telnyx 3.0.0__py3-none-any.whl

telnyx/resources/calls/calls.py

@@ -0,0 +1,866 @@
+# File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+
+from __future__ import annotations
+
+from typing import Union, Iterable
+from typing_extensions import Literal
+
+import httpx
+
+from ...types import (
+    StreamCodec,
+    StreamBidirectionalMode,
+    StreamBidirectionalCodec,
+    StreamBidirectionalTargetLegs,
+    call_dial_params,
+)
+from .actions import (
+    ActionsResource,
+    AsyncActionsResource,
+    ActionsResourceWithRawResponse,
+    AsyncActionsResourceWithRawResponse,
+    ActionsResourceWithStreamingResponse,
+    AsyncActionsResourceWithStreamingResponse,
+)
+from ..._types import Body, Omit, Query, Headers, NotGiven, SequenceNotStr, omit, not_given
+from ..._utils import maybe_transform, async_maybe_transform
+from ..._compat import cached_property
+from ..._resource import SyncAPIResource, AsyncAPIResource
+from ..._response import (
+    to_raw_response_wrapper,
+    to_streamed_response_wrapper,
+    async_to_raw_response_wrapper,
+    async_to_streamed_response_wrapper,
+)
+from ..._base_client import make_request_options
+from ...types.stream_codec import StreamCodec
+from ...types.sip_header_param import SipHeaderParam
+from ...types.call_dial_response import CallDialResponse
+from ...types.custom_sip_header_param import CustomSipHeaderParam
+from ...types.dialogflow_config_param import DialogflowConfigParam
+from ...types.sound_modifications_param import SoundModificationsParam
+from ...types.stream_bidirectional_mode import StreamBidirectionalMode
+from ...types.stream_bidirectional_codec import StreamBidirectionalCodec
+from ...types.call_retrieve_status_response import CallRetrieveStatusResponse
+from ...types.stream_bidirectional_target_legs import StreamBidirectionalTargetLegs
+from ...types.calls.transcription_start_request_param import TranscriptionStartRequestParam
+
+__all__ = ["CallsResource", "AsyncCallsResource"]
+
+
+class CallsResource(SyncAPIResource):
+    @cached_property
+    def actions(self) -> ActionsResource:
+        return ActionsResource(self._client)
+
+    @cached_property
+    def with_raw_response(self) -> CallsResourceWithRawResponse:
+        """
+        This property can be used as a prefix for any HTTP method call to return
+        the raw response object instead of the parsed content.
+
+        For more information, see https://www.github.com/team-telnyx/telnyx-python#accessing-raw-response-data-eg-headers
+        """
+        return CallsResourceWithRawResponse(self)
+
+    @cached_property
+    def with_streaming_response(self) -> CallsResourceWithStreamingResponse:
+        """
+        An alternative to `.with_raw_response` that doesn't eagerly read the response body.
+
+        For more information, see https://www.github.com/team-telnyx/telnyx-python#with_streaming_response
+        """
+        return CallsResourceWithStreamingResponse(self)
+
+    def dial(
+        self,
+        *,
+        connection_id: str,
+        from_: str,
+        to: Union[str, SequenceNotStr[str]],
+        answering_machine_detection: Literal[
+            "premium", "detect", "detect_beep", "detect_words", "greeting_end", "disabled"
+        ]
+        | Omit = omit,
+        answering_machine_detection_config: call_dial_params.AnsweringMachineDetectionConfig | Omit = omit,
+        audio_url: str | Omit = omit,
+        billing_group_id: str | Omit = omit,
+        bridge_intent: bool | Omit = omit,
+        bridge_on_answer: bool | Omit = omit,
+        client_state: str | Omit = omit,
+        command_id: str | Omit = omit,
+        conference_config: call_dial_params.ConferenceConfig | Omit = omit,
+        custom_headers: Iterable[CustomSipHeaderParam] | Omit = omit,
+        dialogflow_config: DialogflowConfigParam | Omit = omit,
+        enable_dialogflow: bool | Omit = omit,
+        from_display_name: str | Omit = omit,
+        link_to: str | Omit = omit,
+        media_encryption: Literal["disabled", "SRTP", "DTLS"] | Omit = omit,
+        media_name: str | Omit = omit,
+        preferred_codecs: str | Omit = omit,
+        record: Literal["record-from-answer"] | Omit = omit,
+        record_channels: Literal["single", "dual"] | Omit = omit,
+        record_custom_file_name: str | Omit = omit,
+        record_format: Literal["wav", "mp3"] | Omit = omit,
+        record_max_length: int | Omit = omit,
+        record_timeout_secs: int | Omit = omit,
+        record_track: Literal["both", "inbound", "outbound"] | Omit = omit,
+        record_trim: Literal["trim-silence"] | Omit = omit,
+        send_silence_when_idle: bool | Omit = omit,
+        sip_auth_password: str | Omit = omit,
+        sip_auth_username: str | Omit = omit,
+        sip_headers: Iterable[SipHeaderParam] | Omit = omit,
+        sip_transport_protocol: Literal["UDP", "TCP", "TLS"] | Omit = omit,
+        sound_modifications: SoundModificationsParam | Omit = omit,
+        stream_bidirectional_codec: StreamBidirectionalCodec | Omit = omit,
+        stream_bidirectional_mode: StreamBidirectionalMode | Omit = omit,
+        stream_bidirectional_sampling_rate: Literal[8000, 16000, 22050, 24000, 48000] | Omit = omit,
+        stream_bidirectional_target_legs: StreamBidirectionalTargetLegs | Omit = omit,
+        stream_codec: StreamCodec | Omit = omit,
+        stream_establish_before_call_originate: bool | Omit = omit,
+        stream_track: Literal["inbound_track", "outbound_track", "both_tracks"] | Omit = omit,
+        stream_url: str | Omit = omit,
+        supervise_call_control_id: str | Omit = omit,
+        supervisor_role: Literal["barge", "whisper", "monitor"] | Omit = omit,
+        time_limit_secs: int | Omit = omit,
+        timeout_secs: int | Omit = omit,
+        transcription: bool | Omit = omit,
+        transcription_config: TranscriptionStartRequestParam | Omit = omit,
+        webhook_url: str | Omit = omit,
+        webhook_url_method: Literal["POST", "GET"] | Omit = omit,
+        # Use the following arguments if you need to pass additional parameters to the API that aren't available via kwargs.
+        # The extra values given here take precedence over values defined on the client or passed to this method.
+        extra_headers: Headers | None = None,
+        extra_query: Query | None = None,
+        extra_body: Body | None = None,
+        timeout: float | httpx.Timeout | None | NotGiven = not_given,
+    ) -> CallDialResponse:
+        """Dial a number or SIP URI from a given connection.
+
+        A successful response will
+        include a `call_leg_id` which can be used to correlate the command with
+        subsequent webhooks.
+
+        **Expected Webhooks:**
+
+        - `call.initiated`
+        - `call.answered` or `call.hangup`
+        - `call.machine.detection.ended` if `answering_machine_detection` was requested
+        - `call.machine.greeting.ended` if `answering_machine_detection` was requested
+          to detect the end of machine greeting
+        - `call.machine.premium.detection.ended` if
+          `answering_machine_detection=premium` was requested
+        - `call.machine.premium.greeting.ended` if `answering_machine_detection=premium`
+          was requested and a beep was detected
+        - `streaming.started`, `streaming.stopped` or `streaming.failed` if `stream_url`
+          was set
+
+        When the `record` parameter is set to `record-from-answer`, the response will
+        include a `recording_id` field.
+
+        Args:
+          connection_id: The ID of the Call Control App (formerly ID of the connection) to be used when
+              dialing the destination.
+
+          from_: The `from` number to be used as the caller id presented to the destination (`to`
+              number). The number should be in +E164 format.
+
+          to: The DID or SIP URI to dial out to. Multiple DID or SIP URIs can be provided
+              using an array of strings
+
+          answering_machine_detection: Enables Answering Machine Detection. Telnyx offers Premium and Standard
+              detections. With Premium detection, when a call is answered, Telnyx runs
+              real-time detection and sends a `call.machine.premium.detection.ended` webhook
+              with one of the following results: `human_residence`, `human_business`,
+              `machine`, `silence` or `fax_detected`. If we detect a beep, we also send a
+              `call.machine.premium.greeting.ended` webhook with the result of
+              `beep_detected`. If we detect a beep before
+              `call.machine.premium.detection.ended` we only send
+              `call.machine.premium.greeting.ended`, and if we detect a beep after
+              `call.machine.premium.detection.ended`, we send both webhooks. With Standard
+              detection, when a call is answered, Telnyx runs real-time detection to determine
+              if it was picked up by a human or a machine and sends an
+              `call.machine.detection.ended` webhook with the analysis result. If
+              `greeting_end` or `detect_words` is used and a `machine` is detected, you will
+              receive another `call.machine.greeting.ended` webhook when the answering machine
+              greeting ends with a beep or silence. If `detect_beep` is used, you will only
+              receive `call.machine.greeting.ended` if a beep is detected.
+
+          answering_machine_detection_config: Optional configuration parameters to modify 'answering_machine_detection'
+              performance.
+
+          audio_url: The URL of a file to be played back to the callee when the call is answered. The
+              URL can point to either a WAV or MP3 file. media_name and audio_url cannot be
+              used together in one request.
+
+          billing_group_id: Use this field to set the Billing Group ID for the call. Must be a valid and
+              existing Billing Group ID.
+
+          bridge_intent: Indicates the intent to bridge this call with the call specified in link_to.
+              When bridge_intent is true, link_to becomes required and the from number will be
+              overwritten by the from number from the linked call.
+
+          bridge_on_answer: Whether to automatically bridge answered call to the call specified in link_to.
+              When bridge_on_answer is true, link_to becomes required.
+
+          client_state: Use this field to add state to every subsequent webhook. It must be a valid
+              Base-64 encoded string.
+
+          command_id: Use this field to avoid duplicate commands. Telnyx will ignore others Dial
+              commands with the same `command_id`.
+
+          conference_config: Optional configuration parameters to dial new participant into a conference.
+
+          custom_headers: Custom headers to be added to the SIP INVITE.
+
+          enable_dialogflow: Enables Dialogflow for the current call. The default value is false.
+
+          from_display_name: The `from_display_name` string to be used as the caller id name (SIP From
+              Display Name) presented to the destination (`to` number). The string should have
+              a maximum of 128 characters, containing only letters, numbers, spaces, and
+              -\\__~!.+ special characters. If ommited, the display name will be the same as the
+              number in the `from` field.
+
+          link_to: Use another call's control id for sharing the same call session id
+
+          media_encryption: Defines whether media should be encrypted on the call.
+
+          media_name: The media_name of a file to be played back to the callee when the call is
+              answered. The media_name must point to a file previously uploaded to
+              api.telnyx.com/v2/media by the same user/organization. The file must either be a
+              WAV or MP3 file.
+
+          preferred_codecs: The list of comma-separated codecs in a preferred order for the forked media to
+              be received.
+
+          record: Start recording automatically after an event. Disabled by default.
+
+          record_channels: Defines which channel should be recorded ('single' or 'dual') when `record` is
+              specified.
+
+          record_custom_file_name: The custom recording file name to be used instead of the default `call_leg_id`.
+              Telnyx will still add a Unix timestamp suffix.
+
+          record_format: Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
+
+          record_max_length: Defines the maximum length for the recording in seconds when `record` is
+              specified. The minimum value is 0. The maximum value is 43200. The default value
+              is 0 (infinite).
+
+          record_timeout_secs: The number of seconds that Telnyx will wait for the recording to be stopped if
+              silence is detected when `record` is specified. The timer only starts when the
+              speech is detected. Please note that call transcription is used to detect
+              silence and the related charge will be applied. The minimum value is 0. The
+              default value is 0 (infinite).
+
+          record_track: The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
+              If only single track is specified (`inbound`, `outbound`), `channels`
+              configuration is ignored and it will be recorded as mono (single channel).
+
+          record_trim: When set to `trim-silence`, silence will be removed from the beginning and end
+              of the recording.
+
+          send_silence_when_idle: Generate silence RTP packets when no transmission available.
+
+          sip_auth_password: SIP Authentication password used for SIP challenges.
+
+          sip_auth_username: SIP Authentication username used for SIP challenges.
+
+          sip_headers: SIP headers to be added to the SIP INVITE request. Currently only User-to-User
+              header is supported.
+
+          sip_transport_protocol: Defines SIP transport protocol to be used on the call.
+
+          sound_modifications: Use this field to modify sound effects, for example adjust the pitch.
+
+          stream_bidirectional_codec: Indicates codec for bidirectional streaming RTP payloads. Used only with
+              stream_bidirectional_mode=rtp. Case sensitive.
+
+          stream_bidirectional_mode: Configures method of bidirectional streaming (mp3, rtp).
+
+          stream_bidirectional_sampling_rate: Audio sampling rate.
+
+          stream_bidirectional_target_legs: Specifies which call legs should receive the bidirectional stream audio.
+
+          stream_codec: Specifies the codec to be used for the streamed audio. When set to 'default' or
+              when transcoding is not possible, the codec from the call will be used.
+
+          stream_establish_before_call_originate: Establish websocket connection before dialing the destination. This is useful
+              for cases where the websocket connection takes a long time to establish.
+
+          stream_track: Specifies which track should be streamed.
+
+          stream_url: The destination WebSocket address where the stream is going to be delivered.
+
+          supervise_call_control_id: The call leg which will be supervised by the new call.
+
+          supervisor_role: The role of the supervisor call. 'barge' means that supervisor call hears and is
+              being heard by both ends of the call (caller & callee). 'whisper' means that
+              only supervised_call_control_id hears supervisor but supervisor can hear
+              everything. 'monitor' means that nobody can hear supervisor call, but supervisor
+              can hear everything on the call.
+
+          time_limit_secs: Sets the maximum duration of a Call Control Leg in seconds. If the time limit is
+              reached, the call will hangup and a `call.hangup` webhook with a `hangup_cause`
+              of `time_limit` will be sent. For example, by setting a time limit of 120
+              seconds, a Call Leg will be automatically terminated two minutes after being
+              answered. The default time limit is 14400 seconds or 4 hours and this is also
+              the maximum allowed call length.
+
+          timeout_secs: The number of seconds that Telnyx will wait for the call to be answered by the
+              destination to which it is being called. If the timeout is reached before an
+              answer is received, the call will hangup and a `call.hangup` webhook with a
+              `hangup_cause` of `timeout` will be sent. Minimum value is 5 seconds. Maximum
+              value is 600 seconds.
+
+          transcription: Enable transcription upon call answer. The default value is false.
+
+          webhook_url: Use this field to override the URL for which Telnyx will send subsequent
+              webhooks to for this call.
+
+          webhook_url_method: HTTP request type used for `webhook_url`.
+
+          extra_headers: Send extra headers
+
+          extra_query: Add additional query parameters to the request
+
+          extra_body: Add additional JSON properties to the request
+
+          timeout: Override the client-level default timeout for this request, in seconds
+        """
+        return self._post(
+            "/calls",
+            body=maybe_transform(
+                {
+                    "connection_id": connection_id,
+                    "from_": from_,
+                    "to": to,
+                    "answering_machine_detection": answering_machine_detection,
+                    "answering_machine_detection_config": answering_machine_detection_config,
+                    "audio_url": audio_url,
+                    "billing_group_id": billing_group_id,
+                    "bridge_intent": bridge_intent,
+                    "bridge_on_answer": bridge_on_answer,
+                    "client_state": client_state,
+                    "command_id": command_id,
+                    "conference_config": conference_config,
+                    "custom_headers": custom_headers,
+                    "dialogflow_config": dialogflow_config,
+                    "enable_dialogflow": enable_dialogflow,
+                    "from_display_name": from_display_name,
+                    "link_to": link_to,
+                    "media_encryption": media_encryption,
+                    "media_name": media_name,
+                    "preferred_codecs": preferred_codecs,
+                    "record": record,
+                    "record_channels": record_channels,
+                    "record_custom_file_name": record_custom_file_name,
+                    "record_format": record_format,
+                    "record_max_length": record_max_length,
+                    "record_timeout_secs": record_timeout_secs,
+                    "record_track": record_track,
+                    "record_trim": record_trim,
+                    "send_silence_when_idle": send_silence_when_idle,
+                    "sip_auth_password": sip_auth_password,
+                    "sip_auth_username": sip_auth_username,
+                    "sip_headers": sip_headers,
+                    "sip_transport_protocol": sip_transport_protocol,
+                    "sound_modifications": sound_modifications,
+                    "stream_bidirectional_codec": stream_bidirectional_codec,
+                    "stream_bidirectional_mode": stream_bidirectional_mode,
+                    "stream_bidirectional_sampling_rate": stream_bidirectional_sampling_rate,
+                    "stream_bidirectional_target_legs": stream_bidirectional_target_legs,
+                    "stream_codec": stream_codec,
+                    "stream_establish_before_call_originate": stream_establish_before_call_originate,
+                    "stream_track": stream_track,
+                    "stream_url": stream_url,
+                    "supervise_call_control_id": supervise_call_control_id,
+                    "supervisor_role": supervisor_role,
+                    "time_limit_secs": time_limit_secs,
+                    "timeout_secs": timeout_secs,
+                    "transcription": transcription,
+                    "transcription_config": transcription_config,
+                    "webhook_url": webhook_url,
+                    "webhook_url_method": webhook_url_method,
+                },
+                call_dial_params.CallDialParams,
+            ),
+            options=make_request_options(
+                extra_headers=extra_headers, extra_query=extra_query, extra_body=extra_body, timeout=timeout
+            ),
+            cast_to=CallDialResponse,
+        )
+
+    def retrieve_status(
+        self,
+        call_control_id: str,
+        *,
+        # Use the following arguments if you need to pass additional parameters to the API that aren't available via kwargs.
+        # The extra values given here take precedence over values defined on the client or passed to this method.
+        extra_headers: Headers | None = None,
+        extra_query: Query | None = None,
+        extra_body: Body | None = None,
+        timeout: float | httpx.Timeout | None | NotGiven = not_given,
+    ) -> CallRetrieveStatusResponse:
+        """
+        Returns the status of a call (data is available 10 minutes after call ended).
+
+        Args:
+          extra_headers: Send extra headers
+
+          extra_query: Add additional query parameters to the request
+
+          extra_body: Add additional JSON properties to the request
+
+          timeout: Override the client-level default timeout for this request, in seconds
+        """
+        if not call_control_id:
+            raise ValueError(f"Expected a non-empty value for `call_control_id` but received {call_control_id!r}")
+        return self._get(
+            f"/calls/{call_control_id}",
+            options=make_request_options(
+                extra_headers=extra_headers, extra_query=extra_query, extra_body=extra_body, timeout=timeout
+            ),
+            cast_to=CallRetrieveStatusResponse,
+        )
+
+
+class AsyncCallsResource(AsyncAPIResource):
+    @cached_property
+    def actions(self) -> AsyncActionsResource:
+        return AsyncActionsResource(self._client)
+
+    @cached_property
+    def with_raw_response(self) -> AsyncCallsResourceWithRawResponse:
+        """
+        This property can be used as a prefix for any HTTP method call to return
+        the raw response object instead of the parsed content.
+
+        For more information, see https://www.github.com/team-telnyx/telnyx-python#accessing-raw-response-data-eg-headers
+        """
+        return AsyncCallsResourceWithRawResponse(self)
+
+    @cached_property
+    def with_streaming_response(self) -> AsyncCallsResourceWithStreamingResponse:
+        """
+        An alternative to `.with_raw_response` that doesn't eagerly read the response body.
+
+        For more information, see https://www.github.com/team-telnyx/telnyx-python#with_streaming_response
+        """
+        return AsyncCallsResourceWithStreamingResponse(self)
+
+    async def dial(
+        self,
+        *,
+        connection_id: str,
+        from_: str,
+        to: Union[str, SequenceNotStr[str]],
+        answering_machine_detection: Literal[
+            "premium", "detect", "detect_beep", "detect_words", "greeting_end", "disabled"
+        ]
+        | Omit = omit,
+        answering_machine_detection_config: call_dial_params.AnsweringMachineDetectionConfig | Omit = omit,
+        audio_url: str | Omit = omit,
+        billing_group_id: str | Omit = omit,
+        bridge_intent: bool | Omit = omit,
+        bridge_on_answer: bool | Omit = omit,
+        client_state: str | Omit = omit,
+        command_id: str | Omit = omit,
+        conference_config: call_dial_params.ConferenceConfig | Omit = omit,
+        custom_headers: Iterable[CustomSipHeaderParam] | Omit = omit,
+        dialogflow_config: DialogflowConfigParam | Omit = omit,
+        enable_dialogflow: bool | Omit = omit,
+        from_display_name: str | Omit = omit,
+        link_to: str | Omit = omit,
+        media_encryption: Literal["disabled", "SRTP", "DTLS"] | Omit = omit,
+        media_name: str | Omit = omit,
+        preferred_codecs: str | Omit = omit,
+        record: Literal["record-from-answer"] | Omit = omit,
+        record_channels: Literal["single", "dual"] | Omit = omit,
+        record_custom_file_name: str | Omit = omit,
+        record_format: Literal["wav", "mp3"] | Omit = omit,
+        record_max_length: int | Omit = omit,
+        record_timeout_secs: int | Omit = omit,
+        record_track: Literal["both", "inbound", "outbound"] | Omit = omit,
+        record_trim: Literal["trim-silence"] | Omit = omit,
+        send_silence_when_idle: bool | Omit = omit,
+        sip_auth_password: str | Omit = omit,
+        sip_auth_username: str | Omit = omit,
+        sip_headers: Iterable[SipHeaderParam] | Omit = omit,
+        sip_transport_protocol: Literal["UDP", "TCP", "TLS"] | Omit = omit,
+        sound_modifications: SoundModificationsParam | Omit = omit,
+        stream_bidirectional_codec: StreamBidirectionalCodec | Omit = omit,
+        stream_bidirectional_mode: StreamBidirectionalMode | Omit = omit,
+        stream_bidirectional_sampling_rate: Literal[8000, 16000, 22050, 24000, 48000] | Omit = omit,
+        stream_bidirectional_target_legs: StreamBidirectionalTargetLegs | Omit = omit,
+        stream_codec: StreamCodec | Omit = omit,
+        stream_establish_before_call_originate: bool | Omit = omit,
+        stream_track: Literal["inbound_track", "outbound_track", "both_tracks"] | Omit = omit,
+        stream_url: str | Omit = omit,
+        supervise_call_control_id: str | Omit = omit,
+        supervisor_role: Literal["barge", "whisper", "monitor"] | Omit = omit,
+        time_limit_secs: int | Omit = omit,
+        timeout_secs: int | Omit = omit,
+        transcription: bool | Omit = omit,
+        transcription_config: TranscriptionStartRequestParam | Omit = omit,
+        webhook_url: str | Omit = omit,
+        webhook_url_method: Literal["POST", "GET"] | Omit = omit,
+        # Use the following arguments if you need to pass additional parameters to the API that aren't available via kwargs.
+        # The extra values given here take precedence over values defined on the client or passed to this method.
+        extra_headers: Headers | None = None,
+        extra_query: Query | None = None,
+        extra_body: Body | None = None,
+        timeout: float | httpx.Timeout | None | NotGiven = not_given,
+    ) -> CallDialResponse:
+        """Dial a number or SIP URI from a given connection.
+
+        A successful response will
+        include a `call_leg_id` which can be used to correlate the command with
+        subsequent webhooks.
+
+        **Expected Webhooks:**
+
+        - `call.initiated`
+        - `call.answered` or `call.hangup`
+        - `call.machine.detection.ended` if `answering_machine_detection` was requested
+        - `call.machine.greeting.ended` if `answering_machine_detection` was requested
+          to detect the end of machine greeting
+        - `call.machine.premium.detection.ended` if
+          `answering_machine_detection=premium` was requested
+        - `call.machine.premium.greeting.ended` if `answering_machine_detection=premium`
+          was requested and a beep was detected
+        - `streaming.started`, `streaming.stopped` or `streaming.failed` if `stream_url`
+          was set
+
+        When the `record` parameter is set to `record-from-answer`, the response will
+        include a `recording_id` field.
+
+        Args:
+          connection_id: The ID of the Call Control App (formerly ID of the connection) to be used when
+              dialing the destination.
+
+          from_: The `from` number to be used as the caller id presented to the destination (`to`
+              number). The number should be in +E164 format.
+
+          to: The DID or SIP URI to dial out to. Multiple DID or SIP URIs can be provided
+              using an array of strings
+
+          answering_machine_detection: Enables Answering Machine Detection. Telnyx offers Premium and Standard
+              detections. With Premium detection, when a call is answered, Telnyx runs
+              real-time detection and sends a `call.machine.premium.detection.ended` webhook
+              with one of the following results: `human_residence`, `human_business`,
+              `machine`, `silence` or `fax_detected`. If we detect a beep, we also send a
+              `call.machine.premium.greeting.ended` webhook with the result of
+              `beep_detected`. If we detect a beep before
+              `call.machine.premium.detection.ended` we only send
+              `call.machine.premium.greeting.ended`, and if we detect a beep after
+              `call.machine.premium.detection.ended`, we send both webhooks. With Standard
+              detection, when a call is answered, Telnyx runs real-time detection to determine
+              if it was picked up by a human or a machine and sends an
+              `call.machine.detection.ended` webhook with the analysis result. If
+              `greeting_end` or `detect_words` is used and a `machine` is detected, you will
+              receive another `call.machine.greeting.ended` webhook when the answering machine
+              greeting ends with a beep or silence. If `detect_beep` is used, you will only
+              receive `call.machine.greeting.ended` if a beep is detected.
+
+          answering_machine_detection_config: Optional configuration parameters to modify 'answering_machine_detection'
+              performance.
+
+          audio_url: The URL of a file to be played back to the callee when the call is answered. The
+              URL can point to either a WAV or MP3 file. media_name and audio_url cannot be
+              used together in one request.
+
+          billing_group_id: Use this field to set the Billing Group ID for the call. Must be a valid and
+              existing Billing Group ID.
+
+          bridge_intent: Indicates the intent to bridge this call with the call specified in link_to.
+              When bridge_intent is true, link_to becomes required and the from number will be
+              overwritten by the from number from the linked call.
+
+          bridge_on_answer: Whether to automatically bridge answered call to the call specified in link_to.
+              When bridge_on_answer is true, link_to becomes required.
+
+          client_state: Use this field to add state to every subsequent webhook. It must be a valid
+              Base-64 encoded string.
+
+          command_id: Use this field to avoid duplicate commands. Telnyx will ignore others Dial
+              commands with the same `command_id`.
+
+          conference_config: Optional configuration parameters to dial new participant into a conference.
+
+          custom_headers: Custom headers to be added to the SIP INVITE.
+
+          enable_dialogflow: Enables Dialogflow for the current call. The default value is false.
+
+          from_display_name: The `from_display_name` string to be used as the caller id name (SIP From
+              Display Name) presented to the destination (`to` number). The string should have
+              a maximum of 128 characters, containing only letters, numbers, spaces, and
+              -\\__~!.+ special characters. If ommited, the display name will be the same as the
+              number in the `from` field.
+
+          link_to: Use another call's control id for sharing the same call session id
+
+          media_encryption: Defines whether media should be encrypted on the call.
+
+          media_name: The media_name of a file to be played back to the callee when the call is
+              answered. The media_name must point to a file previously uploaded to
+              api.telnyx.com/v2/media by the same user/organization. The file must either be a
+              WAV or MP3 file.
+
+          preferred_codecs: The list of comma-separated codecs in a preferred order for the forked media to
+              be received.
+
+          record: Start recording automatically after an event. Disabled by default.
+
+          record_channels: Defines which channel should be recorded ('single' or 'dual') when `record` is
+              specified.
+
+          record_custom_file_name: The custom recording file name to be used instead of the default `call_leg_id`.
+              Telnyx will still add a Unix timestamp suffix.
+
+          record_format: Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
+
+          record_max_length: Defines the maximum length for the recording in seconds when `record` is
+              specified. The minimum value is 0. The maximum value is 43200. The default value
+              is 0 (infinite).
+
+          record_timeout_secs: The number of seconds that Telnyx will wait for the recording to be stopped if
+              silence is detected when `record` is specified. The timer only starts when the
+              speech is detected. Please note that call transcription is used to detect
+              silence and the related charge will be applied. The minimum value is 0. The
+              default value is 0 (infinite).
+
+          record_track: The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
+              If only single track is specified (`inbound`, `outbound`), `channels`
+              configuration is ignored and it will be recorded as mono (single channel).
+
+          record_trim: When set to `trim-silence`, silence will be removed from the beginning and end
+              of the recording.
+
+          send_silence_when_idle: Generate silence RTP packets when no transmission available.
+
+          sip_auth_password: SIP Authentication password used for SIP challenges.
+
+          sip_auth_username: SIP Authentication username used for SIP challenges.
+
+          sip_headers: SIP headers to be added to the SIP INVITE request. Currently only User-to-User
+              header is supported.
+
+          sip_transport_protocol: Defines SIP transport protocol to be used on the call.
+
+          sound_modifications: Use this field to modify sound effects, for example adjust the pitch.
+
+          stream_bidirectional_codec: Indicates codec for bidirectional streaming RTP payloads. Used only with
+              stream_bidirectional_mode=rtp. Case sensitive.
+
+          stream_bidirectional_mode: Configures method of bidirectional streaming (mp3, rtp).
+
+          stream_bidirectional_sampling_rate: Audio sampling rate.
+
+          stream_bidirectional_target_legs: Specifies which call legs should receive the bidirectional stream audio.
+
+          stream_codec: Specifies the codec to be used for the streamed audio. When set to 'default' or
+              when transcoding is not possible, the codec from the call will be used.
+
+          stream_establish_before_call_originate: Establish websocket connection before dialing the destination. This is useful
+              for cases where the websocket connection takes a long time to establish.
+
+          stream_track: Specifies which track should be streamed.
+
+          stream_url: The destination WebSocket address where the stream is going to be delivered.
+
+          supervise_call_control_id: The call leg which will be supervised by the new call.
+
+          supervisor_role: The role of the supervisor call. 'barge' means that supervisor call hears and is
+              being heard by both ends of the call (caller & callee). 'whisper' means that
+              only supervised_call_control_id hears supervisor but supervisor can hear
+              everything. 'monitor' means that nobody can hear supervisor call, but supervisor
+              can hear everything on the call.
+
+          time_limit_secs: Sets the maximum duration of a Call Control Leg in seconds. If the time limit is
+              reached, the call will hangup and a `call.hangup` webhook with a `hangup_cause`
+              of `time_limit` will be sent. For example, by setting a time limit of 120
+              seconds, a Call Leg will be automatically terminated two minutes after being
+              answered. The default time limit is 14400 seconds or 4 hours and this is also
+              the maximum allowed call length.
+
+          timeout_secs: The number of seconds that Telnyx will wait for the call to be answered by the
+              destination to which it is being called. If the timeout is reached before an
+              answer is received, the call will hangup and a `call.hangup` webhook with a
+              `hangup_cause` of `timeout` will be sent. Minimum value is 5 seconds. Maximum
+              value is 600 seconds.
+
+          transcription: Enable transcription upon call answer. The default value is false.
+
+          webhook_url: Use this field to override the URL for which Telnyx will send subsequent
+              webhooks to for this call.
+
+          webhook_url_method: HTTP request type used for `webhook_url`.
+
+          extra_headers: Send extra headers
+
+          extra_query: Add additional query parameters to the request
+
+          extra_body: Add additional JSON properties to the request
+
+          timeout: Override the client-level default timeout for this request, in seconds
+        """
+        return await self._post(
+            "/calls",
+            body=await async_maybe_transform(
+                {
+                    "connection_id": connection_id,
+                    "from_": from_,
+                    "to": to,
+                    "answering_machine_detection": answering_machine_detection,
+                    "answering_machine_detection_config": answering_machine_detection_config,
+                    "audio_url": audio_url,
+                    "billing_group_id": billing_group_id,
+                    "bridge_intent": bridge_intent,
+                    "bridge_on_answer": bridge_on_answer,
+                    "client_state": client_state,
+                    "command_id": command_id,
+                    "conference_config": conference_config,
+                    "custom_headers": custom_headers,
+                    "dialogflow_config": dialogflow_config,
+                    "enable_dialogflow": enable_dialogflow,
+                    "from_display_name": from_display_name,
+                    "link_to": link_to,
+                    "media_encryption": media_encryption,
+                    "media_name": media_name,
+                    "preferred_codecs": preferred_codecs,
+                    "record": record,
+                    "record_channels": record_channels,
+                    "record_custom_file_name": record_custom_file_name,
+                    "record_format": record_format,
+                    "record_max_length": record_max_length,
+                    "record_timeout_secs": record_timeout_secs,
+                    "record_track": record_track,
+                    "record_trim": record_trim,
+                    "send_silence_when_idle": send_silence_when_idle,
+                    "sip_auth_password": sip_auth_password,
+                    "sip_auth_username": sip_auth_username,
+                    "sip_headers": sip_headers,
+                    "sip_transport_protocol": sip_transport_protocol,
+                    "sound_modifications": sound_modifications,
+                    "stream_bidirectional_codec": stream_bidirectional_codec,
+                    "stream_bidirectional_mode": stream_bidirectional_mode,
+                    "stream_bidirectional_sampling_rate": stream_bidirectional_sampling_rate,
+                    "stream_bidirectional_target_legs": stream_bidirectional_target_legs,
+                    "stream_codec": stream_codec,
+                    "stream_establish_before_call_originate": stream_establish_before_call_originate,
+                    "stream_track": stream_track,
+                    "stream_url": stream_url,
+                    "supervise_call_control_id": supervise_call_control_id,
+                    "supervisor_role": supervisor_role,
+                    "time_limit_secs": time_limit_secs,
+                    "timeout_secs": timeout_secs,
+                    "transcription": transcription,
+                    "transcription_config": transcription_config,
+                    "webhook_url": webhook_url,
+                    "webhook_url_method": webhook_url_method,
+                },
+                call_dial_params.CallDialParams,
+            ),
+            options=make_request_options(
+                extra_headers=extra_headers, extra_query=extra_query, extra_body=extra_body, timeout=timeout
+            ),
+            cast_to=CallDialResponse,
+        )
+
+    async def retrieve_status(
+        self,
+        call_control_id: str,
+        *,
+        # Use the following arguments if you need to pass additional parameters to the API that aren't available via kwargs.
+        # The extra values given here take precedence over values defined on the client or passed to this method.
+        extra_headers: Headers | None = None,
+        extra_query: Query | None = None,
+        extra_body: Body | None = None,
+        timeout: float | httpx.Timeout | None | NotGiven = not_given,
+    ) -> CallRetrieveStatusResponse:
+        """
+        Returns the status of a call (data is available 10 minutes after call ended).
+
+        Args:
+          extra_headers: Send extra headers
+
+          extra_query: Add additional query parameters to the request
+
+          extra_body: Add additional JSON properties to the request
+
+          timeout: Override the client-level default timeout for this request, in seconds
+        """
+        if not call_control_id:
+            raise ValueError(f"Expected a non-empty value for `call_control_id` but received {call_control_id!r}")
+        return await self._get(
+            f"/calls/{call_control_id}",
+            options=make_request_options(
+                extra_headers=extra_headers, extra_query=extra_query, extra_body=extra_body, timeout=timeout
+            ),
+            cast_to=CallRetrieveStatusResponse,
+        )
+
+
+class CallsResourceWithRawResponse:
+    def __init__(self, calls: CallsResource) -> None:
+        self._calls = calls
+
+        self.dial = to_raw_response_wrapper(
+            calls.dial,
+        )
+        self.retrieve_status = to_raw_response_wrapper(
+            calls.retrieve_status,
+        )
+
+    @cached_property
+    def actions(self) -> ActionsResourceWithRawResponse:
+        return ActionsResourceWithRawResponse(self._calls.actions)
+
+
+class AsyncCallsResourceWithRawResponse:
+    def __init__(self, calls: AsyncCallsResource) -> None:
+        self._calls = calls
+
+        self.dial = async_to_raw_response_wrapper(
+            calls.dial,
+        )
+        self.retrieve_status = async_to_raw_response_wrapper(
+            calls.retrieve_status,
+        )
+
+    @cached_property
+    def actions(self) -> AsyncActionsResourceWithRawResponse:
+        return AsyncActionsResourceWithRawResponse(self._calls.actions)
+
+
+class CallsResourceWithStreamingResponse:
+    def __init__(self, calls: CallsResource) -> None:
+        self._calls = calls
+
+        self.dial = to_streamed_response_wrapper(
+            calls.dial,
+        )
+        self.retrieve_status = to_streamed_response_wrapper(
+            calls.retrieve_status,
+        )
+
+    @cached_property
+    def actions(self) -> ActionsResourceWithStreamingResponse:
+        return ActionsResourceWithStreamingResponse(self._calls.actions)
+
+
+class AsyncCallsResourceWithStreamingResponse:
+    def __init__(self, calls: AsyncCallsResource) -> None:
+        self._calls = calls
+
+        self.dial = async_to_streamed_response_wrapper(
+            calls.dial,
+        )
+        self.retrieve_status = async_to_streamed_response_wrapper(
+            calls.retrieve_status,
+        )
+
+    @cached_property
+    def actions(self) -> AsyncActionsResourceWithStreamingResponse:
+        return AsyncActionsResourceWithStreamingResponse(self._calls.actions)