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* ViciDial Working * feat(telephony): add FreeSWITCH provider to upstream-PBX seam Generalize the upstream-PBX capture and control paths to dispatch by provider. FreeSWITCH bridges in with X-PBX-* headers and is driven over the Event Socket Library (uuid_kill / uuid_transfer) by the channel UUID, alongside the existing VICIdial ra_call_control path. Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com> * add vici specific configs * feat(telephony): env-based VICIdial config + address3 post-call routing Move VICIdial agent/non-agent API and FreeSWITCH ESL connection settings out of hardcoded POC values into environment variables so the same image works against a PBX on another server. Add VICIdial update_lead forwarding: X-VICI-UPDATE-LEAD_* extracted variables are mapped to lead columns and pushed via the non-agent API before a transfer, with reserved API-control params dropped. Add hardcoded address3 disposition routing (Y/N -> in-group transfer, else hang up) and a synchronous final variable extraction so the transfer path sees the freshest conversation state. Skip upstream_pbx capture on non-PJSIP channels to avoid Asterisk 500s. Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com> * chore: run formatter * feat: make vici configurable from UI * chore: clean up PR * fix: incorporate review comments * chore: incorporate review comments * chore: generate client * chore: incporporate review comments * chore: incorporate review comments --------- Co-authored-by: Dograh POC <payment@dograh.com> Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
75 lines
2.5 KiB
Python
75 lines
2.5 KiB
Python
"""ARI (Asterisk) transport factory."""
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from fastapi import WebSocket
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from pipecat.transports.websocket.fastapi import (
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FastAPIWebsocketParams,
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FastAPIWebsocketTransport,
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)
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from api.services.pipecat.audio_config import AudioConfig
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from api.services.pipecat.audio_mixer import build_audio_out_mixer
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from api.services.pipecat.transport_params import realtime_param_overrides
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from api.services.telephony.factory import load_credentials_for_transport
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from .external_pbx import create_adapter
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from .serializers import AsteriskFrameSerializer
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from .strategies import ARIBridgeSwapStrategy, ARIHangupStrategy
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async def create_transport(
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websocket: WebSocket,
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workflow_run_id: int,
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audio_config: AudioConfig,
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organization_id: int,
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*,
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ambient_noise_config: dict | None = None,
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telephony_configuration_id: int | None = None,
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is_realtime: bool = False,
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channel_id: str,
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):
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"""Create a transport for Asterisk ARI connections."""
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config = await load_credentials_for_transport(
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organization_id, telephony_configuration_id, expected_provider="ari"
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)
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ari_endpoint = config.get("ari_endpoint")
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app_name = config.get("app_name")
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app_password = config.get("app_password")
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if not ari_endpoint or not app_name or not app_password:
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raise ValueError(
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f"Incomplete ARI configuration for organization {organization_id}. "
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f"Required: ari_endpoint, app_name, app_password"
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)
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serializer = AsteriskFrameSerializer(
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channel_id=channel_id,
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ari_endpoint=ari_endpoint,
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app_name=app_name,
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app_password=app_password,
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transfer_strategy=ARIBridgeSwapStrategy(),
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hangup_strategy=ARIHangupStrategy(
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external_pbx_adapter=create_adapter(config.get("external_pbx"))
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),
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params=AsteriskFrameSerializer.InputParams(
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asterisk_sample_rate=audio_config.transport_in_sample_rate,
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sample_rate=audio_config.pipeline_sample_rate,
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),
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)
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mixer = await build_audio_out_mixer(
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audio_config.transport_out_sample_rate, ambient_noise_config
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)
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return FastAPIWebsocketTransport(
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websocket=websocket,
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params=FastAPIWebsocketParams(
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audio_in_enabled=True,
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audio_out_enabled=True,
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audio_in_sample_rate=audio_config.transport_in_sample_rate,
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audio_out_sample_rate=audio_config.transport_out_sample_rate,
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audio_out_mixer=mixer,
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serializer=serializer,
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**realtime_param_overrides(is_realtime),
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),
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)
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