The engine beneath the phone system
Asterisk supplies building blocks for call control and media applications. An administrator combines channel configuration, dialplan rules and feature modules into a working system. It offers substantial flexibility, but the software alone does not supply a carrier agreement, an operating plan or a production support team.
Follow a controlled learning sequence
Begin with PBX fundamentals and a self-hosting plan. Establish the network boundary before installing. The source-build workflow now pins the release artifact and keeps compiled, installed, running and call-accepted states separate. Create internal extensions before connecting a trunk. The dialplan guide now separates the channel's entry context, exact and pattern matching, included contexts, loaded priorities and completed-call evidence before outside routes are added.
The standalone baseline is Asterisk 22 LTS, which the project lists as fully supported through October 16, 2028, with security-fix maintenance through October 16, 2029. Release support does not prove that a host has a current maintenance build. FreePBX-managed systems have a separate configuration lifecycle; do not apply standalone file examples over generated FreePBX settings.
Know what to verify
For each change, identify the expected effect, the observation that would confirm it and the rollback. Test call setup, sound in both directions and hangup. Keep version, module and configuration notes with your results. Asterisk CLI output from your own system is evidence; an illustrative snippet in a guide is not a test record. The CLI inventory guide separates loaded runtime state from files, while the logging guide bounds sensitive trace collection and cleanup.
For inbound PJSIP, record how a request is identified before authentication and dialplan routing. The endpoint-identification guide separates the global method order from each endpoint’s permitted methods.
For contact monitoring, the qualification guide keeps OPTIONS reachability separate from registration, authorization, media and completed-call evidence. For queues, the member-state and caller-outcome guide separates membership, device state, pause scope, attempted delivery, an answered bridge and the queue log. For recording, the MixMonitor lifecycle guide keeps policy, channel ownership, file format, stop and post-processing boundaries explicit.
For protected media, the PJSIP media-encryption guide keeps SIP transport TLS, SDES or DTLS-SRTP keying, loaded endpoint policy, negotiated SDP, packet protection and each call leg's termination point separate.
Treat digest authentication as a two-role exchange
The new PJSIP digest-authentication guide separates Asterisk's inbound UAS auth role from its outbound UAC outbound_auth role, and binds realm, algorithm policy and credential form to the running build. Asterisk added SHA-256 and SHA-512-256 digest support in 20.12.0, 21.7.0 and 22.2.0 with PJProject 2.15.1, but installed build options and peer interoperability still control what one exchange can use.
Treat a Local channel as a temporary pair by default
The Local-channel lifecycle guide maps the ;1 and ;2 halves, default post-answer optimization, the bounded reasons for /n, modifier ownership, bridge events and record changes. It keeps a normal optimized disappearance distinct from failure and does not make retained Local pairs a universal configuration pattern. If a variable or Dial() behavior disappears after answer, the symptom-led diagnosis distinguishes optimization from inheritance, placement and a feature that truly requires retention.
Keep each called-channel cause attached to its owner
The hangup-cause guide separates DIALSTATUS, each HANGUPCAUSE_KEYS() channel, its technology and translated Asterisk causes, CHANNEL(hangupsource) and the original SIP exchange. This matters for parallel and sequential attempts: one final status or CDR disposition can hide losing branches, an earlier trunk, or several different wire responses that map to the same Asterisk cause.
Separate device state, hint aggregation and presence
The device-state and hints guide treats a provider's value, an extension hint's aggregate, a user's presence choice and the SIP subscription that carries updates as separate evidence. This prevents a BLF lamp, watcher count or registered contact from being used as proof of reachability, routing or a completed call.
Keep taskprocessor backlog attached to its named owner
When Asterisk reports that a taskprocessor queue reached its high-water mark, the taskprocessor overload diagnosis preserves the named queue, depth, water marks, processing time and producer workload before a reload or restart. It also keeps a non-PJSIP queue that triggers global PJSIP admission control separate from a queue actually owned by the PJSIP subsystem.
Keep call pickup groups and targets explicit
The Asterisk call-pickup guide separates general group pickup, directed extension pickup and channel targeting. It preserves the requesting channel's pickup groups, the ringing outbound channel's call groups, target context and time-sensitive ringing state as separate evidence.
Separate transfer engines and bridge outcomes
The Asterisk transfer guide distinguishes core in-bridge feature codes, phone-native SIP transfer and dialplan applications. It preserves party permission, transfer context, attended consultation, transfer events and the final bridge as separate evidence.
Trace PJSIP registrations through the AoR
The PJSIP AoR and contacts guide keeps endpoint policy, AoR admission, live contacts, qualification state and first-versus-all-contact dialing distinct.
Trace PJSIP transport selection from URI to connection
The PJSIP transport-selection guide separates URI scheme and parameter, DNS target, explicit or implicit transport object, existing TCP or TLS connection and per-object proxy routing.
Route each PJSIP request owner through a proxy explicitly
The PJSIP outbound-proxy guide keeps endpoint requests, AoR qualification and outbound REGISTER routing on their separate objects. It also preserves the registrar or contact identity, loose-route URI, inbound identification and media topology instead of treating a working registration as proof that every method used the proxy.
Relate outbound registrations to returning calls explicitly
The registration line-matching guide keeps the registered address, Contact user, returned line token, selected endpoint, endpoint context and called extension as separate evidence.
Follow PJSIP session timers per dialog leg
The Asterisk PJSIP session-timer guide separates loaded endpoint policy from the interval, refresher and re-INVITE or UPDATE actually negotiated on each Asterisk dialog.
Prove the final PJSIP media topology
The PJSIP direct-media guide separates loaded endpoint permission, the in-dialog re-INVITE or UPDATE, accepted SDP and observed RTP on both directions. Asterisk disappearing from a capture can be a valid final topology or the first visible failure; the evidence decides which.
Trace DTMF across both PJSIP legs
The PJSIP DTMF guide separates endpoint mode, SDP or SIP signaling, Asterisk receipt, bridge ownership, egress representation and application consumption. Use the focused mismatch diagnostic when Asterisk sees a digit but a far IVR ignores or doubles it.
Keep ARI digit events attached to channel state
The ARI DTMF guide separates source-leg transport, the end-of-digit ChannelDtmfReceived event, bridge decoding, prompt interruption and outbound digit injection. Use the ARI DTMF diagnostic when events are absent, repeated or delivered outside the current input state.
Keep ARI variables attached to scope and channel lifetime
The ARI channel-variable guide separates channel-scoped and global values, explicit reads, configured snapshot visibility, reported updates, child-channel inheritance and Asterisk version. Use the focused diagnostic when a value is absent, stale or present on the wrong call leg.
Separate ARI push configuration from its storage and effect
The ARI push-configuration guide follows one case-sensitive Sorcery object through its configuration class, object type, ID, field tuples, selected persistence wizard, read-back and supplied PJSIP behavior. Use the focused diagnostic when GET, PUT or DELETE disagrees with the backend, survives the wrong lifecycle or does not produce the expected endpoint behavior.
Give each ARI device state one explicit owner
The ARI device-state guide separates provider-owned state from application-controlled state, exact resource names, subscriptions, events, extension hints and downstream display. Use the focused diagnostic when a custom state is stale, an event is missing or a hint or lamp disagrees.
Treat ARI event delivery as its own control plane
The ARI events and subscriptions guide separates application registration, resource subscription, subscribeAll, event-type filtering, WebSocket ownership and client processing. Use the focused event diagnostic when expected events disappear, arrive twice or move to another connection.
Trace early media and PRACK per PJSIP leg
The Asterisk PJSIP early-media guide separates 180, 183, reliable provisional responses, SDP offer/answer, early RTP and final answer. Use the focused diagnostic when ringback disappears or PRACK stalls.
Trace codec negotiation across both PJSIP legs
The PJSIP codec-negotiation guide maps configured, pending and resolved topologies across all four offer-and-answer control points.
Separate PJSIP SDP addresses from observed RTP paths
The PJSIP media-address guide separates endpoint SDP creation, transport rewriting, RTP source binding, symmetric return media and SIP Contact handling. Use the wrong-media-address diagnostic when the negotiated and observed paths diverge.
Treat STIR/SHAKEN as a versioned signaling subsystem
The Asterisk STIR/SHAKEN guide separates loaded attestation, verification, TN and effective-profile state; protects signing material and certificate retrieval; and keeps a verification result distinct from the call policy applied to it.
Trace PJSIP DNS from URI to live state
The PJSIP DNS-resolution guide separates URI and proxy ownership, SRV selection, resolver configuration, TTL, connection reuse and outbound-registration retry. Use the focused diagnostic when Asterisk appears to keep an old target.
Design dual-stack PJSIP transports explicitly
The IPv4 and IPv6 transport guide keeps local bind, address family, URI, DNS candidate, TLS identity and RTP path separate. Use the focused diagnostic when PJSIP selects IPv6 but the target is unreachable.
Keep release status attached to an observation date
The Asterisk release-lifecycle guide records the October 10, 2026 branch state without treating future dates or release candidates as completed transitions. The project had published second release candidates for 20.22.0, 22.12.0, 23.6.0 and 24.0.0 on October 8; those artifacts had not replaced the latest final releases. The guide separates upstream support, final and pre-release point releases, Certified Asterisk, package or appliance ownership and a running installed build, then keeps point updates distinct from rehearsed major-version upgrades. Use the upgrade-regression diagnostic when a module, configuration or call path changes after an update.
Apply PJSIP ACLs at the intended boundary
The PJSIP ACL guide separates module-wide source rules, endpoint rules, named lists and REGISTER Contact addresses. Use the paired diagnostic before widening a permitted network.
Put outbound SIP headers on the channel that sends them
The PJSIP outbound-header guide separates caller and callee ownership, direct and inheritable header stores, request and response phases, and the 20.19.0, 22.9.0 and 23.3.0 version floor for PJSIP_INHERITABLE_HEADER(). Use the paired diagnostic before adding a header in several contexts.
Separate application media from SIP and ARI WebSockets
The Asterisk media-over-WebSocket guide keeps SIP signaling, ARI control and chan_websocket binary media on separate connections, with explicit release, frame, codec and backpressure boundaries. Use the paired diagnostic when the application channel is silent, one-way or disconnected.
Use call files as bounded queue submissions
The Asterisk call-file guide separates atomic filesystem handoff, scheduling, channel attempts, answered actions and archived status. Use the paired diagnostic when a file waits, disappears or produces no expected downstream result.
Operate AMI Originate as a correlated state change
The AMI Originate guide separates action acceptance, the correlated originate event, first-leg state and downstream workflow result. Use the paired diagnostic when an action is rejected, its event is missing or the wrong call stage is being counted as success.
Originate ARI channels with explicit lifecycle gates
The ARI channel-origination guide compares direct originate with create-then-dial and keeps HTTP acceptance, Stasis ownership, answer, bridge membership, media and application result as separate evidence.
Keep External Media as a channel and bridge participant
The ARI External Media guide treats RTP, AudioSocket and WebSocket as versioned driver choices. Channel creation, Stasis ownership, bridge membership, outbound media, inbound media and external processing remain separate evidence.
Keep ARI bridge ownership explicit
The ARI bridges guide separates bridge creation, Stasis ownership, membership, media topology, playback or recording and cleanup. A bridge ID or successful add request is not evidence that the intended call legs remained connected or exchanged media.
Keep ARI playback identity and audible outcome separate
The ARI playbacks guide follows one application-owned playback ID through media resolution, target scope, queueing, events, control and listener evidence. A successful REST response or finish event is not by itself proof that the intended endpoint rendered the prompt.
Keep ARI recording scope and stored state separate
The ARI recordings guide distinguishes channel input from bridge-mixed capture, then follows one application-owned recording name through live state, events, controls, stop or cancel, stored inventory and decoded content. A start or finish event is not proof that the stored file contains the intended participants or directions.
Bound AGI execution and external waits
Use the AGI and FastAGI guide to choose local process, TCP service, EAGI audio or AsyncAGI command ownership deliberately. Keep synchronous channel wait, launch or connection success, protocol progress, application result and caller-visible outcome as separate evidence states; use the paired diagnostic when one script fails or retains a channel.
Keep connected-line and redirecting identity separate
The PJSIP connected-line and redirecting guide separates caller, connected and forwarded-party state across answer and transfer updates. Use the paired diagnostic when Asterisk or a phone shows the wrong remote party.
Keep CEL selection, ordering and storage independently visible
When channel events are absent, repeated or appear misordered, use the CEL diagnostic. It preserves engine selection, per-Primary ordering, cross-channel correlation, backend acceptance and downstream import as separate evidence.
Diagnose queue-log gaps at the correct layer
When a queue event, interval or call attribution is missing, use the queue-log diagnostic. It separates Queue admission, logger policy, file and realtime targets, event identity, rotation and downstream import.
Follow Gosub stack and channel ownership
The Gosub subroutine guide separates the stack frame, arguments and return value from the caller or callee channel that owns each effect. Use the focused diagnostic when one invocation does not resume or changes the wrong leg.
Keep teardown on the owning channel
The hangup-handler guide separates the per-channel LIFO stack from the h extension, final cause and later records. Use the focused diagnostic when execution is missing, duplicated, out of order or slow.
Prove what happens after one channel leaves a bridge
The after-bridge dialplan guide separates channel-owned after_bridge_goto, normal Dial() continuation and party-transfer options. Use the focused diagnostic when the expected surviving channel does not enter its target.
Attach talk detection to the channel that owns the media
The TALK_DETECT guide separates channel scope, silence time, talking magnitude, audiohook order and AMI or ARI event evidence. Use the focused diagnostic before tuning missing, early or late events.
Count a declared channel lifecycle, not an assumed call
The channel-groups guide separates Asterisk channel membership from conversation count, provider entitlement and platform capacity. Use the paired diagnostic when a local guard is high, low or bypassed.
Separate BackGround playback, WaitExten and digit matching
The BackGround and WaitExten guide follows an IVR digit through prompt playback, DTMF reception, destination context, extension matching and timeout. Use the focused diagnostic when the prompt plays but the selected branch does not run.
Dial every current PJSIP contact
The PJSIP_DIAL_CONTACTS guide separates endpoint, AoR, contact inventory, generated dial string, outbound channels and first answer. Use the focused diagnostic when only one registered device—or none—rings.
Override one outbound PJSIP media offer on the correct channel
The PJSIP_MEDIA_OFFER guide separates callee pre-dial ownership, requested audio or video codecs, outbound SDP, endpoint negotiation, session refresh and usable media. Use the focused diagnostic when the offer is unchanged or the call fails.
Parse one PJSIP URI without turning syntax into trust
The PJSIP_PARSE_URI guide separates the source header or variable, direct and variable-name function forms, documented absence values, authorization and final routing. Use the focused diagnostic when a component is blank or unexpected.
Turn a device hold request into a channel event
The HOLD_INTERCEPT guide separates channel attachment, suppressed hold frames, AMI or ARI events, consumer actions and peer media. Use the focused diagnostic when the event or hold result differs.
Keep res_fax modules, session policy and result evidence together
The Asterisk res_fax guide separates the generic resource layer, technology resource, channel-scoped FAXOPT settings, TIFF/F artifact, audio or T.38 mode and final session result. Use the focused diagnostic when an application fails or selects the wrong mode.
Read RTCP quality evidence on the correct channel and stream
The Asterisk RTCP statistics guide keeps channel ownership, audio or video stream, local observations, remote reports, bridge topology and MES direction explicit. Use the focused diagnostic when values are empty or conflict with the reported experience.
Read answered PJSIP response headers on the callee
The PJSIP response-header guide keeps the outbound INVITE, 200 response, callee channel, U() pre-bridge timing and repeated field instance explicit. Use the focused diagnostic when the value is empty or attached to the wrong response.
Inspect one PJSIP AoR and contact without flattening the objects
The AoR and contact inspection guide follows endpoint association, AoR fields, internal contact identifiers and individual contact fields. Use the focused diagnostic when the result is empty, stale or attached to the wrong object.
Inspect one PJSIP endpoint without flattening policy and runtime state
The PJSIP endpoint inspection guide follows a loaded endpoint field, current channel owner, associations, defaults and later SIP evidence. Use the focused diagnostic when the result is empty, missing or attached to the wrong leg.
Override one PJSIP DTMF mode without changing endpoint policy
The PJSIP_DTMF_MODE guide separates the endpoint setting, one channel value, signaling method, digit events and downstream consumer. Use the focused diagnostic when a runtime write is ignored or digits still fail.
Choose core or ARI ownership for an incoming SIP REFER
The PJSIP transfer-handling guide separates endpoint permission, one channel mode, ARI event delivery, application action and the final transfer. Use the focused diagnostic when the event is absent or an accepted transfer stalls.
Set one From parameter without replacing the whole header
The PJSIP_HEADER_PARAM guide separates URI placement, header placement, caller and callee ownership, pre-dial timing and emitted SIP. Use the focused diagnostic when the value is empty, misplaced or absent.
Send one PJSIP NOTIFY without overstating the result
The PJSIP NOTIFY guide separates CLI, AMI and dialplan surfaces, endpoint, URI and in-dialog targets, message construction and the observed endpoint action. Use the focused diagnostic when the request is absent, rejected or accepted but ignored.
Control one outbound registration without masking its owner
The PJSIP registration-control guide separates AMI acceptance, unregister, register, retry scheduling and the later Registry event. Use the focused diagnostic when an action fails or has no visible effect.
Gate inheritable PJSIP headers by maintenance release and channel graph
The PJSIP_INHERITABLE_HEADER guide follows one controlled header from its calling channel to each outbound PJSIP child and keeps the separate PJSIP_HEADER() store visible. Use the focused diagnostic for missing, duplicated or wrong values.
Separate a sent PJSIP refresh from negotiated and observed media
The PJSIP_SEND_SESSION_REFRESH guide follows one immediate re-INVITE or UPDATE through offer-answer and resulting RTP. Use the focused diagnostic when the request is rejected or media changes incorrectly.
Inventory names on the current inbound INVITE
The PJSIP inbound-header inventory guide separates unique names, prefix filters, repeated instances and values. Use the paired diagnostic when the list is empty, incomplete or attached to the wrong message.
Choose the PJSIP hold-media owner per channel
The PJSIP music-on-hold passthrough guide separates endpoint default, current channel override, hold re-INVITE, local music and remote media. Use the paired diagnostic when signaling or audio diverges.
Return an explicit incoming PJSIP final response
The PJSIPHangup guide keeps application availability, incoming-channel ownership, answer timing, selected 4xx–6xx response, cause translation and caller result separate. Use the paired diagnostic when another final response wins.
Read PJSIP signaling context from the exact channel
The channel signaling guide separates Call-ID, dialog URIs and tags, target, original Request-URI, transport security and socket addresses. Use the paired diagnostic when a value is blank or surprising.
Read RTP context from the exact PJSIP media stream
The PJSIP channel RTP guide separates local, remote and direct-media addresses, per-stream encryption and hold restriction. Use the paired diagnostic when a value is blank or surprising.
Trace SIP MESSAGE from object to user-visible result
The PJSIP MessageSend guide separates the Asterisk message object, destination syntax, endpoint and contact routing, Request-URI, headers, protocol response and endpoint presentation. Use the paired diagnostic when sending fails or the wrong device is selected.
Filter one channel's dialplan output by an explicit group
The LOG_GROUP guide separates channel assignment from the active CLI filter, console verbosity, logger destinations and protocol traces. Use the paired diagnostic when expected lines disappear or unrelated output remains visible.
Observe channel frames before suppressing them
The FRAME_TRACE and FRAME_DROP guide separates channel ownership, RX and TX viewpoint, trace filtering, destructive suppression and external signaling or media evidence. Use the paired diagnostic when expected DTMF, voice or control frames are absent.
Serialize one shared-state invariant
The dialplan-lock guide separates named-mutex ownership, SHARED or global data, the protected operation and caller outcome. Use the paired diagnostic when calls pause or concurrent updates disagree.
Keep persistent local state in an owned AstDB namespace
The AstDB guide separates families, keys, blank values, absence, deletion, persistence and node scope. Use the paired diagnostic when a value is missing, stale or disappears.
Route by time without hiding the evaluated branch
The time-based dialplan guide separates timespec, timezone, channel clock, primitive, selected target and caller outcome. Use the paired diagnostic when the loaded branch differs from policy.
Map variable scope before using it as call state
The channel-variable scope guide separates current-channel ownership, one-level and recursive inheritance, Gosub-local frames, master-channel access and explicit cross-channel reads.
Separate dialplan exceptions from application status
The dialplan exception guide scopes the e handler, RaiseException() and EXCEPTION() evidence separately from the immediate TRYSTATUS returned by TryExec().
Separate dialplan lookup from subroutine execution
The dialplan lookup guide distinguishes exact-target existence, target-context value evaluation and a bounded Gosub execution contract.
Place Asterisk jitter buffers by channel and direction
The Asterisk jitter-buffer guide separates read-side channel placement, fixed or adaptive policy, conference behavior, transport evidence and added delay.
Capture PJSIP history in a bounded window
The PJSIP History guide separates in-memory retained messages from live logger output and interface packet capture, with explicit start, stop, clear, filtering and disposal boundaries.
Monitor Asterisk through a bounded metrics route
The Prometheus metrics guide separates module, HTTP listener, route, authentication, metric-provider and service-health evidence.
Use database lookups without hiding the call-path dependency
The func_odbc lookup guide separates the dialplan function, Asterisk connection, system DSN, database query, result shape and caller treatment.
Treat database mutation as a separately authorized call-path effect
The ODBC writes and transactions guide keeps write authorization, statement result, active transaction, completion and caller treatment independently observable.
Treat PJSIP Realtime as a complete mapping chain
The PJSIP Realtime guide follows one object through Sorcery type, source order, Realtime family, connector, schema, cache and runtime consumer.
Treat periodic audio as an owned asynchronous hook
The periodic-hook guide keeps the owner channel, returned hook ID, asynchronous hook channel, interval, injected audio and teardown as separate evidence.
Treat silence and noise waits as channel-owned media gates
The silence and noise wait guide keeps channel ownership, received frames, millisecond duration, iterations, second-based timeout, status and downstream call flow separate.
Treat a condition wait as a deferred expression over one named state owner
The WaitForCondition guide keeps expression deferral, mutable-state ownership, polling, timeout, result and later call flow separate.
Treat an epoch wait as channel work, not a detached scheduler
The WaitUntil guide keeps target conversion, channel lifetime, status, call timeout and later action independently observable.
Treat Dial as a multi-channel lifecycle
The Dial application guide follows requested resources, created outbound legs, answer, bridge, application status and continuation without collapsing them into one event.
Separate Originate acceptance from downstream call completion
The dialplan Originate guide follows one created channel through synchronous or asynchronous return, destination handoff and any later call stage.
Bridge two existing channels with explicit owners
The dialplan Bridge guide separates target lookup, answer state, bridge entry, application return, exit ownership and media evidence.
Build bounded holding workflows with BridgeWait and BridgeAdd
The BridgeWait and BridgeAdd guide separates holding handles, roles, entertainment, duration exits, live targets and actual bridge membership.
Control bridge membership through AMI without collapsing evidence layers
The AMI bridge-control guide separates action correlation, live channels, bridge snapshots, lifecycle events and media.
Control transfers through AMI without collapsing outcome layers
The AMI transfer-control guide separates Redirect, BlindTransfer and Atxfer contracts from transfer events, target contact and media.
Cancel an AMI attended transfer without guessing the active leg
The AMI cancel-and-recovery guide separates abort configuration, transferer identity, re-bridging and callback recovery.
Model Local optimization as a channel and bridge transition
The AMI Local optimization guide separates /n, explicit permission, correlated events, final topology and media.
Model AMI channel lists as bounded snapshots
The AMI channel snapshots guide separates inventory, connected maps, Status detail, stable identity and lifecycle evidence.
Trace AMI event delivery through every session layer
The AMI event-delivery guide separates read permissions, masks, filters, transport, queueing and client processing.
Define custom AMI telemetry as a versioned contract
The AMI UserEvent guide separates producer acknowledgement, custom fields, delivery and business outcome.
Synchronize remote AMI device state without replacing local providers
The AMI device-state guide gates the recent write action by maintenance release and keeps remote ownership, list boundaries, hints and endpoint evidence separate.
Separate AMI presence queries, lists and hint events
The AMI presence-state guide keeps the direct provider response, bounded list, every-provider event and hint-scoped event distinct.
Separate AMI extension queries, lists and hint events
The AMI extension-state guide keeps one Exten-and-Context query, a completed list, hint aggregation and downstream outcomes distinct.
Keep AMI dialplan mutations exact and reversible
The AMI runtime dialplan guide separates scoped inspection, exact mutation, durable source, reload survival and call outcome.
Keep AMI configuration edits inside the full lifecycle
The AMI configuration-file guide separates file inspection, ordered update, templates, includes, reload, loaded state and service outcome.
Separate module presence from capability readiness
The AMI module-control guide keeps loaded state, registration, configuration, runtime objects and behavior as separate checkpoints.
Inventory AMI commands from both user and administrator views
The AMI command-inventory guide distinguishes the actions exposed to one manager user from the actions registered on the running instance.
Separate AMI authentication from session authorization
The AMI session-authentication guide keeps listener reachability, Login, event selection, permissions and session lifetime as separate evidence.
Keep the built-in HTTP service beneath its consumers
The Asterisk HTTP and HTTPS server guide separates listener, TLS, URI registration, module, authentication and application results for ARI, AMI over HTTP, WebSockets and metrics.
Keep AMI core status and settings as separate evidence
The AMI core-evidence guide correlates CoreStatus and CoreSettings without treating either response as proof of capacity, channel state or call health.
Collect queue summary and detail as bounded AMI snapshots
The AMI queue-snapshot guide separates filters, response items, completion and runtime interpretation.
Query mailbox counts and waiting state without collapsing their meanings
The AMI mailbox-query guide preserves mailbox context, raw fields and voicemail-store ownership across MailboxCount and MailboxStatus.
Separate voicemail user inventory from message state
The AMI voicemail-user guide keeps VoicemailUsersList, IMAP-specific VoicemailUserStatus, counts and MWI delivery distinct.
Separate mailbox queries from MWI control
The AMI voicemail-refresh and MWI control guide keeps backend polling, external counts, remote state synchronization and endpoint indication separate.
Inventory ConfBridge rooms before diagnosing outcomes
The AMI ConfBridge snapshot guide separates active rooms, per-room participants, complete event framing and caller outcomes.
Separate ConfBridge control acceptance from resulting state
The AMI ConfBridge control guide bounds lock, mute, kick and recording actions by exact target, response, event and later state.
Interpret ConfBridge lifecycle events without inferring media
The AMI ConfBridge lifecycle guide separates room start and end, participant joins and leaves, detector state and caller outcomes.
Resolve ConfBridge profiles and menu outcomes explicitly
The ConfBridge profiles and menus guide separates explicit arguments, dynamic channel overrides, defaults, DTMF collection, prompts, actions and application results.
Control MixMonitor with channel and file evidence
The AMI MixMonitor guide separates manager actions, channel-bound audiohooks, events, file close, decoded audio and retention.
Operate parked calls with explicit AMI lifecycle evidence
The AMI parking guide separates the channel action, list framing, lot and space assignment, retrieval, timeout and caller outcome.
Control queue members without collapsing state and outcome
The AMI queue-member control guide scopes add, remove, pause and penalty changes to exact member identities and verifies later state separately.
Reload queue state with explicit subsection scope
The AMI queue reload and reset guide separates one-queue or all-queue scope, members, rules, parameters, statistics and retained history.
Control one waiting caller without collapsing later outcome
The AMI queue caller-control guide separates exact channel identity, priority, position, withdrawal, dialplan return and caller outcome.
Separate ring-in-use policy from queue eligibility
The queue-member ring-in-use guide keeps member identity, device state, policy, selection and call outcome independent.
Treat custom queue-log events as evidence, not outcomes
The AMI QueueLog guide separates action acceptance, row serialization, storage, ingestion and reporting.
Read release status and runtime capability as dated evidence
The refreshed release lifecycle guide separates displayed support labels from scheduled dates and documentation from the installed, loaded and registered capability.
Structure temporary dialplan state with HASH and ARRAY
The HASH and ARRAY guide separates keyed state, positional assignment, stable key snapshots, channel scope and lifecycle cleanup.
Manage dynamic dial targets without hiding call outcomes
The DIALGROUP guide separates named membership, PJSIP contact expansion, blank-list fallback and per-leg Dial evidence.
Bound remote dialplan lookups before they enter routing
The Asterisk CURL guide separates URL ownership, option scope, connection and total timeouts, HTTP policy, parsing and fallback.
Validate dialplan input before it becomes control data
The input-validation guide keeps null, length, filtering, full-shape matching, field count and extraction separate.
Inventory live channels without freezing their lifecycle
The channel identity guide separates a changing list, one exact channel identity and a later control action.
Migrate deprecated dialplan checks explicitly
The VALID_EXTEN migration guide converts each old lookup into an explicit context, extension and priority or label contract.
Authorize custom features per channel
The custom dynamic features guide separates applicationmap definition, DYNAMIC_FEATURES authorization, digit delivery and execution.
Read and write AMI variables with explicit scope
The AMI Getvar and Setvar guide separates channel, global, shared and function targets from response, event and outcome evidence.
Operate AstDB through AMI with exact namespace evidence
The AMI AstDB guide separates point and tree reads, exact and subtree deletion, readback and consumer evidence.
Inventory inbound PJSIP registrations without counting static contacts
The AMI inbound-registration inventory guide separates configured AoRs, dynamic contacts, all-contact lists and endpoint detail.
List PJSIP endpoints without collapsing their objects
The AMI endpoint inventory guide separates compact endpoint rows, exact object detail, aggregate state and service evidence.
Watch the explanation
Fred Posner explains how SIP routing, registration ownership and media services can be split across Kamailio, Asterisk and RTPengine. The discussion of backend probing and reconnect surges helps frame a larger deployment.
Read applicability and editorial notes →Watch this video here
AstriCon 2021 talk using Kamailio 5.5 examples. Treat it as an architecture lesson; our current written cluster uses 6.0 documentation. Review used the available transcript and primary references; audio and screen readability remain unverified.
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Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- Asterisk architecture ↗
- Asterisk release and support policy ↗
- Asterisk PJSIP authentication and digest-algorithm guide ↗
- Asterisk 22 res_pjsip configuration reference ↗
- Asterisk PJSIP configuration objects and relationships ↗
- RFC 8760: SIP Digest Access Authentication ↗
- RFC 7616: HTTP Digest Access Authentication ↗
- Asterisk Local Channel overview ↗
- Asterisk Local Channel modifiers ↗
- Asterisk Local Channel optimization ↗
- Asterisk hangup-cause overview and examples ↗
- Asterisk hangup-cause mapping table ↗
- Asterisk HANGUPCAUSE function ↗
- Asterisk HANGUPCAUSE_KEYS function ↗
- Asterisk HangupCauseClear application ↗
- Asterisk Dial application and DIALSTATUS ↗
- Asterisk CHANNEL function and hangupsource ↗
- Asterisk device-state concepts ↗
- Asterisk extension state and hints ↗
- Asterisk state query and manipulation ↗
- Asterisk res_pjsip presence subscriptions ↗
- Asterisk presence-state concepts ↗
- Asterisk states and presence overview ↗
- Asterisk taskprocessor CLI debugging reference ↗
- Asterisk taskprocessor implementation and CLI definitions ↗
- Asterisk call-pickup architecture ↗
- Asterisk certified 22.8 Pickup application ↗
- Asterisk PickupChan application ↗
- Asterisk 22 AMI Pickup event ↗
- Sangoma FreePBX feature-code descriptions ↗
- Sangoma FreePBX Ring Groups guide ↗
- Sangoma phone BLF key behavior ↗
- Asterisk feature-code call transfers ↗
- Asterisk 22 BlindTransfer application ↗
- Asterisk 22 AttendedTransfer application ↗
- Asterisk 22 features configuration ↗
- Asterisk 22 FEATUREMAP function ↗
- Asterisk AMI v2 transfer-event specification ↗
- Asterisk CEL specification ↗
- Sangoma FreePBX end-user transfer documentation ↗
- Sangoma FreePBX feature-code descriptions ↗
- Sangoma phone BLF transfer-type guide ↗
- Asterisk current res_pjsip configuration reference ↗
- Asterisk PJSIP troubleshooting guide ↗
- Asterisk dialing PJSIP channels ↗
- Asterisk PJSIP_AOR dialplan function ↗
- Asterisk 22 ContactStatus AMI event ↗
- Asterisk 22 PJSIPShowContacts AMI action ↗
- Sangoma FreePBX Extensions module guide ↗
- Sangoma FreePBX Core GraphQL APIs ↗
- Asterisk PJSIP transport-selection process ↗
- Asterisk 22 outbound-registration reference ↗
- Asterisk PJSIP proxy configuration ↗
- Asterisk outbound-registration architecture ↗
- Asterisk 22 OutboundRegistrationDetail event ↗
- Certified Asterisk 22.8 PJSIP wizard reference ↗
- IETF RFC 4028: Session Timers in SIP ↗
- IETF RFC 3261: SIP dialogs and in-dialog requests ↗
- Asterisk 22 PJSIP_SEND_SESSION_REFRESH function ↗
- Asterisk res_pjsip NAT configuration guide ↗
- RFC 3264: SDP offer/answer model ↗
- Asterisk bounded debug-information collection ↗
- RFC 4733: RTP telephone events ↗
- RFC 6086: SIP INFO package framework ↗
- Asterisk early-media and Progress guidance ↗
- Asterisk 22 Progress application ↗
- RFC 3262: reliable provisional responses and PRACK ↗
- RFC 6337: SIP offer/answer usage ↗
- Asterisk PJSIP advanced codec negotiation ↗
- Asterisk codec modules and translation paths ↗
- Asterisk audio and video capabilities ↗
- Asterisk chan_sip to res_pjsip migration guide ↗
- Asterisk STIR/SHAKEN deployment guide ↗
- Asterisk 22 res_stir_shaken reference ↗
- Asterisk 22 STIR_SHAKEN dialplan function ↗
- Asterisk 22 PJSIPHangup application ↗
- RFC 8224: authenticated identity management in SIP ↗
- RFC 8225: PASSporT ↗
- Asterisk DNS API overview ↗
- Asterisk current res_resolver_unbound reference ↗
- RFC 3263: locating SIP servers ↗
- RFC 2782: DNS SRV ↗
- Asterisk res_pjsip IPv6 configuration ↗
- RFC 3986: URI generic syntax ↗
- Asterisk current release news ↗
- Updating or upgrading Asterisk ↗
- Asterisk download and LTS selection guidance ↗
- Asterisk supported platforms ↗
- Asterisk release and branch policy ↗
- Asterisk 22 res_pjsip_acl configuration reference ↗
- Asterisk pjsip.conf sample ↗
- Asterisk PJSIP_HEADER dialplan function ↗
- Asterisk PJSIP_INHERITABLE_HEADER dialplan function ↗
- Asterisk PJSIP_HEADERS dialplan function ↗
- Asterisk PJSIP_HEADER_PARAM dialplan function ↗
- Asterisk PJSIP_RESPONSE_HEADER dialplan function ↗
- Asterisk pre-dial handlers ↗
- Asterisk WebSocket channel driver ↗
- Asterisk 22 chan_websocket module configuration ↗
- Asterisk WebSocket client configuration ↗
- Asterisk ARI outbound WebSockets ↗
- Asterisk 22 Dial application WebSocket parameters ↗
- Official Asterisk ARI and media WebSocket examples ↗
- Official Asterisk release archive ↗
- Asterisk call-file documentation ↗
- Official Asterisk sample.call ↗
- Asterisk outgoing spool source ↗
- Asterisk module-loading troubleshooting ↗
- Asterisk main configuration reference ↗
- Asterisk call-file and Local-channel example ↗
- Asterisk 23 AMI Originate action ↗
- Asterisk 23 OriginateResponse event ↗
- Asterisk Manager TCP/IP API ↗
- Asterisk network security considerations ↗
- Asterisk REST Interface recommended practices ↗
- Asterisk 23 ARI Channels REST API ↗
- Asterisk configuration for ARI ↗
- Asterisk introduction to ARI and channels ↗
- Asterisk ARI channel-state operations ↗
- Asterisk ARI bridge operations ↗
- Asterisk ARI basic mixing bridges ↗
- Asterisk External Media and ARI ↗
- Asterisk AudioSocket protocol ↗
- Asterisk ARI Bridges REST API ↗
- Asterisk bridges in a Stasis application ↗
- Asterisk ARI holding bridges ↗
- Asterisk ARI data models ↗
- Asterisk ARI recording API ↗
- Asterisk ARI simple media playback ↗
- Asterisk ARI playback operations ↗
- Asterisk ARI Playbacks REST API ↗
- Asterisk ARI Sounds REST API ↗
- Asterisk ARI DTMF handling ↗
- Asterisk AGI overview ↗
- Asterisk 23 AGI application reference ↗
- Asterisk 23 EAGI application reference ↗
- Asterisk 23 AGI command index ↗
- Asterisk 23 AGI VERBOSE command ↗
- Asterisk AsyncAGI AMI action ↗
- Asterisk 23 AsyncAGIStart event ↗
- Asterisk 23 AsyncAGIExec event ↗
- Asterisk 23 ASYNCAGI BREAK command ↗
- Asterisk deprecated DeadAGI application reference ↗
- Asterisk party-ID manipulation ↗
- Asterisk party-ID interception routines ↗
- Asterisk 22 CONNECTEDLINE function ↗
- Asterisk 22 REDIRECTING function ↗
- Asterisk PJSIP_ENDPOINT function ↗
- Asterisk Queue application ↗
- Asterisk NewConnectedLine AMI event ↗
- Asterisk Channel Event Logging overview ↗
- Asterisk CEL configuration files ↗
- Asterisk 22 CEL configuration reference ↗
- Asterisk queue logs ↗
- Asterisk logging configuration ↗
- Asterisk 22 Queue application ↗
- Asterisk 22 QueueLog application ↗
- Certified Asterisk 22.8 QueueLog AMI action ↗
- Asterisk QUEUE_VARIABLES function ↗
- Asterisk dialplan subroutines overview ↗
- Asterisk Gosub guide ↗
- Asterisk 22 Gosub application reference ↗
- Asterisk 22 Return application reference ↗
- Asterisk STACK_PEEK function reference ↗
- Asterisk pre-bridge handlers ↗
- Asterisk hangup handlers ↗
- Asterisk 22 CHANNEL function reference ↗
- Asterisk 22 HangupHandlerPush AMI event ↗
- Asterisk 22 HangupHandlerPop AMI event ↗
- Asterisk 22 HangupHandlerRun AMI event ↗
- Asterisk 22 Hangup AMI event ↗
- Asterisk 22 Bridge application reference ↗
- Asterisk 22 Goto application reference ↗
- Asterisk 22 BridgeEnter AMI event ↗
- Asterisk 22 BridgeLeave AMI event ↗
- Asterisk 22 TALK_DETECT function ↗
- Asterisk 22 DENOISE function ↗
- Asterisk 22 AMI ChannelTalkingStart event ↗
- Asterisk 22 AMI ChannelTalkingStop event ↗
- Asterisk 22 ARI data models ↗
- Asterisk GROUP function ↗
- Asterisk GROUP_COUNT function ↗
- Asterisk GROUP_LIST function ↗
- Asterisk GROUP_MATCH_COUNT function ↗
- Asterisk channels key concepts ↗
- Asterisk channel-variable inheritance ↗
- Asterisk 24 BackGround application ↗
- Asterisk latest WaitExten application ↗
- Asterisk latest TIMEOUT function ↗
- Asterisk dialplan pattern matching ↗
- Asterisk contexts, extensions and priorities ↗
- Asterisk PJSIP_DIAL_CONTACTS function ↗
- Asterisk PJSIP_CONTACT function ↗
- Asterisk AMI PJSIPShowAors action ↗
- Asterisk AMI PJSIPShowEndpoint action ↗
- Asterisk PJSIP_MEDIA_OFFER function ↗
- Asterisk PJSIP_SEND_SESSION_REFRESH function ↗
- Asterisk PJSIP_PARSE_URI function ↗
- Asterisk PJSIP_PARSE_URI_FROM function ↗
- Asterisk FILTER function ↗
- Asterisk important security considerations ↗
- RFC 3261: SIP Session Initiation Protocol ↗
- Asterisk HOLD_INTERCEPT function ↗
- Asterisk AMI Hold event ↗
- Asterisk AMI Unhold event ↗
- Asterisk PJSIP_MOH_PASSTHROUGH function ↗
- Asterisk 22 FAXOPT function ↗
- Asterisk 22 ReceiveFAX application ↗
- Asterisk 22 SendFAX application ↗
- Asterisk branch 22 res_fax sample configuration ↗
- Asterisk branch 22 res_fax implementation ↗
- Asterisk Media Experience Score ↗
- Asterisk MES implementation ↗
- Asterisk bridge media paths ↗
- Asterisk PJSIP_RESPONSE_HEADERS dialplan function ↗
- Asterisk res_pjsip header-function source ↗
- Asterisk PJSIP_DTMF_MODE dialplan function ↗
- Asterisk Dial application and pre-dial handler guidance ↗
- Asterisk AMI DTMFBegin event ↗
- Asterisk AMI DTMFEnd event ↗
- Asterisk PJSIP_TRANSFER_HANDLING dialplan function ↗
- Asterisk ARI transfer handling guide ↗
- Asterisk res_pjsip remote attended transfers ↗
- RFC 3261: SIP ↗
- Asterisk PJSIPNotify dialplan application ↗
- Asterisk PJSIPNotify AMI action ↗
- Asterisk res_pjsip_notify configuration reference ↗
- Asterisk PJSIPNotify application introduction ↗
- RFC 6665: SIP-Specific Event Notification ↗
- Asterisk PJSIPRegister AMI action ↗
- Asterisk PJSIPUnregister AMI action ↗
- Asterisk PJSIPShowRegistrationsOutbound AMI action ↗
- Asterisk OutboundRegistrationDetail AMI event ↗
- Asterisk Registry AMI event ↗
- Asterisk res_pjsip_outbound_registration configuration reference ↗
- Asterisk 20 PJSIP_INHERITABLE_HEADER reference ↗
- Asterisk 22 PJSIP_INHERITABLE_HEADER reference ↗
- Asterisk 23 PJSIP_INHERITABLE_HEADER reference ↗
- RFC 3311: SIP UPDATE method ↗
- RFC 3264: SDP offer/answer model ↗
- RFC 6337: SIP usage of the offer-answer model ↗
- Asterisk PJSIPHangup dialplan application ↗
- Asterisk Hangup dialplan application ↗
- RFC 3711: Secure RTP ↗
- Asterisk MessageSend dialplan application ↗
- Asterisk MESSAGE dialplan function ↗
- Asterisk MESSAGE_DATA dialplan function ↗
- Asterisk 22 PJSIP_ENDPOINT reference ↗
- RFC 3428: SIP MESSAGE method ↗
- Asterisk LOG_GROUP dialplan function ↗
- Asterisk 22 LOG_GROUP dialplan function ↗
- Asterisk logging overview ↗
- Asterisk basic logging commands ↗
- Asterisk call identifier logging ↗
- Asterisk FRAME_TRACE dialplan function ↗
- Asterisk FRAME_DROP dialplan function ↗
- Asterisk LOCK dialplan function ↗
- Asterisk TRYLOCK dialplan function ↗
- Asterisk UNLOCK dialplan function ↗
- Asterisk 22 SHARED dialplan function ↗
- Asterisk global variables basics ↗
- Asterisk internal database overview ↗
- Asterisk DB dialplan function ↗
- Asterisk DB_EXISTS dialplan function ↗
- Asterisk DB_DELETE dialplan function ↗
- Asterisk DB_KEYS dialplan function ↗
- Asterisk 22 GotoIfTime application ↗
- Asterisk 22 ExecIfTime application ↗
- Asterisk 22 IFTIME function ↗
- Asterisk 22 TESTTIME function ↗
- Asterisk 22 STRFTIME function ↗
- Sangoma: Time Conditions module overview ↗
- Sangoma: Time Conditions user guide ↗
- Sangoma: Time Group user guide ↗
- FreePBX Time Conditions release/17.0 metadata ↗
- Asterisk 22 Set application ↗
- Asterisk 22 MASTER_CHANNEL function ↗
- Asterisk 22 LOCAL function ↗
- Asterisk 22 LOCAL_PEEK function ↗
- Asterisk 22 VARIABLE_EXISTS function ↗
- Asterisk 22 IMPORT function ↗
- Asterisk special dialplan extensions ↗
- Asterisk 23 RaiseException application ↗
- Asterisk 23 EXCEPTION function ↗
- Asterisk 23 TryExec application ↗
- Asterisk 23 DIALPLAN_EXISTS function ↗
- Asterisk 23 EVAL_EXTEN function ↗
- Asterisk 23 EVAL_SUB function ↗
- Asterisk 23 Return application ↗
- Asterisk 23 JITTERBUFFER dialplan function ↗
- Asterisk ConfBridge configuration ↗
- Asterisk IP Quality of Service ↗
- Asterisk packet-loss concealment overview ↗
- Asterisk PJSIP History how-to ↗
- Asterisk res_pjsip_history source and CLI definitions ↗
- Asterisk 23 res_prometheus configuration ↗
- Asterisk prometheus.conf sample ↗
- Asterisk res_prometheus implementation ↗
- Asterisk built-in HTTP server setup ↗
- Asterisk project release history ↗
- Prometheus overview and pull model ↗
- Asterisk Configuring res_odbc ↗
- Asterisk func_odbc.conf sample ↗
- Asterisk res_odbc.conf sample ↗
- Asterisk 23 ODBC_FETCH function ↗
- Asterisk 23 SQL_ESC function ↗
- Asterisk 23 SQL_ESC_BACKSLASHES function ↗
- Asterisk security advisory AST-2022-003 ↗
- Asterisk database transactions ↗
- Asterisk latest ODBC transaction function ↗
- Asterisk latest ODBC_Commit application ↗
- Asterisk latest ODBC_Rollback application ↗
- Asterisk Realtime database configuration ↗
- Asterisk setting up PJSIP Realtime ↗
- Asterisk Sorcery overview and mapping ↗
- Asterisk Sorcery caching ↗
- Asterisk Realtime schema management with Alembic ↗
- Asterisk extconfig.conf sample ↗
- Asterisk sorcery.conf sample ↗
- Asterisk 23 PERIODIC_HOOK function ↗
- Asterisk 22 PERIODIC_HOOK function ↗
- Asterisk 20 PERIODIC_HOOK function ↗
- Asterisk Playback application ↗
- Asterisk 23 WaitForSilence application ↗
- Asterisk 23 WaitForNoise application ↗
- Asterisk 22 WaitForSilence application ↗
- Asterisk 22 WaitForNoise application ↗
- Asterisk 23 BackgroundDetect application ↗
- Asterisk 23 WaitForCondition application ↗
- Asterisk 22 WaitForCondition application ↗
- Asterisk dialplan expressions ↗
- Asterisk conditional applications ↗
- Asterisk 23 VARIABLE_EXISTS function ↗
- Asterisk 23 GLOBAL_EXISTS function ↗
- Asterisk 23 WaitUntil application ↗
- Asterisk 22 WaitUntil application ↗
- Asterisk 23 STRFTIME function ↗
- Asterisk 23 STRPTIME function ↗
- Asterisk 23 Originate application ↗
- Asterisk 22 Originate application ↗
- Asterisk 23 Bridge application ↗
- Asterisk Bridge application overview ↗
- Asterisk 23 CHANNELS function ↗
- Asterisk 23 BridgeWait application ↗
- Asterisk 22 BridgeWait application ↗
- Asterisk 23 BridgeAdd application ↗
- Asterisk 22 BridgeAdd application ↗
- Asterisk 23 AMI Bridge action ↗
- Asterisk 22 AMI Bridge action ↗
- Asterisk 23 AMI BridgeList action ↗
- Asterisk 23 AMI BridgeInfo action ↗
- Asterisk 23 AMI BridgeKick action ↗
- Asterisk 22 AMI BridgeKick action ↗
- Asterisk 23 AMI BridgeCreate event ↗
- Asterisk 23 AMI BridgeEnter event ↗
- Asterisk 23 AMI BridgeLeave event ↗
- Asterisk 23 AMI BridgeInfoChannel event ↗
- Asterisk 23 AMI BridgeInfoComplete event ↗
- Asterisk 23 AMI Redirect action ↗
- Asterisk 22 AMI Redirect action ↗
- Asterisk 23 AMI BlindTransfer action ↗
- Asterisk 22 AMI BlindTransfer action ↗
- Asterisk 23 AMI Atxfer action ↗
- Asterisk 22 AMI Atxfer action ↗
- Asterisk 23 AMI BlindTransfer event ↗
- Asterisk 23 AMI AttendedTransfer event ↗
- Asterisk 23 AMI CancelAtxfer action ↗
- Asterisk 22 AMI CancelAtxfer action ↗
- Asterisk 23 attended-transfer feature configuration ↗
- Asterisk 23 AMI LocalOptimizeAway action ↗
- Asterisk 22 AMI LocalOptimizeAway action ↗
- Asterisk 23 AMI LocalOptimizationBegin event ↗
- Asterisk 23 AMI LocalOptimizationEnd event ↗
- Asterisk 23 AMI LocalBridge event ↗
- Asterisk 23 AMI CoreShowChannelMap action ↗
- Asterisk 22 AMI CoreShowChannelMap action ↗
- Asterisk 23 AMI CoreShowChannelMapComplete event ↗
- Asterisk 22 AMI CoreShowChannelMapComplete event ↗
- Asterisk 23 AMI CoreShowChannels action ↗
- Asterisk 23 AMI CoreShowChannel event ↗
- Asterisk 23 AMI CoreShowChannelsComplete event ↗
- Asterisk 23 AMI Status action ↗
- Asterisk 23 AMI Status event ↗
- Asterisk 23 AMI StatusComplete event ↗
- Asterisk 23 AMI Newchannel event ↗
- Asterisk 23 AMI Rename event ↗
- Asterisk 23 AMI Events action ↗
- Asterisk 22 AMI Events action ↗
- Asterisk 23 AMI Filter action ↗
- Asterisk 22 AMI Filter action ↗
- Asterisk 23 AMI WaitEvent action ↗
- Asterisk 22 AMI WaitEvent action ↗
- Asterisk AMI event filtering ↗
- Asterisk 23 AMI Login action ↗
- Asterisk 23 AMI Ping action ↗
- Asterisk 23 AMI UserEvent action ↗
- Asterisk 22 AMI UserEvent action ↗
- Asterisk 23 AMI UserEvent event ↗
- Asterisk 22 AMI UserEvent event ↗
- Asterisk 23 UserEvent dialplan application ↗
- Asterisk 22 UserEvent dialplan application ↗
- Asterisk AMI command syntax ↗
- Asterisk 23 AMI DeviceStateChange action ↗
- Asterisk 22 AMI DeviceStateChange action ↗
- Asterisk 23 AMI DeviceStateList action ↗
- Asterisk 22 AMI DeviceStateList action ↗
- Asterisk 23 AMI DeviceStateChange event ↗
- Asterisk 22 AMI DeviceStateChange event ↗
- Asterisk 23 AMI DeviceStateListComplete event ↗
- Asterisk 22 AMI DeviceStateListComplete event ↗
- Asterisk 23 DEVICE_STATE function ↗
- Asterisk 22 DEVICE_STATE function ↗
- Asterisk 23 AMI PresenceState action ↗
- Asterisk 22 AMI PresenceState action ↗
- Asterisk 23 AMI PresenceStateList action ↗
- Asterisk 22 AMI PresenceStateList action ↗
- Asterisk 23 AMI PresenceStateChange event ↗
- Asterisk 22 AMI PresenceStateChange event ↗
- Asterisk 23 AMI PresenceStatus event ↗
- Asterisk 22 AMI PresenceStatus event ↗
- Asterisk 23 AMI PresenceStateListComplete event ↗
- Asterisk 22 AMI PresenceStateListComplete event ↗
- Asterisk 23 PRESENCE_STATE function ↗
- Asterisk 22 PRESENCE_STATE function ↗
- Asterisk 23 AMI ExtensionState action ↗
- Asterisk 22 AMI ExtensionState action ↗
- Asterisk 23 AMI ExtensionStateList action ↗
- Asterisk 22 AMI ExtensionStateList action ↗
- Asterisk 23 AMI ExtensionStatus event ↗
- Asterisk 22 AMI ExtensionStatus event ↗
- Asterisk 23 AMI ExtensionStateListComplete event ↗
- Asterisk 22 AMI ExtensionStateListComplete event ↗
- Asterisk 23 EXTENSION_STATE function ↗
- Asterisk 22 EXTENSION_STATE function ↗
- Asterisk 23 HINT function ↗
- Asterisk 22 HINT function ↗
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- Asterisk manager.conf sample ↗
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- Asterisk 23 Documentation ↗
- Asterisk: CLI Syntax and Help Commands ↗
- Asterisk HASH dialplan function ↗
- Asterisk HASHKEYS dialplan function ↗
- Asterisk ARRAY dialplan function ↗
- Asterisk ClearHash dialplan application ↗
- Asterisk SHIFT dialplan function ↗
- Asterisk channel-variable setting and substitution ↗
- Asterisk func_strings implementation ↗
- Asterisk DIALGROUP dialplan function ↗
- Asterisk command-line interface ↗
- Asterisk func_dialgroup implementation ↗
- Asterisk CURL dialplan function ↗
- Asterisk CURLOPT dialplan function ↗
- Asterisk JSON_DECODE dialplan function ↗
- Asterisk URIENCODE dialplan function ↗
- Asterisk func_curl implementation ↗
- libcurl CURLOPT_TIMEOUT reference ↗
- libcurl CURLOPT_CONNECTTIMEOUT reference ↗
- Asterisk REGEX dialplan function ↗
- Asterisk ISNULL dialplan function ↗
- Asterisk LEN dialplan function ↗
- Asterisk FIELDQTY dialplan function ↗
- Asterisk CUT dialplan function ↗
- Asterisk dialplan-functions overview ↗
- Asterisk CHANNEL_EXISTS dialplan function ↗
- Asterisk CLI channel creation and manipulation ↗
- Asterisk func_channel implementation ↗
- Asterisk 22 VALID_EXTEN function and deprecation notice ↗
- Asterisk 23 dialplan-function index ↗
- Asterisk func_dialplan implementation ↗
- Asterisk dialplan reload and show workflow ↗
- Asterisk custom dynamic features ↗
- Asterisk built-in dynamic features ↗
- Asterisk features overview ↗
- Asterisk upstream features.conf sample ↗
- Asterisk upstream features configuration implementation ↗
- Asterisk 23 AMI Getvar action ↗
- Asterisk 23 AMI Setvar action ↗
- Asterisk 23 AMI VarSet event ↗
- Asterisk 23 SHARED dialplan function ↗
- Asterisk external-protocol dialplan-function safeguards ↗
- Asterisk 23 AMI DBGet action ↗
- Asterisk 23 AMI DBGetTree action ↗
- Asterisk 23 AMI DBPut action ↗
- Asterisk 23 AMI DBDel action ↗
- Asterisk 23 AMI DBDelTree action ↗
- Asterisk SQLite3 AstDB back end ↗
- Asterisk 23 PJSIPShowRegistrationsInbound action ↗
- Asterisk 22 PJSIPShowRegistrationsInbound action ↗
- Asterisk 23 PJSIPShowRegistrationInboundContactStatuses action ↗
- Asterisk 22 PJSIPShowRegistrationInboundContactStatuses action ↗
- Asterisk 23 PJSIPShowContacts action ↗
- Asterisk 23 InboundRegistrationDetail event ↗
- Asterisk 23 ContactStatusDetail event ↗
- Asterisk 23 PJSIPShowEndpoints action ↗
- Asterisk 23 EndpointList event ↗
- Asterisk 23 EndpointListComplete event ↗
- Asterisk 23 EndpointDetail event ↗
- Asterisk 23 EndpointDetailComplete event ↗