protection under voice service voip; the no form disables it.

voice register global enters the Cisco Unified CME configuration mode for global parameters of supported SIP phones, including the settings used for SIP-phone registration and provisioning.

DTMF and BFCP must be configured as the extended capabilities for this Fast Start-to-early-media interworking scenario. Cisco documents that Fast Start-to-Early Offer supplementary-service interworking requires extended capabilities under the dial peer, while basic media and fax relay are handled separately.

For a secure SIP trunk, the SIP Trunk Security Profile Incoming Port specifies the local port on which Unified CM accepts inbound SIP traffic. The SIP Trunk configuration’s SIP Information Destination Port specifies the remote provider port used for outbound SIP traffic. Setting both to 5065 satisfies the required inbound and outbound port configuration.

Client Matter Code does not support overlap sending because Cisco Unified Communications Manager cannot determine when to prompt the user for the code. Selecting Allow Overlap Sending disables Require Client Matter Code. Authorization Level applies to Forced Authorization Codes rather than Client Matter Codes.

In Cisco Unified CM route patterns, square brackets match a single digit from the listed set. 813510[012345] fixes the first six digits and permits only a final digit from 0 through 5, so it matches exactly 8135100–8135105.