VoIP.info Troubleshooting

MicroSIP registers but a call or audio fails

Follow the active account, transformed destination, SIP dialog, advertised addresses, negotiated media and Windows audio device in order before changing NAT or codecs.

Reviewed 2026-10-17MicroSIP project, current-version pointer, stable downloads, help, troubleshooting and FAQ pages reviewed October 17, 2026. The current-version pointer and primary download table listed 3.22.19 for the regular and Lite editions with an October 3, 2026 release date. Package listing, installed build, active account, call authorization, negotiated signaling and media, Windows audio path and audible outcome remain separate evidence. No package, installation, Windows host, Wine environment, account, credential, PBX, provider, network, VPN, audio device, driver, SIP exchange, media stream, protected session, emergency workflow or live call was inspected or testedIntermediateDocumentation based · not lab tested

Freeze one exact 3.22.19-or-other attempt and a control

Record the MicroSIP regular or Lite edition, exact installed build, installer or portable form, Windows and driver versions, active account, Single Call Mode or Call Manager, entered destination, timestamp and timezone, network or VPN state, visible call state, SIP response, DTMF symptom and expected audio. The publisher listed 3.22.19 on October 17, 2026, but that does not establish what is installed. Protect credentials and full customer numbers.

Classify the earliest symptom: no INVITE, transformed destination, immediate rejection, remote ringing without answer, connected with no packets, one-way packets, packets without local sound, silent microphone, failed digits or fixed-duration drop. Compare with one authorized known-good destination or client using the same PBX boundary; do not update, reinstall or change several settings before the evidence is repeatable.

Prove the active identity rather than the saved account list

MicroSIP can store multiple accounts but uses one active account. Preserve the active server, proxy chain, username, domain, authorization login, transport and current registration transaction. A green or registered state establishes one transaction, not permission to call the destination, correct caller identity or a reachable media path.

If Local Account was used, classify the result as a direct IP or domain call without registration. It cannot validate registrar authentication, production inbound routing or the ordinary SIP account. Correct the account-selection boundary before changing ports, codecs or encryption.

Trace the destination and asserted identity on the installed build

Retain the value the user entered, selected contact or full SIP URI, dialing prefix and rule that transformed, allowed or blocked it. Then capture the Request-URI, From identity, P-Asserted-Identity and first final response. The 3.22.19 release note describes a fix where PAI replaced the originally dialed extension, but a release note is not evidence that one failed request used that defect or that the installed client contains the fix.

If the INVITE leaves with the correct target, move to PBX or provider route evidence. Do not infer a carrier outage from a client-side transformation, and do not remove the whole dial plan to test one number. Preserve asserted identity and caller-ID presentation separately from the destination.

Separate SIP address correction from the media address

Compare the local interface and port, Via, Contact, registrar-observed source and SDP connection address. MicroSIP's manual public-address, Allow IP Rewrite, STUN and ICE controls can affect different evidence. A local-port conflict can also move the client from the configured port to a random one.

Choose one correction only when the mismatched field is known. Do not combine manual public address, STUN, ICE, rewrite, VPN binding and router forwarding as a blanket remedy. ICE can introduce a bad candidate path when the server or network is unsuitable, and a correct Contact does not establish the RTP destination.

Distinguish RTP, DTMF and Windows capture or playback

After a successful dialog, retain negotiated codec and direction, expected RTP endpoints, packets in both directions and decode evidence. For digits, retain the configured method, SDP event mapping or INFO message, event timing and remote application result. Local DTMF playback is a different observation from transmission.

Then inspect the exact MicroSIP microphone and speaker, Windows per-app route, mute and level, device availability and driver state. The 3.22.19 microphone-muted fix and audio-device memory-corruption detection are clues, not diagnoses for an uninspected host. A negotiated codec and arriving packets do not prove playback; a Windows test tone or input meter does not prove MicroSIP captured, rendered or transmitted the call.

Distinguish RTP from Windows capture and playback

After a successful dialog, retain negotiated codec and direction, expected RTP endpoints, packets in both directions and decode evidence. Then inspect the exact MicroSIP microphone and speaker, Windows per-app device route, mute and level, device availability and driver state. A negotiated codec and arriving packets do not prove playback; a working Windows test tone does not prove MicroSIP captured or rendered the call.

The 3.22.18 changelog's audio-device or driver corruption detection is a clue to preserve logs and exact hardware state. It does not justify declaring a driver faulty without evidence. Use the wrong-audio-device diagnostic or one-way-audio diagnostic when the failure crosses those boundaries.

Correct one owner and retest the complete path

Apply the smallest reversible correction to the active account, client transformation, PBX route, address field, protection policy, session refresh, RTP path or audio device that the trace identifies. Repeat outbound and inbound calls, answer, two-way audio, DTMF and hangup; include VPN or network change only if it is part of the reported symptom.

Retain the before-and-after SIP and media evidence, the exact configuration change and rollback. Confirm emergency calling and location only with the provider's approved procedure and never with an uncoordinated live emergency call. No account, packet, device, protected session or call was tested for this diagnostic.

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.