Restore local STT command fallback for voice transcription, detect whisper and ffmpeg in common local install paths, and avoid bogus no-provider messaging when only a backend-specific key is missing.
- Add 'emoji' field to ToolEntry and 'get_emoji()' to ToolRegistry
- Add emoji= to all 50+ registry.register() calls across tool files
- Add get_tool_emoji() helper in agent/display.py with 3-tier resolution:
skin override → registry default → hardcoded fallback
- Replace hardcoded emoji maps in run_agent.py, delegate_tool.py, and
gateway/run.py with centralized get_tool_emoji() calls
- Add 'tool_emojis' field to SkinConfig so skins can override per-tool
emojis (e.g. ares skin could use swords instead of wrenches)
- Add 11 tests (5 registry emoji, 6 display/skin integration)
- Update AGENTS.md skin docs table
Based on the approach from PR #1061 by ForgingAlex (emoji centralization
in registry). This salvage fixes several issues from the original:
- Does NOT split the cronjob tool (which would crash on missing schemas)
- Does NOT change image_generate toolset/requires_env/is_async
- Does NOT delete existing tests
- Completes the centralization (gateway/run.py was missed)
- Hooks into the skin system for full customizability
SSH persistent shell now defaults to true — non-local backends benefit
most from state persistence across execute() calls. Local backend
remains opt-in via TERMINAL_LOCAL_PERSISTENT env var.
New config.yaml option: terminal.persistent_shell (default: true)
Controls the default for non-local backends. Users can disable with:
hermes config set terminal.persistent_shell false
Precedence: per-backend env var > TERMINAL_PERSISTENT_SHELL > default.
Wired through cli.py, gateway/run.py, and hermes_cli/config.py so the
config.yaml value reaches terminal_tool via env var bridge.
When a cronjob is created from within a Telegram or Slack thread,
deliver=origin was posting to the parent channel instead of the thread.
Root cause: the gateway never set HERMES_SESSION_THREAD_ID in the
session environment, so cronjob_tools.py could not capture thread_id
into the job's origin metadata — even though the scheduler already
reads origin.get('thread_id').
Fix:
- gateway/run.py: set HERMES_SESSION_THREAD_ID when thread_id is
present on the session context, and clear it in _clear_session_env
- tools/cronjob_tools.py: read HERMES_SESSION_THREAD_ID into origin
Closes#1219
Hermes startup entrypoints now load ~/.hermes/.env and project fallback env files with user config taking precedence over stale shell-exported values. This makes model/provider/base URL changes in .env actually take effect after restarting Hermes. Adds a shared env loader plus regression coverage, and reproduces the original bug case where OPENAI_BASE_URL and HERMES_INFERENCE_PROVIDER remained stuck on old shell values before import.
Add an explicit messaging-extra install hint to the missing PyNaCl/davey error path, cover it with a voice-channel join regression test, and skip the low-level NaCl packet tests when PyNaCl is not installed locally.
When PyNaCl or davey is not installed, joining a voice channel fails
with a raw exception. Now shows a human-readable message pointing
the user to reinstall with voice support.
Closes#1336
play_tts was returning success without playing anything when bot was
in a voice channel. Now it calls play_in_voice_channel directly.
Simplified skip_double dedup: base adapter handles voice input TTS
via play_tts (which now works for VC), runner skips to avoid double.
Track adapter background message-processing tasks, cancel them during gateway shutdown, and interrupt running agents before disconnecting adapters. This prevents old gateway instances from continuing in-flight work after stop/replace, which was contributing to the restart-time task continuation/flicker behavior reported in #1414. Adds regression coverage for adapter task cancellation and shutdown interrupts.
- bridge stt.enabled from config.yaml into gateway runtime config
- preserve the flag in GatewayConfig serialization
- skip gateway voice transcription when STT is disabled
- add regression tests for config loading and disabled transcription flow
Add base_url/api_key overrides for auxiliary tasks and delegation so users can
route those flows straight to a custom OpenAI-compatible endpoint without
having to rely on provider=main or named custom providers.
Also clear gateway session env vars in test isolation so the full suite stays
deterministic when run from a messaging-backed agent session.
Follow up on salvaged PR #1052.
Restore current-main gateway lifecycle handling after conflict resolution and
adapt the update fallback to use shell-quoted argv parts safely.
When shutil.which('hermes') returns None, _resolve_hermes_bin() now tries
sys.executable -m hermes_cli.main as a fallback. This handles setups where
Hermes is launched via a venv or module invocation and the hermes symlink is
not on PATH for the gateway process.
Fixes#1049
Use getattr() when returning model metadata from GatewayRunner._run_agent so fake agents and minimal stubs without a model attribute do not break unrelated gateway flows while preserving the session-model backfill behavior.
Gateway sessions end up with model=NULL because the session row is
created before AIAgent is constructed. After the agent responds,
update_session() writes token counts but never fills in the model.
Thread agent.model through _run_agent()'s return dict into
update_session() → update_token_counts(). The SQL uses
COALESCE(model, ?) so it only fills NULL rows — never overwrites
a model already set at creation time (e.g. CLI sessions).
If the agent falls back to a different provider, agent.model is
updated in-place by _try_activate_fallback(), so the recorded value
reflects whichever model actually produced the response.
Fixes#987
- keep CLI voice prefixes API-local while storing the original user text
- persist explicit gateway off state and restore adapter auto-TTS suppression on restart
- add regression coverage for both behaviors
1. Gate _streaming_api_call to chat_completions mode only — Anthropic and
Codex fall back to _interruptible_api_call. Preserve Anthropic base_url
across all client rebuild paths (interrupt, fallback, 401 refresh).
2. Discord VC synthetic events now use chat_type="channel" instead of
defaulting to "dm" — prevents session bleed into DM context.
Authorization runs before echoing transcript. Sanitize @everyone/@here
in voice transcripts.
3. CLI voice prefix ("[Voice input...]") is now API-call-local only —
stripped from returned history so it never persists to session DB or
resumed sessions.
4. /voice off now disables base adapter auto-TTS via _auto_tts_disabled_chats
set — voice input no longer triggers TTS when voice mode is off.
Remove web UI gateway (web.py, tests, docs, toolset, env vars, Platform.WEB
enum) per maintainer request — Nous is building their own official chat UI.
Fix 1: Replace sd.wait() with polling pattern in play_audio_file() to prevent
indefinite hang when audio device stalls (consistent with play_beep()).
Fix 2: Use importlib.util.find_spec() for faster_whisper/openai availability
checks instead of module-level imports that trigger heavy native library
loading (CUDA/cuDNN) at import time.
Fix 3: Remove inspect.signature() hack in _send_voice_reply() — add **kwargs
to Telegram send_voice() so all adapters accept metadata uniformly.
Fix 4: Make session loading resilient to removed platform enum values — skip
entries with unknown platforms instead of crashing the entire gateway.
- web.py: pass stt_model from config like discord.py and run.py do
- run.py: match new error messages (No STT provider / not set)
- _transcribe_local: add missing "provider": "local" to return dict
- Use hmac.compare_digest for timing-safe token comparison (3 endpoints)
- Default bind to 127.0.0.1 instead of 0.0.0.0
- Sanitize upload filenames with Path.name to prevent path traversal
- Add DOMPurify to sanitize marked.parse() output against XSS
- Replace add_static with authenticated media handler
- Hide token in group chats for /remote-control command
- Use ctypes.util.find_library for Opus instead of hardcoded paths
- Add force=True to 5 interrupt _vprint calls for visibility
- Log Opus decode errors and voice restart failures instead of swallowing
Duplicated YAML config parsing for stt.model existed in gateway/run.py
and gateway/platforms/discord.py. Moved to a single helper in
transcription_tools.py and added 5 tests covering all edge cases.
Code fixes:
- STT model, Groq base URL, and OpenAI STT base URL are now
configurable via env vars (STT_GROQ_MODEL, STT_OPENAI_MODEL,
GROQ_BASE_URL, STT_OPENAI_BASE_URL) instead of hardcoded
- Gateway and Discord VC now read stt.model from config.yaml
(previously only CLI did this — gateway always used defaults)
Doc fixes:
- voice-mode.md: move Web UI troubleshooting to web.md (was duplicated)
- voice-mode.md: simplify "How It Works" for end users (remove NaCl,
DAVE, RTP internals)
- voice-mode.md: clarify STT priority (OpenAI used first if both keys
set, Groq recommended for free tier)
- voice-mode.md: document new STT env overrides in config reference
- web.md: remove duplicate Quick Start / Step 1-3 sections
- web.md: add mobile HTTPS mic workarounds (moved from voice-mode.md)
- web.md: clarify STT fallback order
1. VoiceReceiver.stop() now acquires _lock before clearing shared state
to prevent race with _on_packet on the socket reader thread
2. _packet_debug_count moved from class-level to instance-level to avoid
cross-instance race condition in multi-guild setups
3. play_in_voice_channel uses asyncio.get_running_loop() instead of
deprecated asyncio.get_event_loop()
4. _send_voice_reply uses uuid for filenames instead of time-based names
that can collide when two replies happen in the same second
5. Voice timeout now notifies runner via _on_voice_disconnect callback
so runner cleans up _voice_mode state (prevents orphaned TTS replies)
6. play_in_voice_channel adds PLAYBACK_TIMEOUT (120s) to prevent
infinite blocking when FFmpeg callback is never called
7. _send_voice_reply moves temp file cleanup to finally block so files
are always cleaned up even when send_voice/play raises
8. Base adapter auto-TTS wraps play_tts in try/finally with os.remove
to clean up generated audio files after playback
18 new tests (120 total voice tests)
- Add lock protection around VoiceReceiver buffer writes in _on_packet
to prevent race condition with check_silence on different threads
- Wire _voice_input_callback BEFORE join_voice_channel to avoid
losing voice input during the join window
- Add try/except around leave_voice_channel to ensure state cleanup
(voice_mode, callback) even if leave raises an exception
- Guard against empty text after markdown stripping in base.py auto-TTS
- Add 11 tests proving each bug and verifying the fix
When bot is in a Discord voice channel, both base auto-TTS and Discord
play_tts override skip audio. The skip_double guard was also blocking
the runner's _send_voice_reply, resulting in zero audio output in VC.
Now skip_double is overridden when the bot is actively connected to a
voice channel, allowing play_in_voice_channel to handle TTS.
Add comprehensive test matrix covering all platform x input x mode
combinations with full decision table documentation.
Base adapter auto-TTS already generates and sends audio for voice
messages in _process_message_background. The gateway runner's
_send_voice_reply was causing double audio on all platforms (not
just Web). Now skip_double applies to any voice input regardless
of platform.
When voice mode is enabled and user sends a voice message on Web UI,
both the base adapter auto-TTS (play_audio) and the gateway voice reply
(send_voice) would fire, causing duplicate audio playback. Skip the
gateway voice reply for Web platform voice input since base adapter
already handles it.
- Auto-TTS: voice messages get spoken response (audio first, then text)
- STT: Groq Whisper fallback when VOICE_TOOLS_OPENAI_KEY not set
- Futuristic UI: glassmorphism, centered container, purple theme, glow effects
- Voice bubble: custom waveform player with seek and progress
- Invisible TTS playback via play_tts() method (no audio file in chat)
- Add hermes-web toolset with full tool access
- Register Platform.WEB in toolset/config maps
- Update docs for voice conversation feature
Type /remote-control from any platform (Telegram, Discord, etc.) to
instantly start the web UI without restarting the gateway.
- Auto-generates access token if not provided
- Shows URL + token in response
- Optional: /remote-control [port] [token]
- Reports status if already running
- Added to /help command list
New platform adapter that serves a full-featured chat interface via HTTP.
Enables access from any device on the network (phone, tablet, desktop).
Features:
- aiohttp server with WebSocket real-time messaging
- Token-based authentication
- Markdown rendering (marked.js) + code highlighting (highlight.js)
- Voice recording via MediaRecorder API + STT transcription
- Image, voice, and document display
- Typing indicator + message editing (streaming support)
- Mobile responsive dark theme
- Auto-reconnect on disconnect
- Media file cleanup (24h TTL)
Config: WEB_UI_ENABLED=true, WEB_UI_PORT=8765, WEB_UI_TOKEN=<token>
No new dependencies — uses aiohttp already in [messaging] extra.
Phase 2 of voice channel support: bot listens to users speaking in VC,
transcribes speech via Groq Whisper, and processes through the agent pipeline.
- Add VoiceReceiver class for RTP packet capture, NaCl/DAVE decryption, Opus decode
- Add silence detection and per-user PCM buffering
- Wire voice input callback from adapter to GatewayRunner
- Fix adapter dict key: use Platform.DISCORD enum instead of string
- Fix guild_id extraction for synthetic voice events via SimpleNamespace raw_message
- Pause/resume receiver during TTS playback to prevent echo
- Send Discord voice messages with flags=8192 and waveform metadata
so they render as native voice bubbles instead of file attachments
- Use .mp3 output path for TTS so edge-tts opus conversion works
correctly (edge always outputs mp3, convert was skipped for .ogg)
- Use actual file_path from TTS result after potential opus conversion
- Register /voice as Discord slash command with mode choices
- Fix _send_voice_reply to handle adapters that don't accept metadata
parameter (Discord) by inspecting the method signature at runtime
- /voice on: reply with voice when user sends voice messages
- /voice tts: reply with voice to all messages
- /voice off: disable, text-only replies
- /voice status: show current mode
- Per-chat state persisted to gateway_voice_mode.json
- Dedup: skips auto-reply if agent already called text_to_speech tool
- drop_pending_updates=True to ignore stale Telegram messages on restart
- 25 tests covering command handler, reply logic, and edge cases
