VoIP.infoThe independent guide to business communications.
VoIP.info Troubleshooting

Asterisk AMI PlayDTMF, PlayMF or SendFlash hits the wrong leg or has no effect

Find the first divergence across action availability, channel identity, direction, signalling, Asterisk events and downstream behavior before retrying control input.

Reviewed 2026-10-25Current branch-23-generated and Asterisk 22 PlayDTMF, PlayMF and SendFlash action references, the current DTMFBegin event reference and res_pjsip configuration reference were reviewed October 25, 2026; installed release, loaded app_senddtmf or app_mf module, manager permission, channel technology, live channel identity, bridge topology, direction, negotiated media, DTMF mode and downstream application behavior remain deployment specific; no PBX, channel, manager session, action, digit, hook flash, signalling frame, packet, media stream, endpoint, IVR, carrier or live call was inspected or testedAdvancedDocumentation based · not lab tested

Freeze one failed action and its call topology

Record the intended and observed node, release, loaded module, manager identity, unique ActionID, action, Channel, Uniqueid, Linkedid, bridge membership, direction, digit, duration, receive flag, raw response and synchronized timestamps. Redact credentials and sensitive dialed input.

Classify the first symptom as unknown or denied action, vanished or wrong channel, wrong direction, unsupported digit, signalling mismatch, missing core event, missing downstream input, duplicate input or hook flash with no defined consumer.

Confirm action and module availability before media diagnosis

Check the running AMI action inventory, installed branch and maintenance release, loaded app_senddtmf or app_mf module and manager permissions. SendFlash is version-gated in the generated record from Asterisk 20.3.0 and 18.18.0.

An unknown or denied action is not evidence of a codec, RTP, SIP INFO, endpoint or IVR fault. Do not widen the production manager account merely to make the workflow pass.

Reconcile the live channel leg and direction

Resolve the exact live channel immediately before the action and confirm Uniqueid, Linkedid, bridge peers and call direction. For PlayDTMF, verify whether the intent was to send out on the channel or use Receive: true to emulate input into Asterisk.

If a channel was replaced, masqueraded, transferred, locally optimized or hung up, stop. Do not reuse a stale name or fall back to a regex that can select another call.

Trace DTMF at each signalling boundary

Record PJSIP dtmf_mode, negotiated RFC 4733 event mapping where applicable, SIP INFO messages when used, in-band audio only when that mode is intended and Asterisk DTMFBegin or end observations. Compare the requested duration and digit with what each boundary exposed.

A successful PlayDTMF response is not proof that a peer received a telephone event or that an IVR consumed it. Likewise, an Asterisk event does not prove downstream application handling.

Diagnose MF and hook flash on their own semantics

For PlayMF, verify that the remote system actually expects MF signalling and that the requested symbol and timing are valid; do not translate a DTMF failure into an MF test. For SendFlash, confirm the channel technology and downstream gateway or service defines a hook-flash behavior.

Neither operation is a generic SIP feature. Separate action acceptance, locally generated signal, transport, remote recognition and application effect.

Retest one harmless input with a control

After correcting one bounded cause, use a controlled call, exact channel identity, one harmless digit or authorized flash, one unique ActionID and synchronized evidence. Compare a known-good channel or application path without changing several variables at once.

Stop after the first confirmed divergence and avoid automatic retries. No PBX, channel, manager action, packet, audio stream, digit, MF signal, hook flash, endpoint, IVR, carrier 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-23-generated and Asterisk 22 PlayDTMF, PlayMF and SendFlash action references, the current DTMFBegin event reference and res_pjsip configuration reference were reviewed October 25, 2026; installed release, loaded app_senddtmf or app_mf module, manager permission, channel technology, live channel identity, bridge topology, direction, negotiated media, DTMF mode and downstream application behavior remain deployment specific; no PBX, channel, manager session, action, digit, hook flash, signalling frame, packet, media stream, endpoint, IVR, carrier or live call was inspected or tested

Asterisk AMI DTMF, MF and hook-flash control: channel and direction evidence

Use PlayDTMF, PlayMF and SendFlash without confusing action acceptance, receive emulation, signalling family, channel leg or downstream effect.

Read guide →
Guide · Current Asterisk interface documentation reviewed September 30, 2026; implementation must be checked against the installed branch and integration library

Asterisk AMI, ARI and AGI: choose the control boundary

Separate management events, synchronous dialplan work and application-owned call control before exposing an integration.

Read guide →
Guide · 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 command inventory and permission evidence

Use ListCommands and the administrator CLI to distinguish the actions visible to one manager user from the actions registered by the running Asterisk instance.

Read guide →
Guide · Current Asterisk Latest API CHANNEL_EXISTS, CHANNELS and CHANNEL references, CLI channel-operation and Local-channel optimization documentation, AMI CoreShowChannels surfaces and the upstream func_channel implementation were reviewed October 19, 2026. The generated API pages reported branch 23; installed branch, loaded modules, channel drivers, names, unique IDs, linked IDs, Local optimization, permissions, timing and call outcomes remain deployment specific. No PBX, module, dialplan, channel, bridge, manager session, packet, media stream or live call was inspected or tested

Asterisk channel inventory, identity and lifetime

Treat a channel list as a changing observation, distinguish name from unique identity, and revalidate the exact leg before an authorized operation.

Read guide →
Guide · Asterisk 22 res_pjsip, logging and troubleshooting references plus RFC 4733 and RFC 6086 reviewed October 7, 2026; no Asterisk host, endpoint, trunk, IVR, bridge, RTP event, SIP INFO request, audio tone, digit, packet capture, media stream or live call was inspected or tested

Asterisk PJSIP DTMF: mode, negotiation and call-leg evidence

Keep RFC 4733 events, in-band tones and SIP INFO separate on each PJSIP leg, then prove what Asterisk received and sent.

Read guide →
Guide · Current first-party Asterisk 22 ARI DTMF event, Channels REST API, bridge-mixing, ARI configuration and PJSIP DTMF-mode documentation reviewed October 9, 2026; no HTTP server, ARI identity, WebSocket, Stasis application, channel, bridge, endpoint, SIP message, RTP event, audio tone, digit, playback, application state or call was inspected or tested

Asterisk ARI DTMF: events, bridges and digit injection

Handle received digit events and outbound digit injection as separate operations, preserving channel identity, transport, bridge decoding, duration and application state.

Read guide →
Guide · Current Asterisk PJSIP_DTMF_MODE, PJSIP_ENDPOINT, CHANNEL, Dial, res_pjsip and AMI DTMF event documentation reviewed October 13, 2026; PJSIP_DTMF_MODE is documented since Asterisk 13.18.0, 14.7.0 and 15.1.0, while installed branch, maintenance release, loaded channel driver, endpoint configuration, channel ownership, negotiated SDP, received and transmitted digits, downstream IVR behavior and media remain deployment specific; no PBX, endpoint, channel, dialplan, SIP message, RTP event, audio stream, digit or call was inspected or tested

Asterisk PJSIP_DTMF_MODE: channel override, signaling method and proof

Read or change the DTMF mode on one PJSIP call leg without confusing endpoint policy, channel ownership, advertised telephone events or a downstream IVR result.

Read guide →
Protocol reference

DTMF over VoIP: why menus sometimes miss digits

Separate in-band tones, RTP telephone events and signaling-based methods across each call leg.

Read guide →
Guide · Operational guidance; Asterisk 22 / FreePBX 17 where applicable

Least privilege for PBX users and integrations

Separate the right to register, place calls, administer the PBX and read sensitive data.

Read guide →
Troubleshooting · Current Asterisk 22 res_pjsip DTMF-mode, logging, Background, WaitExten and special-extension documentation reviewed September 28, 2026; no endpoint, trunk, codec, RTP event, SIP INFO request, audio tone, IVR, digit or live call was tested

DTMF digits are ignored by an IVR

Locate a mismatch between audio tones, RTP events and the receiving call leg.

Read guide →
Troubleshooting · Asterisk 22 res_pjsip, logging and troubleshooting references plus RFC 4733 and RFC 6086 reviewed October 7, 2026; no Asterisk host, endpoint, trunk, IVR, bridge, RTP event, SIP INFO request, audio tone, digit, packet capture, media stream or live call was inspected or tested

Asterisk receives DTMF but the far IVR ignores or doubles it

Correlate one digit across ingress receipt, bridge ownership, egress representation and the far application before changing endpoint modes.

Read guide →
Troubleshooting · Current first-party Asterisk 22 ARI DTMF event, Channels REST API, bridge-mixing, ARI configuration and PJSIP DTMF-mode documentation reviewed October 9, 2026; no HTTP server, ARI identity, WebSocket, Stasis application, channel, bridge, endpoint, SIP message, RTP event, audio tone, digit, playback, application state or call was inspected or tested

Asterisk ARI DTMF events are missing, duplicated or late

Trace one expected digit sequence from its source transport and channel through Stasis, bridge decoding, WebSocket delivery and the application's current input state.

Read guide →
Troubleshooting · Asterisk Latest API branch 23 FRAME_TRACE, FRAME_DROP and CHANNEL documentation plus current Asterisk logging and debug-collection guidance reviewed October 14, 2026; FRAME_TRACE is documented since Asterisk 10.0.0 and FRAME_DROP since 16.21.0 and 18.7.0, while installed-version availability, loaded functions, channel ownership, frame direction, frame type, media topology, logger state and observed output remain deployment specific; no PBX, module, dialplan, channel, frame, DTMF event, control event, media stream, log, packet or call was inspected or tested

Asterisk frame trace shows no DTMF, voice or expected control event

Locate the first mismatch among function availability, channel leg, trace filter, RX or TX viewpoint, logger output and the external signaling or media path before dropping frames.

Read guide →
Hub

VoIP troubleshooting

Use symptoms and evidence to isolate registration, routing, media and device faults.

Explore topic →