Vobiz

Send a real SMS, MMS or RCS message through Vobiz (POST /api/v1/sms/messages). Each recipient becomes its own message and is billed. Required: `from`, `to`, `idempotency_key`, and `confirm: true` after the user approved the exact sender, recipients and text. For sms/mms, `from` is one of your Vobiz messaging numbers (see list_messaging_numbers); for rcs it is your RCS sender name. Use preview_message_cost first to show the price and segment count. Opted-out recipients are refused by Vobiz (opted_out). status `scheduled` = held until scheduled_at; `sent` = handed to the network; delivery receipts update it to `delivered` or `failed` (check get_message). Repeating an idempotency_key never sends twice.

Read `text` aloud into a LIVE call with Vobiz text-to-speech (POST /Call/{uuid}/Speak/). Use it for dynamic sentences; for a pre-recorded file use play_audio, and to connect your own AI voice agent or transcription use start_audio_stream. Plain text or SSML (`type: ssml`), up to 3000 characters, spoken in `language` (default en-US) with a WOMAN (default) or MAN voice; `mix` and `loop` work as in play_audio. Speech ends when the text finishes, or earlier with stop_speaking (stop_audio does not stop it). A call that is not live fails with call_not_active.

Fork the audio of a LIVE Vobiz call to your WebSocket server in near real time (POST /Call/{uuid}/Stream/), for example for live transcription or an AI voice agent. service_url must be wss:// on an approved host. bidirectional: true lets your server play audio back into the call and requires audio_track inbound (the default when bidirectional). content_type sets the inbound audio format (L16 8 kHz default, L16 16 kHz, or mu-law 8 kHz). Returns a stream_id for get_audio_stream / stop_audio_stream. This sends live call audio to an external server, so it requires confirm: true after the user approves the destination. Fails with call_not_active once the call has ended. To play audio into the call without a socket use play_audio; to see streams already running use list_audio_streams.

Start recording the mixed audio of a live Vobiz conference room (POST /Conference/{name}/Record/) in mp3 (default) or wav. Recording may require consent from every participant in your jurisdiction, so confirm with the user first (consider announcing it). Vobiz queues the recording; the reply carries no recording ID or file URL, and the file appears in list_recordings after the room ends. Stop with stop_conference_recording. Requires confirm: true. A room that is not running returns the not_found error.

Start recording a LIVE call (POST /Call/{uuid}/Record/) into one file, either mixed (mono, default) or one party per channel (stereo). Recording ends at `time_limit` seconds (default 60, max 8 hours), when the call hangs up, or with stop_recording; the file then appears in get_call_recording / list_recordings and its storage is billed. Playback and speech on the call (play_audio, speak_text) are unaffected; to capture only a caller's answer inside an IVR use the record action of generate_answer_xml instead. Recording may require consent from the other party in your jurisdiction, so confirm with the user first. Requires confirm: true.

Stop audio started by play_audio on a LIVE call (DELETE /Call/{uuid}/Play/); the call itself continues. It does not stop text-to-speech (use stop_speaking), a recording (stop_recording) or an audio stream (stop_audio_stream). A call that is not live fails with call_not_active.

Stop one audio stream on a LIVE Vobiz call (DELETE /Call/{uuid}/Stream/{stream_id}/): the undo of start_audio_stream. Other streams on the call keep running, and the call itself continues; to end the call use end_call. No confirmation is needed because it only stops something started on a call the user already placed. If the call has ended, or the stream is unknown or already stopped, it fails with call_not_active and nothing is streaming for it.

Stop audio started by play_audio_to_conference_member for members of a live Vobiz conference (DELETE /Conference/{name}/Member/{id}/Play/). It succeeds even when nothing is playing. A room that is not running, or a member who already left, returns the not_found error.

Stop the recording started by start_conference_recording on a live Vobiz conference room (DELETE /Conference/{name}/Record/). Use it when the part you needed is captured; the conference itself keeps running and the file appears later in list_recordings. A room that is not running returns the not_found error.

Stop a recording on a LIVE call (DELETE /Call/{uuid}/Record/): pass the `url` returned by start_recording to stop that one recording, or omit it to stop every recording on the call. The file is finalised and appears in get_call_recording shortly after; the call itself, playback (stop_audio) and speech (stop_speaking) are unaffected. A call that is not live fails with call_not_active (the hangup already closed its recordings).

Stop text-to-speech started by speak_text on a LIVE call (DELETE /Call/{uuid}/Speak/); the call itself continues. It does not stop play_audio playback (use stop_audio), a recording (stop_recording) or an audio stream (stop_audio_stream). A call that is not live fails with call_not_active.

Blind-transfer a LIVE call (POST /Call/{uuid}/): Vobiz stops whatever the chosen leg is doing (IVR, playback, bridge) and fetches new VobizXML from the transfer URL. Give either `destination` (an E.164 number or sip: URI; this server hosts a <Dial> flow for it, A leg only, HTTP mode with VOBIZ_PUBLIC_URL) or your own `aleg_url` / `bleg_url` on an approved host, never both. For an attended (warm) transfer build the flow with generate_answer_xml (hold music, then dial) and pass its URL. To park a leg use hold_call, to run a keypad menu on it use gather_dtmf, to end the call use end_call. A destination number must pass the server's destination policy and becomes a new, billed call leg; a call that is not live fails with call_not_active. Requires confirm: true.

Move a number from a sub-account back to this Vobiz account's pool (DELETE /account/{id}/numbers/{e164}/assign-subaccount). It is the undo of assign_number_to_subaccount. A 15-day cool-off applies: a number with a call in the last 15 days stays with the sub-account and the tool fails with not_allowed until the cool-off ends. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Let deafened members of a live Vobiz conference hear the room again (DELETE /Conference/{name}/Member/{id}/Deaf/); the reverse of deafen_conference_member. Undeafening a member who is not deafened still succeeds. A room that is not running, or a member who already left, returns the not_found error.

Take a leg of a LIVE call off emulated hold by transferring it to `resume_url` (VobizXML on an approved host, for example a hosted flow from generate_answer_xml or the url from gather_dtmf). Vobiz cannot return to the point the call was at before hold_call, so the resume XML decides what happens next (speak, dial the other party again, a menu). To end the call instead use end_call. Only meaningful after hold_call; a call that is not live fails with call_not_active. Requires confirm: true.

Unmute members of a live Vobiz conference (DELETE /Conference/{name}/Member/{id}/Mute/) so the room can hear them again; the reverse of mute_conference_member. Unmuting a member who is not muted still succeeds. A room that is not running, or a member who already left, returns the not_found error.

Change selected fields of an existing Vobiz voice application (name, webhook URLs and methods, default-app flags). Only the fields you pass change; app_id and sub_account cannot change. Vobiz resets default_endpoint_app to false whenever it is omitted, so this tool reads the current value and re-sends it. New URLs must be https on an approved host (VOBIZ_ALLOWED_URL_HOSTS). Use create_application for a new application and update_phone_number to change which application a number uses. An unknown app_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Replace a Vobiz IP ACL entry's label and address. Both name and ip_address are required on every update; resend the current value of the one you keep. Traffic from the old address stops being trusted (changes can take a little time to reach the SIP edge); the entry keeps its id, so trunks referencing it need no change. Read the current values with list_ip_acl_entries first. The same address rules as add_ip_acl_entry apply. An unknown ip_acl_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Replace a Vobiz origination URI's label and priority, for example to promote a failover destination to primary. Both name and priority are required; resend the current value you keep. To point at a different host, create a new origination URI and delete this one; weight and enabled are managed in the Vobiz console. Read current values with list_origination_uris first. A priority change applies to the next outbound call on every trunk using the URI. An unknown uri_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Change one owned phone number's configuration: replace its tags, link it to a voice application (inbound calls then fetch that application's answer_url) or to a SIP trunk (inbound calls are delivered through the trunk), or unlink either. A number can be linked to an application or a trunk, not both; unlinks run before links, so one call can switch. The number must already be on the account (see list_phone_numbers), otherwise not_found. To buy or release a number use buy_phone_number or release_phone_number. Returns the list of changes applied. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Change a Vobiz SIP endpoint's alias, password, attached application or voice/message/video permissions. Only the fields you pass change; the username is permanent (create a new endpoint to rename). A new password signs out every registered device until it re-registers with it. Vobiz answers 202 with no body, so the result lists the fields sent rather than the new state; read it back with get_sip_endpoint. An unknown endpoint_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Change settings of an existing Vobiz SIP trunk: only the fields you pass change; trunk_id and trunk_domain never change. Use it to attach authentication (credential_uuid, ipacl_uuid), set outbound routing (primary_uri_uuid, fallback_uri_uuid), adjust limits, turn on recording or the event webhook, or pause a trunk with trunk_status: disabled instead of deleting it. Read current values first with get_sip_trunk; for a new trunk use create_sip_trunk. Returns the updated trunk object with credential fields redacted. An unknown trunk_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.

Change an existing Vobiz sub-account's name, contact details, description, rate limit, permissions, enabled flag or kyc_mode. Only the fields you pass change; omitted fields keep their values, and a call with no fields fails with invalid_input. Moving to customer_use needs an email on the sub-account and can pause its calls until its KYC is verified. Returns the updated record with credentials removed. To make a new sub-account use create_sub_account; to remove one use delete_sub_account. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true; needs main-account (MA_) credentials.

Rotate a trunk credential's password, enable or disable it, or change its description on Vobiz. The username is permanent; to rename, create a new credential and delete the old one. A new password takes effect for the next SIP authentication; only the fields you pass change. Prefer enabled: false over delete_trunk_credential when access may be needed again. Returns the updated credential with the password redacted. An unknown credential_id raises not_found. Gated by VOBIZ_ENABLE_PROVISIONING and confirm: true.