Cisco 300-815 CLACCM cheat sheet

Dial-peer matching order: exact-match dial-peer first, then longest match on the destination pattern; preference (lower value wins) breaks ties among equal-length matches, then the dial-peer with the lowest tag if preference also ties.

Cisco IOS destination patterns: '.' matches any single digit, 'T' matches one or more digits and inserts interdigit timeout, '[1-4]' is a digit range, and '$' is an end-of-string anchor; '^' anchors the start.

Translation rules operate in two steps: the match pattern selects the rule, and the replace pattern (with backslash-referenced wildcards such as \1) rewrites the number; unmatched calls are dropped when a translation-profile has no pass-through.

For SIP trunking, dial-peer voice sip-ua with 'session protocol sipv2' and 'session target ipv4:' sends calls to a peer; 'codec' and 'dtmf-relay' are negotiated per dial-peer.

Call routing uses the called number unless 'incoming called-number' or 'answer-address' or 'destination-pattern' with 'incoming called-number' is configured; voice translation profiles can be applied inbound or outbound on a POTS or VoIP dial-peer.

Digit manipulation order in CUCM: translation patterns, route patterns, then route group/route list selection; partitions and calling search spaces determine which patterns a calling device can reach.

Route lists are ordered; the first route group with an available member is chosen, and if all members of a group are busy, the list rolls to the next group unless 'route list' is configured with a stop-on-busy option.

Preemption and CAC: locations and regions govern bandwidth per call; a call that exceeds the location bandwidth is either rejected or, with a configured AAR group, rerouted through the PSTN.

Extension Mobility: a user logs in to a phone, which downloads the device profile; the physical phone MAC is mapped to a device profile, and EM uses the Cisco Extension Mobility service on CUCM.

Device Mobility: a phone in a remote site uses the device mobility group to select the correct location, region, and CSS based on the IP subnet of the phone and the device mobility info.

Mobile Connect (SNR): an incoming call to the desk phone rings the remote destination (mobile) and the desk phone; answer on either leg, and the remote destination can be configured with a delay.

Mobile Voice Access: an IVR allows a user to place calls and change Mobile Connect settings from a remote phone; it uses the application dial rules and a CTI route point.

Hunt groups: broadcast, longest idle, circular, and top-down algorithms distribute calls; hunt pilots have a CSS and a queue/agent configuration.

Call Park: a call is parked on a park slot number and retrieved by dialing the slot; the park range is configured in the CUCM cluster-wide call park settings.

Intercom: a one-way call between two users; the called party hears beep and the call is auto-answered; intercom uses a dedicated directory number and 'intercom' as the line type.

Presence and BLF: BLF uses a phone button template and a presence group; the presence group controls which users can see each other's status.

Call forwarding types: CFNA, CFB, CFU; each uses a CSS that determines the forwarding destination reachable by the user.

Use 'show dial-peer voice summary' to verify the matched dial-peer; the 'in' and 'out' columns show the inbound and outbound peer for a test call.

Use 'debug voice ccapi inout' to trace call setup, dial-peer matching, translation, and disconnect cause codes; 'debug isdn q931' shows the ISDN cause value.

Common disconnect causes: 16 normal call clearing, 21 call rejected, 34 no circuit/channel available, 38 network out of order, 47 resource unavailable.

CUCM traces: use the Real-Time Monitoring Tool (RTMT) to collect traces; the SDL and CCM traces show the call flow, and the 'Call Control' trace shows digit analysis.

Digit analysis: in CUCM, a route pattern with 'Urgent Priority' unchecked is not used if a longer match exists; check the 'Called Party Number' transformation and the CSS/partition.

For SIP, check the 'Reason' header and the SIP response code; 404 not found means no route pattern matched, 503 service unavailable means no gateway available.

Check 'show sip-ua status' and 'show sip-ua calls' on the gateway; verify the SIP trunk registration and the 'session target' on the dial-peer.

Media issues: verify the codec, the MTP/transcoder resource, and the region bandwidth; a 'no media' symptom often means a codec mismatch or a missing MTP.

For EM login failures, check the device profile, the EM service, and the user's associated device profile; the phone must have the EM service subscribed.

Cisco Jabber: uses the UDS or SOAP service to discover the CUCM; the service profile defines the voicemail, directory, and IM&P servers.

Mobile and Remote Access (MRA) uses Expressway-C and Expressway-E; the internal Expressway-C connects to CUCM, and the external Expressway-E is in the DMZ.

MRA requires a traversal zone between Expressway-C and Expressway-E, and a SIP/HTTPS zone between Expressway-C and CUCM.

VPN access: a remote phone can use a VPN client or the AnyConnect VPN; the phone uses the VPN to reach CUCM through the ASA.

Cisco Unified Mobility: the remote destination is configured on the CUCM, and the Mobile Connect feature uses the remote destination profile.

Certificate requirements for MRA: the Expressway-E certificate must be trusted by the client, and the certificate must include the SIP domain and the Expressway-E FQDN in the SAN.

The Expressway-E uses a public certificate; the Expressway-C uses an internal certificate signed by the CUCM certificate authority.

Remote phone firmware: the phone downloads firmware from the TFTP server; for MRA, the phone uses the Expressway-E as the proxy for TFTP.

CUCM partitions and CSS enforce calling privileges: a CSS is a list of partitions, and a device can only dial numbers in partitions in its CSS.

Time-of-day routing uses a time schedule and a time period; a route pattern can be assigned a time schedule to allow calls only during certain hours.

Toll fraud prevention: use a CSS on the gateway that blocks international patterns, and use 'forbidden' patterns in a partition not in the CSS.

SIP security: TLS for signaling and SRTP for media; configure a secure SIP trunk with 'srtp' and a certificate trust.

Phone security: a secure phone uses a certificate (MIC) and an encrypted TFTP configuration; the phone must be in a secure cluster.

The 'SIP trunk security profile' defines the TLS port, the digest authentication, and the incoming/outgoing transport.

Application dial rules and directory lookup dial rules control how numbers are dialed from the directory; they are applied before the CSS.

Cisco UCM 'Enterprise Parameters' include the 'Cluster Security Mode' (mixed or secure) and the 'Certificate Expiry' monitoring.

For 802.1X, the phone uses EAP-TLS or EAP-FAST; the switch port must be configured for multi-domain authentication.