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Asterisk PJSIP maintenance mode is missing, stale or calls still start

Trace release support, AMI framing, endpoint scope, node ownership and actual SIP or channel behavior before repeating or widening maintenance actions.

Reviewed 2026-10-21Current branch-20, branch-22, certified-22.8 and branch-23 PJSIP maintenance action, status-event and completion-event references plus current AMI, Dial and upgrade guidance were reviewed October 21, 2026. The action is documented as introduced in Asterisk 20.20.0, 22.10.0 and 23.4.0, so installed release, distributor patches, loaded modules, manager actions, permissions, endpoint names, active dialogs, node scope, reload and restart behavior remain deployment specific. No PBX, module, manager account, endpoint, dialog, registration, SIP request, channel, ARI operation, packet, route, media stream or live call was inspected or testedAdvancedDocumentation based · not lab tested

Freeze one action, one snapshot and one behavior sample

Record node, release, loaded modules, manager identity and permissions, exact endpoint, requested state, ActionID, initial response, status event and completed show transaction. Separately retain one authorized inbound or outbound attempt with timestamps and the node that handled it.

Classify the first divergence: action missing or denied; endpoint rejected; request accepted but no event; snapshot incomplete; state differs; wrong node queried; new traffic behaves unexpectedly; or an existing dialog is being mistaken for a new origin.

Resolve missing actions before editing endpoints

The maintenance action is documented from Asterisk 20.20.0, 22.10.0 and 23.4.0. Verify the installed patch release, package provenance, loaded res_pjsip components and AMI permissions. A documentation page for the major branch is not evidence that an earlier maintenance release contains the action.

If the action is unknown or denied, correct the supported release, module state or least-privilege manager permission through the normal owner. Repeating it or switching to Endpoint: all does not repair capability.

Prove endpoint and node scope

Use the exact endpoint name and a specific-endpoint PJSIPShowMaintenance request. The unfiltered action lists endpoints currently in maintenance mode; absence there is not proof the endpoint is absent. Retain the terminal PJSIPMaintenanceStatusComplete event before calling the snapshot complete.

On multi-node systems, query every relevant node separately and prove which node receives the SIP request or originates the call. Do not infer cluster-wide state from one manager session.

Distinguish stale telemetry from stale state

Manager event filters, disconnected sessions, reused ActionID values and delayed consumers can hide or misassociate PJSIPMaintenanceStatus. Compare the initial response with a fresh completed show transaction rather than trusting a cached event.

After reload or restart, obtain a new snapshot. The reviewed documentation defines the runtime action and events but does not establish universal persistence across every reload, process restart or distribution.

Read call behavior against the documented boundary

For a new inbound out-of-dialog request, correlate maintenance state with the documented 503 Service Unavailable; remember the SUBSCRIBE and REGISTER exception when Expires: 0. For outbound testing, separate a new Dial() or ARI origin from a call already in progress.

In-progress dialogs are documented as unaffected. A surviving established call is not evidence that maintenance failed, while an unrelated 503, dialplan rejection, registration loss or unreachable contact is not proof maintenance caused the result.

Correct the first owner and verify restoration

Correct only the demonstrated layer: supported release, module load, manager privilege, event delivery, endpoint name, node selection, action state or traffic test. Re-run one exact-endpoint set request and one completed show snapshot before repeating authorized behavior checks.

After the work, set the endpoint off, verify the status change and a fresh snapshot, then confirm expected new inbound and outbound behavior. No PBX, endpoint, dialog, SIP request, channel, ARI operation, packet, media stream or live call was inspected or tested.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

Examples require adaptation to your topology. No live PBX or hardware testing is claimed.

Continue your research

Guide · Current branch-20, branch-22, certified-22.8 and branch-23 PJSIP maintenance action, status-event and completion-event references plus current AMI, Dial and upgrade guidance were reviewed October 21, 2026. The action is documented as introduced in Asterisk 20.20.0, 22.10.0 and 23.4.0, so installed release, distributor patches, loaded modules, manager actions, permissions, endpoint names, active dialogs, node scope, reload and restart behavior remain deployment specific. No PBX, module, manager account, endpoint, dialog, registration, SIP request, channel, ARI operation, packet, route, media stream or live call was inspected or tested

Asterisk AMI PJSIP maintenance mode and status evidence

Use versioned AMI actions to drain one PJSIP endpoint without treating command acceptance, status events or SIP rejection as the same proof.

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Troubleshooting · Asterisk branch-23-generated and branch-22-generated ListCommands references plus official AMI syntax, v2 specification, manager-command, change/versioning, permission, module-loader and network-security documentation reviewed October 16, 2026; installed branch and maintenance release, build, package and module set, manager user, read and write authorizations, session mask, action registration, request framing, parameters, target objects, response, related events and service outcome remain deployment specific; no PBX, module, package, manager user, manager session, request, response, event, configuration file, channel, packet, media stream or live call was inspected or tested

Asterisk AMI action is missing, unknown or permission denied

Trace one unavailable AMI action through session identity, user-scoped inventory, administrator registry, release and module support, authorization, framing and target behavior.

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Troubleshooting · Current Asterisk branch-23-generated Events, Filter, WaitEvent, Login and Ping action references, matching branch-22 Events, Filter and WaitEvent references, the official AMI event-filtering guide and Manager TCP/IP API reviewed October 16, 2026; installed branch, manager user permissions, login mask, static and dynamic filters, connection type, event queue, client parser, reconnect behavior, action concurrency, downstream storage and call activity remain deployment specific; no PBX, AMI session, manager user, permission, configuration file, socket, HTTP session, event queue, channel, bridge, packet, media stream or live call was inspected or tested

Asterisk AMI events are missing, filtered, delayed or duplicated

Locate the first divergence among manager permissions, the per-session event mask, configured and dynamic filters, HTTP WaitEvent queueing, TCP reads, reconnects, parsing and downstream deduplication.

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Troubleshooting · Asterisk 22 LTS; PJSIP endpoint maintenance controls require 22.10.0 or later

Asterisk PJSIP returns or receives SIP 503 Service Unavailable

Identify the responding SIP leg, then separate maintenance mode, local channel creation, registration retry and an upstream service response.

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Guide · Current Asterisk PJSIP_ENDPOINT and CHANNEL function documentation, res_pjsip endpoint configuration reference, object-relationship guidance and PJSIP troubleshooting guidance reviewed October 13, 2026; PJSIP_ENDPOINT is documented since Asterisk 12.0.0, while installed branch, maintenance release, loaded modules, configuration backend, object names, defaults, endpoint identification, channel association, contacts, signaling and media remain deployment specific; no PBX, module, configuration, endpoint, identify object, channel, SIP exchange, packet or call was inspected or tested

Asterisk PJSIP endpoint inspection: configuration fields, channel identity and proof

Read one loaded endpoint field without confusing configuration, inbound endpoint identification, AoR location, a live channel or negotiated SIP and media state.

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Guide · FreePBX 17 / Debian 12 editorial baseline; Sangoma Asterisk SIP Settings guidance, FreePBX sipsettings release/17.0 source and current Asterisk PJSIP configuration, NAT, transport-selection and Asterisk 22 option references reviewed October 11, 2026; older Sangoma screenshots and field descriptions are semantic guidance rather than an exact map of every installed module build; installed FreePBX, sipsettings and Asterisk versions, saved values, generated files, loaded objects, listeners, firewall, DNS, certificates, endpoints, trunks, registrations, SDP, RTP and calls require local evidence, and no PBX or call was inspected or tested

FreePBX Asterisk SIP Settings: transport, NAT and media boundaries

Map each GUI control to generated PJSIP transport, address, codec, media or TLS state, then prove the loaded listener and call path after activation.

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