The exhibit shows a transcoder profile whose codec list contains only G.711 µ-law, G.711 A-law, G.729r8 and G.729ar8.

A DSP-farm can transcode only between codecs that are explicitly registered in that profile.

Therefore, the permutations that must—and can—be handled by this DSP-farm are:

• G.729r8 ↔ G.711 µ-law (C)

• G.729ar8 ↔ G.711 A-law (E)

All other listed permutations include a codec (iLBC, G.728br8, G.722) that is absent from the profile and hence are not eligible for transcoding by this resource.

A. iLBC is not in the profile; the DSP-farm cannot advertise or transcode it.

B. G.728br8 is unsupported/unregistered in the shown profile; no transcoder match.

D. G.722 (wide-band) is missing from the profile and not supported by IOS-based narrow-band DSP transcoders in CUCM 12.x.

1. Cisco Unified Communications Manager Features and Services Guide, Release 12.5 – “Audio Transcoding in DSP Farms”, Table 10-2 Supported Codecs (G.711 µ-law, G.711 A-law, G.729, G.729a) – cisco.com, p. 10-19.

2. Cisco IOS XE Voice Infrastructure Services Configuration Guide, 17.3 – “Configuring DSP Farm Transcoding”, section “codec pass-through and negotiation”, lists same supported codecs; G.722, G.728, iLBC absent – cisco.com, §“Supported Transcoding Codecs”.

3. Cisco Unified Communications SRND, Release 12.x – Ch. 5 “Media Resources”, “Transcoder Codec Support”, confirms only permutations between registered codecs are selectable – cisco.com, p. 5-21.

(Each reference publicly available on Cisco.com; page numbers refer to PDF versions.)

In ISDN Q.931 / Q.850 signalling, Cause Number 1 is defined as “unallocated (unassigned) number.”

The network switch cannot find any endpoint or directory entry that owns the dialed DNIS, so it immediately releases the call with cause 1. This exactly matches option D: “The dialed number is not assigned to an endpoint.” Media resources, user-busy, or call-rejection conditions use different cause numbers (16-series or 20-series).

A. Media resource problems occur after call setup; no ISDN cause code corresponds—cause 1 is issued before media allocation.

B. “Destination rejects the call” maps to Cause 21 (Call rejected), not 1.

C. “Destination number is busy” maps to Cause 17 (User busy), not 1.

1. ITU-T Recommendation Q.850 (02/1998), §3.2, Table 1 – Cause 1: “Unallocated (unassigned) number.”

2. Cisco Systems, “Understanding and Troubleshooting ISDN Call Disconnect Problems,” Doc ID 10036, Table “Cause Codes,” cause 1 – Unallocated/unassigned number.

3. Cisco Press, Caslow et al., “Cisco Voice Gateways and Gatekeepers,” 2001, Ch. 3 “ISDN Call Control,” p. 83 – list of Q.931/Q.850 cause codes shows cause 1 as unallocated number.

CUCM treats any account imported from an external LDAP directory as read-only. When that account is removed from LDAP, the next DirSync cycle only changes its state in CUCM to “Inactive LDAP Synchronized User.” Physical deletion is handled by the database garbage-collector task, which runs automatically once every 24 hours (default start time 01:00 server local time). Administrators therefore simply wait for the garbage-collector to purge the inactive record; no manual deletion or service restart is required or supported for LDAP-synchronized users.

A. CUCM disables the Delete button for LDAP-synchronized users; manual deletion is blocked.

C. Restarting the DirSync service refreshes status but does not trigger the garbage-collector, so the user remains.

D. A manual sync again just flags the user as inactive; only the nightly garbage-collector removes it.

1. Cisco Systems, “Administration Guide for Cisco Unified Communications Manager, Release 12.5(1),” Ch. 11 ‘Directory Integration’, sec. ‘Deleting LDAP Synchronized Users’; pp. 11-31–11-32.

2. Cisco Systems, “Features and Services Guide for Cisco Unified Communications Manager, Release 14,” Ch. ‘Directory Synchronization’, para. ‘Garbage-Collection Process’ (page 1420).

3. Cisco Press. Kapur et al., “Implementing Cisco Unified Communications Manager, Part 1,” 2nd ed., 2012, Ch. 6 ‘LDAP Integration’, pp. 414-416 (explanation of inactive user flag and nightly purge).

When Unity Connection communicates with Cisco UCM across a SIP trunk, it must be allowed to send unsolicited NOTIFY messages for Message-Waiting Indicator (MWI) updates and to perform supervised transfer operations that use the SIP Replaces header. Therefore the SIP trunk security profile applied to the Unity Connection trunk must have the options “Accept Unsolicited Notification” and “Accept Replaces Header” enabled.

A. Accept presence subscription – Presence is not used in the Unity Connection voice-mail integration; this setting is unnecessary.

B. Allow charging header – Pertains to SIP charging information for billing; irrelevant to Unity Connection.

D. Enable application-level authorization – Used for SIP application tokens; Unity Connection does not require it for basic voice-mail integration.

1. Cisco Unity Connection and Cisco Unified Communications Manager Integration Guide for SIP, Release 14, “Create a SIP Trunk Security Profile,” Step 6 – lists required boxes: Accept Replaces Header & Accept Unsolicited Notification (pp. 17-18).

2. Cisco Unified Communications Manager Features and Services Guide, Release 14, “SIP Trunk Security Profile Field Descriptions,” explains functions of Accept Replaces Header and Accept Unsolicited Notification (Section “SIP Normalization and Transparency,” pp. 321-322).

Cisco Prime Collaboration requests network devices to send SNMP notifications (Traps and InformRequests). By definition and in Cisco Prime Collaboration Assurance configuration, these unsolicited SNMP messages are delivered to the manager over UDP port 162. Prime must therefore listen on UDP/162 to receive the notifications it solicits.

A. UDP 161 – Used for SNMP polling (Get, Get-Next, Set), not for traps/informs.

B. TCP 161 – TCP transport is optional and for polling; traps/informs still default to UDP 162.

D. TCP 80 – Standard port for HTTP web traffic, unrelated to SNMP or Prime notification delivery.

1. Cisco Prime Collaboration Assurance and Analytics Administration Guide, Release 12.1, “Configuring SNMP,” p. 18: “PCA listens for SNMP traps and inform requests on UDP port 162.”

2. Cisco Prime Collaboration 11.x Quick Start Guide, Appendix B “Port Utilization,” p. B-4: “SNMP Traps/Informs – UDP 162 inbound to Prime.”

3. Cisco Systems, “Simple Network Management Protocol (SNMP) Overview,” Technical Note, 2019, p. 2: “Managers receive traps and inform requests on UDP port 162.”

4. Case, J. et al., RFC 3412 – SNMP Message Processing and Dispatching, §5: “All SNMP engines use UDP/162 as the well-known port for traps and informs.”

Unity Connection determines the subscriber mailbox from the SIP Diversion/Redirecting number delivered by CUCM.

If the SIP trunk toward Unity Connection is not configured to send this header, Unity Connection cannot match the call to a user and therefore plays the default (Opening) greeting.

Enabling “Redirecting Diversion Header Delivery – Outbound” on the CUCM-to-Unity Connection SIP trunk causes CUCM to include the necessary header. A trunk reset is required for the new setting to be applied to active call legs.

A. Rebooting Unity Connection does not change SIP header delivery; problem is in CUCM trunk configuration.

C. Checking the box without resetting leaves existing trunk instances running with old settings; change will not take effect until a reset.

D. Logs aid troubleshooting but do not modify signaling; voicemail mismatch persists.

1. Cisco “Integration Guide for Cisco Unity Connection 12.x and Cisco Unified Communications Manager using SIP”, Section ‘Configuring the SIP Trunk’, steps 12-14, pp. 6-7 – requirement to enable “Redirecting Diversion Header Delivery—Outbound”; states Opening Greeting played when unchecked.

2. Cisco Unity Connection 12.x “Troubleshooting Guide”, Topic ‘Caller Hears Opening Greeting Instead of Subscriber Greeting’, para. 2 – confirms Diversion header necessity.

3. Cisco Unified Communications Manager Administration Guide 14, Chapter ‘Resetting and Restarting Devices’, subsection ‘Resetting a SIP Trunk’ – reset required for configuration changes to take effect on the trunk.

When Cisco Expressway is deployed in the standard core/edge pair (Expressway-C inside, Expressway-E in the DMZ), it delivers two flagship functions.

1. Firewall-traversal for B2B and MRA traffic, providing a highly secure path through enterprise firewalls (option A).

2. Mobile and Remote Access (MRA), which lets off-premises users register to CUCM and use voice, video, IM&P, and directory services through a TLS–SRTP tunnel—no separate VPN client is required (option D).

B. Expressway offers basic call detail records; it is not designed for deep edge-traffic visibility or analytics.

C. Local registration/monitoring is still performed by CUCM; Expressway only proxies remote endpoint registrations, so coverage is not “complete.”

E. Utilization/adoption metrics come from Webex Control Hub, CUCM, or analytics tools, not from Expressway.

1. Cisco Expressway Series Data Sheet, “Features and Benefits – Secure firewall traversal; Mobile and Remote Access,” pp. 1-2, Cisco Systems, 2023.

2. Cisco Unified Communications Mobile and Remote Access via Cisco Expressway Deployment Guide, v14, Sec. 2.1 “Solution Benefits,” Cisco Systems, 2023.

3. Cisco Expressway Administrator Guide, X14, Ch. 20 “Call Detail Records” (limited reporting only), Cisco Systems, 2023.

For SIP phones the transport protocol that CUCM actually uses (UDP, TCP, or TLS) is taken from the device’s “Phone Security Profile.”

If the phone is still assigned to a Secure profile (TLS), CUCM will attempt to signal on TCP/5061. Because the administrator changed CUCM to use UDP, the signaling ports/protocols no longer match and the phone cannot set up calls.

Assigning or editing the phone to a Non-Secure SIP Phone Security Profile whose Transport Protocol is UDP restores registration and call establishment.

A. Common Phone Profile controls soft-key, feature and firmware parameters, not signaling transport; changing it will not affect UDP/TCP/TLS selection.

B. SIP Dial Rules manipulate digit strings dialed by the user; they have no impact on SIP transport.

C. SIP Profile adjusts timers, message contents and options; the transport protocol continues to be governed solely by the phone security profile.

1. Cisco Unified Communications Manager Security Guide, Release 14, “Configure SIP Phone Security Profiles,” Table 2-1: Device Security Mode vs. Transport Protocol, pp. 2-9–2-10.

2. Cisco Unified Communications Manager Administration Guide, Release 14, “Adding a SIP Phone,” step 7: “Select the appropriate Phone Security Profile; this setting determines whether UDP/TCP (Non-Secure) or TLS (Secure) is used,” pp. 2663-2664.

3. Cisco Unified Communications Manager Security Guide, Release 12.5(1), Chapter 3 “How CUCM Determines SIP Transport,” para. 2.

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