Merge Organizations into Configuration page with unified tab nav
- Add Organizations as a tab in **************** (General · Organizations · VOIP & SMS · System User · Advanced order) - Delete **************** standalone page...
Rebuild to pick up kamo-shared-library f2583e9 **************** fix)
Use @CreationTimestamp(source=DB) on ****************
The DB is the single source of truth for timestamps in this codebase. The canonical Hibernate pattern is @CreationTimestamp(source = SourceType.DB), which makes...
Let DB fill **************** via @Generated(INSERT)
INSERTs were failing with: null value in column "date_created" violates not-null constraint @ColumnDefault only declares the DB-side default for new rows; Hi...
Drop the [from-internal-custom] _X. catch-all that broke softphone calls
The catch-all exten => _X.,1 in [from-internal-custom] was intercepting every internal dial — softphones started reporting "phone is not registered" / calls wou...
Enable JPA repos for commerce + email + dao.repositories packages
ConversionService new pod CrashLoopBackOff with the same error DaemonService hit earlier: Parameter 0 of constructor in **************** required a bean of ...
Carve /api/voip/recordings/** out of the wildcard VOIP forwarder
The /api/voip/** catch-all in APIGatewayController was shadowing the dedicated VoipRecordingUploadController POST /api/voip/recordings/upload mapping when Sprin...
Write managed config files 0644 so the asterisk daemon can read them
The kamo-asterisk-support pod runs as root and writes to host-mounted **************** Asterisk inside the FreePBX pod runs as user "asterisk" and could not rea...
Use envsubst-compatible \${VAR} syntax in config template + re-render on stale placeholders
Pods were CrashLoopBackOff with "config errors: API_KEY is missing or too short" because the config template used literal __API_KEY__ markers that envsubst sile...
Drop unused watcher import
uploader.go only uses CallBatch (declared in grouper.go in the same package), not anything from internal/watcher directly. Go rejects unused imports.
Enable JPA repos for commerce + email + dao.repositories packages
DaemonService crashed on startup with: Parameter 0 of constructor in **************** required a bean of type **************** that could not be found. A r...
Surface PHONE_SERVER_ID on phone-instance overview tab
Operators configuring kamo-asterisk-support on a FreePBX host need to paste the matching VoipProviderInstance UUID into the helper's config file as PHONE_SERVER...
Capture newExtNumber as final for lambda capture in MemberVoipConfigController
newExtNumber is conditionally reassigned inside the if (isVoipEnabled && ...) block, which makes it not effectively final. The TeamMember sync block reads it fr...
Clickable VOIP extension in Interaction Center opens softphone with number prefilled
Backfill TeamMember phoneWork extension from MemberVoipConfig on startup
Wire call features settings end-to-end through provider abstraction
- voipApi: add CallFeaturesDto type, getCallFeatures() and setCallFeatures() API calls - useSoftphone: add **************** state loaded from API on startup; ...
Add call features API — call forward + auto-answer wired through provider abstraction
- Add CallFeaturesDto and SetCallFeaturesRequest POJOs - Add callForward/autoAnswer flags to ProviderCapabilities - Add default getCallFeatures/setCallFeatures ...
Add MemberCallFeatures entity and repository for call forward and auto-answer preferences
Initial commit — FreePBX recording uploader (GO)
Watches **************** for finished WAV recordings, groups multi-leg call legs by Asterisk UNIQUEID, and uploads each batch to APIService /api/voip/recordings...
VOIP_RECORDING_UPLOADS scope + recording pipeline counters/tunables
- APIManager: new VOIP_RECORDING_UPLOADS scope option (used by kamo-asterisk-support on FreePBX hosts to upload voicemail and call recordings via APIService)....
System-config recording stats + recording-processing tunables
- GET **************** returns the four counters exposed on the kamo-internal Settings → Platform → Configuration → VoIP/SMS tab (vmPending, callPending, vm...
RecordingProcessorJob + NATS shortcut for fresh uploads
- New Quartz job RecordingProcessorJob ticks every minute, dispatches independent VM and call batches concurrently to ConversionService **************** usi...
Recording ingest + FFmpeg processing pipeline
- New **************** endpoint stages multipart uploads forwarded by APIService into MinIO bucket **************** and creates an UnprocessedRecording stag...
VoipRecordingUploadController for kamo-asterisk-support uploads
POST /api/voip/recordings/upload accepts multipart bodies from kamo-asterisk-support running on FreePBX hosts. Validates the X-Api-Key has the new VOIP_RECORDIN...
UnprocessedRecording staging entity, ImgDat track tracks, recording imaging service
- Fix ImageAssocType ID collision: VM_RECORDING=8, CALL_RECORDING=9 (were both duplicating MY_DOCS at 4). - Add **************** so a master recording can car...
Initial commit — FreePBX recording uploader (GO)
Watches **************** for finished WAV recordings, groups multi-leg call legs by Asterisk UNIQUEID, and uploads each batch to APIService /api/voip/recordings...
Update OrgThemeProvisionServiceTest for buildConfigJson description parameter
Replace mock contacts tab with fully-featured SoftphoneContacts
Extracts a new SoftphoneContacts component that wires the softphone contacts tab to the real contacts API via useContacts(). Replaces 4 hardcoded mock contacts ...
Replace extension Chip with inline text next to role in Interaction Center
Remove manual phone extension field; write VOIP extension through to SecurityService on save
Include phoneWork extension in by-department response; accept phoneWorkExtension on member PUT
Cast voiceActivityDetection to bypass missing TS DOM type
voiceActivityDetection is a valid WebRTC MediaTrackConstraints property but is absent from TypeScript's DOM lib, causing a type error at build time.
Wire VAD + push-to-talk; remove unimplementable settings
- vadEnabled now passed as voiceActivityDetection in buildAudioConstraints so the browser honours the toggle when opening mic for a call - pushToTalk auto-mut...
debug+fix(softphone): inspect <audio> element state, auto-recover when bytes arrive but sound stays silent
WebRTC stats from a recent call showed inboundBytes=39680 / receivers unmuted / ICE pair succeeded — the bridge was delivering audio to the browser — yet the op...
Scrollable ringtone list with search filter
Make the ringtone catalog panel match the right-column height and scroll vertically when the list overflows, so members can navigate as more tones are added. Ad...
Dump WebRTC state again 5s after Established + track all state changes
The Established-time dump always shows receivers with muted=true because DTLS handshake and SRTP keying haven't completed yet. To diagnose "call connects but no...
Add seven more ringtones to the catalog
Adds Battleship Horn, Demon Horn, Land Line, 1985 Ring, Office Phone, Ringing Old Phone, and Olé under /public/sounds/ringtones/ and wires them into RINGTONE_CA...
Add Sci-Fi and Buzzer ringtones to the catalog
Drops two new ringtone assets under /public/sounds/ringtones/ and registers them in RINGTONE_CATALOG so they appear in the Ringtones subtab and can be set as th...
Configure STUN so browser ICE can complete bidirectionally
Two-party browser-to-browser calls connected at the SIP layer but had no audio either direction. Voicemail (one-way Asterisk → browser) worked because Asterisk'...
Dump WebRTC negotiation state on Established
To diagnose "call connects but no audio between parties", log the peer-connection state (ice / dtls / signaling) along with sender and receiver tracks (kind / e...
Live audio that survives load-interrupts, centered spinner, outbound ringback
Three coordinated fixes for the in-call audio path: 1. AbortError on remoteAudio.play() (visible in production console as "The play() request was interrupte...
Hangup tears down ALL live sessions, not just the primary
Self-call edge case left a session leaked: dialing your own extension creates an outbound session, then Asterisk loops the call back as a second inbound session...
Instant connecting feedback on outbound dial
Same problem the inbound Answer button had: clicking Call kicked off the SIP INVITE round-trip (1-3 seconds) but the button kept its idle "Call" appearance the ...
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