Changelog

What's changingin v0.x.

A running log of the Connect API, versioned v0.x. The capabilities underneath are real and shipped in Dial; what changes here is the public API surface we're exposing them through.

v0.x · additive by default

Connect's public API is pre-1.0, so it is versioned in the v0.x range and breaking changes are possible before general availability. Entries are tagged by version, not by calendar date. Every API artifact referenced below is illustrative; see the API reference for current shapes.

Release history

v0.x releases.

Newest first. Each version notes what was added, changed or deprecated as the v0.x API took shape.

  1. v0.7

    Access tokens for WebRTC clients

    • AddedPOST /v1/access-tokens mints a short-lived JWT with a voice grant so a browser or mobile client can place and receive calls. The API secret never reaches the device.
    • AddedDocumented the server-mints / client-consumes model, grounded in the real single-use token rotation and JsSIP WebRTC softphone already shipped in Dial.
  2. v0.6

    Conferences & queues reference

    • AddedConference resource with a participants subresource, add, remove, hold and mute, over the real ARI snoop/bridge engine.
    • AddedACD queue resource with a members subresource, surfacing the shipped Redis-backed FIFO / longest-idle queue, MOH, maxWait/maxSize and overflow.
    • ChangedConference docs no longer imply a participant ceiling. The platform does not cap participants, so none is advertised.
  3. v0.5

    Webhooks & events subscription API

    • AddedSubscription endpoints over the 9 real lifecycle events (web-form, call-start, hangup, no-agent, disposition, callback, queued, abandoned, voicemail-left), with custom headers, JSON payload templates, retries and encrypted secrets.
    • AddedDocumented per-org host allow-listing (the SSRF guard) and signed delivery for every webhook.
    • Changedstatus_callback event names aligned to initiated / ringing / answered / completed across the calls and recordings docs.
  4. v0.4

    Recording resource & signed-URL retrieval

    • AddedRecording resource with fetch/list/delete and a RecordingStatusCallback (in-progress / completed / absent / failed), grounded in the real AES-256-GCM pipeline.
    • AddedDocumented recording modes (always / on_demand / never) with pause/resume, S3 / local / SFTP backends, wav→mp3 and audited signed-URL playback.
  5. v0.3

    Numbers & SIP/WebRTC docs

    • AddedPhoneNumbers (IncomingPhoneNumber) reference for listing assigned numbers and setting an answer URL and caller-ID rotation strategy.
    • AddedSIP trunking and WebRTC capability pages, with live MOS / codec / transport metrics surfaced from the real softphone.
    • DeprecatedMarked self-serve number search and purchase, and self-serve BYOC trunk config, as roadmap: numbers and trunks are provisioned with the team during the preview, not bought via API.
  6. v0.2

    Call control via the answer URL

    • AddedDefined the primary call-control model: your answer_url returns a JSON call-control document with play / gather / dial / record / hangup instructions.
    • AddedMapped the instruction set to the real 12-node IVR engine, including the api_call data-dip node for branching on an external API mid-call.
    • DeprecatedScoped the XML call-control markup dialect to roadmap only: the JSON answer-URL model is the supported path for the preview.
  7. v0.1

    Developer preview, first cut

    • AddedBase URL https://api.telvox.dev with flat resource paths (/v1/calls, /v1/recordings, …) and API-key bearer auth.
    • AddedCalls resource: create an outbound call (from / to / answer_url / status_callback), fetch, list and modify in-progress, over TelVox's real place/receive engine with AMD, alternate-number failover and caller-ID/DID rotation.
    • ChangedAdopted the developer-preview honesty convention: every endpoint, field, enum and JSON sample is stamped illustrative, shape may differ at GA.

more lines coming

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