Describe the digit failure
Record one affected call's timestamp and timezone, source device, ingress trunk or endpoint, destination menu, prompt state and exact digit sequence. Classify whether the menu misses all digits, misses selected digits, doubles them, accepts them only slowly, fails after transfer or recognizes them but routes elsewhere. A local keypad sound, a SIP registration or audible two-way speech does not prove the PBX received one usable digit event.
Inspect the negotiated method
Asterisk's current PJSIP endpoint options distinguish RFC 4733 RTP events, in-band audio, SIP INFO, automatic selection and no DTMF. Preserve the effective endpoint and trunk settings plus the offer and answer on the exact leg. For RTP events, inspect the negotiated telephone-event mapping and event range; do not copy a dynamic payload number from another call. For SIP INFO, preserve the request, content type and response.
Asterisk can terminate and regenerate DTMF between different call legs. Internal success therefore does not prove carrier ingress, a transferred leg or a remote service uses the same method. Direct media can also bypass Asterisk and prevents features that require it to intercept digits; do not change media topology until the actual path is known.
Reproduce a bounded test
Use bounded Asterisk logging that includes the DTMF level and PJSIP messages, then reproduce only on an authorized test call. Establish whether Asterisk logged the digit before interpreting IVR behavior. RTP packet debug can expose sensitive call information and high volume; scope it narrowly and remove it after capture.
Enter a single digit, repeated digits and a short multi-digit choice at a controlled pace. Compare one known-good endpoint or route. If failure begins after answer, early media, hold, transfer or re-INVITE, preserve the later SDP and channel path rather than generalizing the initial negotiation.
Make the compatible change
When Asterisk logs the correct digit but the menu ignores or misroutes it, move to the application layer: current dialplan context, Background or WaitExten matching, digit and response timeouts, valid extensions and invalid or timeout branches. Use the focused FreePBX IVR keypress diagnostic for managed menus.
Correct only the demonstrated owner: endpoint or trunk DTMF mode, offer/answer mismatch, direct-media path, application timing or menu mapping. Avoid enabling several methods simultaneously because duplicate representations can produce doubled input. Do not change codecs merely because in-band tones exist; first prove the deployment intentionally depends on in-band detection. No endpoint, trunk, codec, RTP event, SIP INFO request, audio tone, IVR, digit or live call was tested for this guide.
When Asterisk logs the digit but the far IVR still fails
Follow the Asterisk PJSIP mismatch diagnostic across ingress receipt, bridge and media ownership, egress method and far application timing. Asterisk can receive one correct event while a different representation, duplicate or no event leaves the other leg.
Check the exact Asterisk PJSIP leg and override
When dialplan changes a live PJSIP mode, use the focused override diagnostic before changing codecs, endpoint templates or every IVR route.
Trace the exact channel that should own the digit
Use the Asterisk frame-trace diagnostic when the protocol evidence and application result disagree about whether a DTMF event reached one channel.
Continue from DTMF receipt to ConfBridge sequence matching
When Asterisk receives digits but a conference menu still fails, use the ConfBridge menu diagnostic to inspect profile resolution, sequence collection, role and action evidence.
Trace MicroSIP digits beyond local audible feedback
A locally heard or suppressed tone does not establish what MicroSIP transmitted. Use the MicroSIP call diagnostic to compare method selection, SDP event mapping, RTP or INFO evidence, bridge conversion and remote application result.
Check whether the channel may activate the custom feature
Use the Asterisk dynamic-feature diagnostic when digits arrive but a bridge feature remains silent.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- Asterisk res_pjsip DTMF mode reference ↗
- Asterisk PJSIP troubleshooting guide ↗
- Asterisk debug-information collection ↗
- Asterisk Background and WaitExten behavior ↗
- Asterisk special invalid and timeout extensions ↗
- Asterisk 22 res_pjsip DTMF mode reference ↗
- RFC 4733: RTP telephone events ↗
- RFC 6086: SIP INFO package framework ↗
- Asterisk PJSIP_DTMF_MODE dialplan function ↗
- Asterisk PJSIP_ENDPOINT dialplan function ↗
- Asterisk CHANNEL dialplan function ↗
- Asterisk Dial application and pre-dial handler guidance ↗
- Asterisk AMI DTMFBegin event ↗
- Asterisk AMI DTMFEnd event ↗
- Asterisk FRAME_TRACE dialplan function ↗
- Asterisk FRAME_DROP dialplan function ↗
- Asterisk logging overview ↗
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- MicroSIP project and platform scope ↗
- MicroSIP stable downloads and changelog ↗
- MicroSIP current-version pointer ↗
- MicroSIP account, call and network settings ↗
- MicroSIP troubleshooting ↗
- MicroSIP frequently asked questions ↗
- Asterisk custom dynamic features ↗
- Asterisk 23 features.conf configuration reference ↗
- Asterisk built-in dynamic features ↗
- Asterisk features overview ↗
- Asterisk upstream features.conf sample ↗
- Asterisk upstream features configuration implementation ↗
- Asterisk 23 Dial application ↗
- Asterisk channel-variable inheritance ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.