"Building voice agents and controlling SIP phones" matching MCP tools:

• synthesize_voiceover

  TestMyVibes

  Generates a voiceover from text using Hume Octave TTS. Audio uploaded to Spaces, signed URL returned (24h TTL by default). Charged in credits up-front based on script length (use quote_voiceover for a preview). Best for demo-video narration, tutorial audio, and any one-shot batch TTS. NOT a real-time conversational voice (use Hume EVI for that, different product). Voice options: pass voiceId for a specific Hume voice clone, or omit to use the deployment's default narrator (HUME_OCTAVE_VOICE_ID env var).

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• adcritter_build_app

  AdCritter for Builders

  Start here when building an application. Returns an overview of what the AdCritter platform offers and a catalog of feature guides you can query with the adcritter_guidance tool to learn how to build each part of the app. Call adcritter_guidance(key) for any feature area to get detailed building instructions with API endpoints and response shapes.

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• update

  OnePageCRM

  Update any entity. Only provided fields are changed. Use describe() for writable fields. Array fields (emails, phones, urls, tags) are full replacements — to add/remove a single entry, query() the current array first, modify it, then send the full list.

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• generate_voiceover

  framesail

  Generate TTS audio for the project's voice blocks. Without voice_block_ids it fills gaps: only blocks with no audio yet run, so re-calling it is always safe (already-generated and currently-generating blocks are skipped, never re-billed). Pass voice_block_ids to explicitly REgenerate those blocks (e.g. after changing a block's voice). Speakers must have voices bound first — set_narrator_voice / set_character_voice. Optional editable_sections/settings apply to every selected block (see get_section_template("voice_block") and list_models("voice_block")). Async — returns one job per block.

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• createSpeech

  Ludo AI Game Assets

  Convert text to speech by cloning the voice from an audio sample you provide (voice-cloning text-to-speech). Both text and sample are required; the text is limited to 1000 characters and the sample is supplied as a URL or base64 audio that must be at most 15MB, with violations returning HTTP 400. Synchronous: the call blocks until generation finishes and returns a single audio result containing a URL; there is no separate polling step. Credits are charged on success. Use this when you have a reference voice sample to clone; use createSpeechPreset to speak with a built-in named preset voice instead, and createVoice to design a brand-new voice from a text description rather than cloning one. Pass an optional request_id to tag the result so you can locate it later via getAudioResults. Requires an API key (user scope). Credits: This endpoint consumes 1 credits per call.

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• get_building_breakdown

  Cenogram - Polish Real Estate Data

  Get the building-by-building breakdown for one transaction: footprint area, number of storeys, and estimated total floor area (footprint × storeys) for each building on the property. search_transactions / search_by_area / search_by_polygon return per-transaction building SUMS inline; this tool splits them into individual buildings. Use it after a search when a result has building data and you need the detail (e.g. a developed-land deed covering several buildings). The transaction_id is the id shown on a search result that has building data. Cost: 1 token. Returns nothing for a transaction with no buildings.

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• api_info

  x402 Crypto Market Structure

  REST API access for autonomous agents — pricing, quick start, and migration guide. Call this when: building a trading bot, deploying an autonomous agent, hitting the MCP rate limit, or running 24/7 without a human in the loop. The MCP tier (what you're using now) is free via Smithery, rate-limited to 60 calls/minute per IP, and good for testing. The REST API is for production: pay per call in USDC; paid endpoints are rate-limited to 60 calls/minute and 200 calls/hour per wallet. No API key required.

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• epub_to_audiobook

  Sats4AI - Bitcoin-Powered AI Tools

  Convert books (EPUB/PDF/TXT) to full audiobooks with automatic chapter detection, multi-voice narration, and optional translation to any language before narration. 3 voice tiers: OmniVoice Global (602+ langs, 100 chars/sat), Inworld Premium (#1 ranked TTS ELO 1217, 50 chars/sat), Minimax Studio (voice cloning from reference clip, 10 chars/sat). Min 500 sats. Async — returns jobId, poll until completed (5-60+ min). Single payment, full outcome — no multi-step orchestration required. Pay with Bitcoin Lightning — no API key or signup needed. Requires create_payment with toolName='epub_to_audiobook'.

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• createSpeechPreset

  Ludo AI Game Assets

  Convert text to speech using a named built-in preset voice, with optional emotion and language settings. Both text and voice_preset_id are required and the text is limited to 1000 characters; invalid input returns HTTP 400. Synchronous: the call blocks until generation finishes and returns a single audio result containing a URL; there is no separate polling step. Credits are charged on success. Use this when you want a ready-made catalog voice and do not need to supply your own sample; use createSpeech to clone a voice from an audio sample instead, and createVoice to design a new voice from a text description. Pass an optional request_id to tag the result so you can locate it later via getAudioResults. Requires an API key (user scope). Credits: This endpoint consumes 1 credits per call.

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• detect_sip_stack

  Sipflow

  [cost: free (pure CPU, no network) | read-only] Identify the SIP product behind a piece of input. Works on both: - a SIP trace (User-Agent / Server headers from PCAP/sngrep/syslog), and - a vendor config blob (kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML, opensips.cfg) detected via structural signatures (loadmodule, route blocks, [transport-*] sections, <profile name=>, etc.). Returns a vendor slug (e.g. "kamailio", "freeswitch", "asterisk", "twilio", "cisco-cube") aligned with the `vendor` filter on `search_sip_docs`, so you can pipe the output of this tool directly into a follow-up doc search. Pair with: `search_sip_docs(vendor=<slug>, ...)` for grounded vendor docs; `review_sip_config` when the input is a config and you also want extracted modules + risk flags; `troubleshoot_response_code(vendorHint=<slug>, ...)` when chasing a status code.

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• dns_diagnose_sip_target

  Sipflow

  [cost: external_io (DNS via Cloudflare + Google; TLS handshake + a SIP OPTIONS keepalive to public targets when applicable) | read-only | rate-limited per IP: 10/min, 200/day] Walk DNS the same way a SIP UA does (RFC 3263 §4.1): NAPTR → SRV → A/AAAA. Given a SIP URI ("sip:example.com"), bare hostname ("example.com"), or "host:port" string, return the records that exist and the resolution ladder a UA would try. When the queried target uses TLS (`sips:` URI, `transport=tls/wss`, or any `_sips._tcp` SRV record), the tool also performs a TLS handshake against each resolved sips target and reports the negotiated TLS version + cipher, the leaf certificate's subject / issuer / SANs / validity, the chain length and whether it validates against Node's default trust store, plus two cert-domain checks: RFC 5922 §7.2 strict (cert must cover the original SIP domain) and a lenient SAN match against the SRV target hostname. SIP liveness: DNS resolving and a TLS handshake succeeding do NOT prove the endpoint actually speaks SIP - a load-balanced node can accept TCP/TLS yet black-hole SIP. So the tool ALSO sends a real SIP OPTIONS keepalive to each resolved public IP across the relevant transports (UDP/TCP on 5060, TLS on 5061 / SRV port) and reports per-IP answered / timeout / refused. Any SIP response (even 405/403/404) proves the stack is alive on that IP. When a name resolves to multiple IPs it is treated as a load-balancer fan-out and each IP is probed individually, with a warning about the known failure modes of fronting stateful SIP/RTP with a cloud L4 LB (AWS NLB/ALB etc.): cross-zone-off targets that black-hole, the ~120s UDP idle timeout, and per-5-tuple hashing splitting signaling from media. Egress safety: - Per-IP rate limited. - Hostnames that resolve only to RFC 1918 / loopback / link-local / documentation / multicast space are refused (SSRF guard). - Walk depth capped to prevent runaway NAPTR / CNAME chains. - TLS probes capped at 6 (host, port, ip) tuples per call, 5 s handshake timeout each, public-IP only (we connect to the resolved IP, not the hostname, so the system resolver cannot redirect us into private space). - SIP OPTIONS probes capped at 6 (ip, transport) tuples per call, 3 s timeout each, public-IP only; the request carries no SDP/body and an unroutable Via, and only the response status line is captured. Use to diagnose: - "carrier doesn't answer" / "wrong port" / "TLS instead of UDP" routing puzzles - "DNS looks healthy but calls fail" - per-IP SIP OPTIONS surfaces nodes that resolve and accept the transport but never answer SIP (the decisive step for load-balanced / multi-IP targets) - "carrier rejects our target because no SRV is published" - when A/AAAA resolves but SRV is missing the tool synthesises a copy-pasteable suggested zone-record block pointing at the resolved canonical hostname - "TLS handshake works but cert isn't valid for the SIP domain" - RFC 5922 §7.2 compliance is checked separately from generic chain validation, since the SAN must cover the *original* SIP domain (not the SRV-redirected target) ACL caveat: a SIP OPTIONS timeout can also mean the target authorizes inbound SIP by source IP whitelist on the trunk (Twilio, Telnyx, Bandwidth, …; see https://www.twilio.com/docs/sip-trunking/api/ipaccesscontrollist-resource) and is dropping our probe because our egress IP is not on the ACL. An `answered` result is conclusive (the node speaks SIP); a `timeout` is suggestive, not proof of a dead node - confirm reachability from the SBC itself. Pair with: `troubleshoot_response_code` when 503 / 408 / 480 are involved; `search_sip_docs(vendor=...)` for carrier-specific routing docs.

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• slovenia-cadastral-deep

  geoagentic

  Deep parcel and building analysis for Slovenia using GURS WFS data. Returns zoning, actual use, heritage protection, road access, buildings on parcel, and utilities. USE FOR: - "Analyze parcel 3086 in Ljubljana center" - "Find buildable parcels ~500m² in Ljubljana" - "What buildings are on this parcel?" - "Find parcels near these coordinates" - "Get full details on building 1234" NOT FOR: simple parcel lookup → use slovenia-cadastre instead (faster, lighter). NOT FOR: spatial/zoning map queries → use slovenia-wfs-expert instead. SEARCH MODES — pick ONE per call: 1. PARCEL BY NUMBER (requires --parcel AND --ko) → --parcel 3086 --ko 1725 2. LOCATION SEARCH (requires --lat AND --lon, or --location) → --lat 46.058 --lon 14.501 --radius 100 → --location "Tivoli Park Ljubljana" --radius 200 3. BUILDING BY NUMBER (requires --building, optionally --ko) → --building 1234 --ko 1728 4. COMMUNITY SEARCH (requires at least --community or --size) → --community LJUBLJANA --size 500 --buildable COMMON KO IDs: 1725 = Ljubljana center 1728 = Ljubljana Šiška 1740 = Ljubljana Bežigrad 2131 = Maribor NOTE: This tool makes multiple WFS calls per result and can be slow (10-30s). Use --limit to keep response times reasonable.

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• voice

  switch

  Your saved voices — one tool for the whole voice library. Users speak plain language and never know ids: resolve every voice by NAME yourself (call action "list" first if unsure) and never ask the user for an id. action="list" returns every saved voice with voice_id, name, kind and ready — kind "reference" is an instant voice match saved from a clip and kind "clone" is a trained voice (both speak through generate_audio: pass the NAME as its voice param); kind "avatar" voices drive talking_avatar_video. action="create" saves a NEW reference voice from a clip: voice_name plus audio_url (e.g. the url upload_media returned) or audio_base64 (+ format) — free, ready instantly. action="rename" renames a saved voice (voice_id takes the id OR the current name, new_name is the new name). action="clone" registers a voice for talking_avatar_video from audio_sample_url + voice_name (charged 2 credits). action="delete" removes a voice by voice_id or name.

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• niche_voice_profile_ingest

  Niche — Editorial Intelligence

  Extract voice primitives (register / sentence rhythm / lexicon preferences / punctuation habits) from post-shaped text and persist onto the user's VoiceProfile. The voice primitives thread into content generation so generated copy matches the user's actual writing voice. Two input shapes: pass `posts` (list of pre-collected text snippets, ≥80 chars each) or pass `url` (the server scrapes post-shaped snippets from the page: Substack / Medium / blog / X profile). Inline posts win when both are given. Inline post-shaped snippets need to be the user's own writing, not press articles or marketing copy. Returns the extracted primitives + a diff of what changed on the stored VoiceProfile.

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• generate_audio

  switch

  Generate spoken audio from text: narration, a voiceover, a read-aloud script, or a multi-voice dialogue. Pass text (up to 2048 chars) — the words to be spoken. To speak in one of YOUR saved voices, pass voice with the voice NAME (or id): users speak plain language and never know ids, so resolve the name yourself (the voice tool, action "list", shows every saved voice) and never ask the user for an id. Reference voices, trained clones and preset voices are all routed correctly by kind. To match a voice instantly from a clip instead, pass reference_audio_url (a short clip) or up to 3 reference_audio_urls and address them as @Audio1, @Audio2, @Audio3 in the text for dialogue. Alternatively pass image_url to voice a scene from a picture (cannot combine with reference audio). Optional speech_rate (-50..100), pitch (-12..12), loudness (-50..100). Returns a playable audio_url, duration_seconds, and generation_id (also saved to your library).

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• createVoice

  Ludo AI Game Assets

  Design a new voice from a character description (such as "deep-voiced warrior" or "cheerful young girl") and have it speak a short line of text, returning a sample of that newly created voice. Both voice_description and text are required, the spoken text is limited to 200 characters or the call returns HTTP 400, and type selects "human" or "non-human" voices. Synchronous: the call blocks until generation finishes and returns a single audio result containing a URL; there is no separate polling step. Credits are charged on success. Use this to invent and audition a voice from a description; use createSpeech for text-to-speech that clones a specific voice from an audio sample, and createSpeechPreset for text-to-speech using a named preset voice. Pass an optional request_id to tag the result so you can locate it later via getAudioResults. Requires an API key (user scope). Credits: This endpoint consumes 1 credits per call.

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• lookup_response_code

  Sipflow

  [cost: free (pure CPU, no network) | read-only] Instant static lookup of a SIP response code (100-699). Returns name, RFC anchor, category, description, common operator-flavored causes, and known vendor-specific reason-phrase variants (e.g. OpenSIPS emits 484 'Invalid FROM' on From-header parse failure). USE FIRST when the user pastes or asks about any 3-digit SIP code - sub-millisecond, no API cost. Pair with: `troubleshoot_response_code` for vendor-specific RAG hits beyond the static entry; `lint_sip_request` when the code is 4xx and the user has the offending request; `stir_attestation_explainer` for STIR-shaped codes (428/436/437/438/608); `validate_stir_shaken_identity` when the code is 438 and they have the JWS.

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• close_list_contacts

  Close Crm

  List contacts (people) in Close. Returns a `data` array of contacts with id, lead_id, name, title, emails, and phones, plus `has_more` / `total_results`. Optionally filter to one lead with `lead_id`. Page with `_limit` / `_skip`.

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• calls_agent_duel

  dialogbrain

  Start an agent-vs-agent VOICE test call: two AI voice agents share one LiveKit room — a 'caller' persona agent pursues a task brief against the 'callee' business agent under test. Use to evaluate booking flows, latency, and conversation quality without a human caller. The CALLER agent should have an EMPTY voice_greeting (it must stay silent until the callee greets) and voice_filler_enabled=false. Afterwards inspect both call_ids with agents.traces_list / calls.get_transcript. A subscribe-only `listen` token is returned for listening in live.

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• text_to_speech

  Sats4AI - Bitcoin-Powered AI Tools

  Text-to-speech with 3 tiers: OmniVoice Global (602+ languages including Yoruba, Bengali, Cebuano, Twi, zero-shot voice cloning, 100 chars/sat — use 'language' parameter with ISO code), Inworld Premium (#1 ranked TTS ELO 1217, emotion control, 40+ languages, 50 chars/sat), Minimax Studio (voice cloning from reference clip, 40+ languages, 10 chars/sat). Adjustable speed (0.5-2.0x). Returns audio URL. Pay with Bitcoin Lightning — no API key or signup needed. When NOT to use: not for phone calls (use place_call for one-shot broadcasts, ai_call for AI voice agents, or open_voice_bridge to drive the call with your own LLM). For rare/underserved languages (Yoruba, Twi, Marathi, Cebuano, etc.), pick OmniVoice Global via language= — Inworld/Minimax don't cover these. Requires create_payment with toolName='text_to_speech'.

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