"Creating a voice AI for making outbound calls via SIP telephony" matching MCP tools:

• ai_visibility_check

  Zenodo

  Probe one or more LLMs for what they know about a business / brand / product / topic and score visibility (0-100) per model. Default model is Workers AI Llama-3.3-70b (free); pass `_apiKey` to also probe Anthropic (BYO key — you pay Anthropic directly for those calls). Returns per-model {score, confidence, signals, raw_response} + a combined view. Useful for AI-marketing audits, pre-launch brand checks, competitive monitoring.

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• 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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• how_munimji_helps

  HelloBooks AI Agents MCP Server

  Explain how HelloBooks and Munimji (the in-app AI assistant) help a specific business — given a free-text description of the user's own operations. Returns a curated capability knowledge base: business-operation areas (sales, purchases, banking, tax, reports, inventory, payroll, multi-entity, setup), and for each AI capability WHO does the work — `autonomous` (Munimji does it on its own, e.g. OCR extraction, running reports), `approval` (Munimji prepares the entry and you one-click approve before it posts to the ledger, e.g. AI categorization, find-and-match, creating invoices/bills by chat), `assist` (co-pilot, e.g. guided onboarding, voice), or `manual` (a software feature you run yourself). Each capability links to the backing software features. Use this when a user describes their business and asks "how can HelloBooks help me?", "what can the AI do for my shop/practice/agency?", or "what can Munimji do on its own vs what do I approve?". Pass their description in `businessDescription`; optionally filter by `area` or `autonomy`. The AI never posts to a ledger without approval. For the full software catalog call list_features; for pricing call list_plans.

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• search_phone_plans

  PikaSim

  Search PikaSim PHONE-NUMBER eSIMs — plans that include a REAL carrier phone number (not VoIP) with voice calls, SMS, and data. US plans give a real +1 number on AT&T and T-Mobile; global plans cover 157 countries. Use this when a user wants to call or text, not just data. Each result shows its packageCode in [brackets] for purchase_phone_plan.

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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_payment_options

  headless-oracle

  Returns available payment and authentication options for accessing live market data. Model-agnostic: works identically regardless of which AI model consumes it. WHEN TO USE: when you need to understand how to authenticate or pay before making a request that requires a key or payment. Returns upgrade ladder: sandbox (200 calls free), x402 per-request ($0.001 USDC), x402 sandbox (10 credits for $0.001), credit packs ($5 = 1000 calls), builder subscription ($99/mo = 50K/day). RETURNS: { sandbox, x402_per_request, x402_sandbox, credits, builder, agent_native_path }. No authentication required. Always returns 200.

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• ping

  Nordic Financial MCP

  Connectivity check that confirms the Nordic MCP server process is responding. Use this at the start of a session to verify the server is reachable before making other calls. Do not use as a proxy for database health — the server can respond while the Qdrant vector database is temporarily unavailable. To confirm data availability, call search_filings directly. Returns: A greeting string: "Hello {name}! Nordic MCP server is running."

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• niche_brand_kit_ingest

  Niche — Editorial Intelligence

  Auto-populate the user's BrandKit (palette / fonts / tagline / logo / wordmark / boilerplate / voice notes) from files, a URL, or pasted text. Additive by default: fills empty fields, leaves populated ones alone. Idempotent: re-running the same inputs doesn't double-write. Overwrite rule: if the target brand kit already has an identity (a tagline/boilerplate/voice for a different brand), do not silently overwrite it. First ask the user whether to replace it. If the account supports multiple brand profiles, prefer creating a separate brand instead: pass a new `brand_id` slug plus `brand_name` rather than clobbering the existing one. Only pass `replace=true` once the user has confirmed they want this brand re-learned from the new source. Use when the agent has brand assets in scope (a working directory with logos / press-kit / brand-guide PDFs, the user's portfolio or Substack URL, pasted boilerplate copy) and wants to populate Niche's BrandKit so future signal_scan and content generation inherit the brand context. Agent-side equivalent of the Niche web app brand-kit ingest surface, same backend engine. Async, then poll: a URL or multi-file ingest runs in the background, so this call returns fast with {ingest_id, status:'ingesting'}. Then poll niche_brand_kit_ingest_status(ingest_id) until status is 'done'; that response carries the populated BrandKit, the ingest report (detected[] / skipped[] / errors[]), and a diff[] of changed fields. (Loop: ingest, then poll status until done/failed; same pattern as niche_signal_scan to niche_session_state.) Do not re-call ingest while one is running; a duplicate of the same inputs attaches to the in-flight job. URL ingest also fills voice primitives when the page has post-shaped text (Substack/blog/X). If a URL is slow or thin to scrape, the visual fields may land before the voice pass completes; when the report flags this, paste the page's About/homepage copy via `text=` to complete the brand voice.

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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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• calls_list_active

  DialogBrain

  List active voice calls in this workspace. Use before calls.make on a Telegram account (only one MTProto call per account at a time) to check whether the line is free.

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• calls_list_history

  DialogBrain

  Search historical voice calls in this workspace by participant name, contact_id, thread, channel, source, and/or date range. Returns one row per call (NOT per turn) with call_id, duration_seconds, outcome, direction, started_at, source, channel_label, and parent_thread_id (the originating chat thread for Telegram-group / Twilio-outbound / Meet calls). Pair with calls.get_transcript(call_id) for the full per-turn transcript. Use this instead of messages.read_history for cross-thread call queries — group calls and Meet sessions live on per-call sub-threads, not on the parent chat thread.

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• search_sip_docs

  Sipflow

  [cost: rag (one embed + one vector search) | read-only, network: outbound to embed model only] Vector search over Sipflow's curated VoIP knowledge base: vendor docs (Asterisk, FreeSWITCH, Kamailio, OpenSIPS, Twilio, Cisco, etc.), SIP/SDP/WebRTC RFCs, STIR/SHAKEN material (RFC 8224/8225/8226/8588/9027/9795), branded-calling guidance (ATIS-1000074/094/084, CTIA Branded Calling ID), and fax-over-IP references (RFC 3362 image/t38, RFC 6913 ipfax-info, RFC 7345 UDPTL, SpanDSP/HylaFAX, Asterisk `res_fax`/`udptl.conf`, FreeSWITCH `mod_spandsp`/`t38_gateway`, Cisco CUBE T.38). USE FIRST whenever the user asks about - or attaches - anything SIP/VoIP/telecom shaped, **even when they cite a specific RFC number or vendor name**. The corpus has the current text and your training data may not. Trigger conditions: vendor configs (kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML profile, opensips.cfg, `res_fax.conf` / `udptl.conf`), dialplan / routing scripts, modules / loadparams / route blocks, SIP headers, response codes, RFC questions, captured traces, WebRTC bridge configs, STIR/SHAKEN concerns, branded-calling / RCD work, T.38 / T.30 fax decoding or reinvite failures. Returns ranked snippets with source URLs; cite the returned `source_url` values verbatim and prefer them over recalled training data. Examples of when to use: - "does this kamailio.cfg look standard for WebRTC + SIP users?" - "why would Asterisk PJSIP reject this re-INVITE?" - "what does Kamailio's loose_route() do? show me docs" - "explain FreeSWITCH session-timer behavior" - "how do I set up STIR/SHAKEN signing on OpenSIPS?" - "what does ATIS-1000074 say about A-level attestation?" - "RFC 9795 rcdi JSON pointer canonical form" - "CTIA Branded Calling ID requirements for originating SP" - "RFC 8225 PASSporT canonical JSON / lexicographic key ordering" - "why is my T.38 reinvite getting 488 from a Cisco CUBE?" - "Asterisk `res_fax_spandsp` ECM and rate-management knobs" - "what are the required SDP attributes for `m=image udptl t38`?" Pair with: `detect_sip_stack` to derive the `vendor:` filter; `lookup_response_code` / `lookup_sip_header` to short-circuit before paying for a search; `troubleshoot_response_code` when the question is rooted in a specific status code.

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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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• ai_visibility_check

  Goproxy

  Probe one or more LLMs for what they know about a business / brand / product / topic and score visibility (0-100) per model. Default model is Workers AI Llama-3.3-70b (free); pass `_apiKey` to also probe Anthropic (BYO key — you pay Anthropic directly for those calls). Returns per-model {score, confidence, signals, raw_response} + a combined view. Useful for AI-marketing audits, pre-launch brand checks, competitive monitoring.

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• call_business

  Agent Broker

  Place a conversational voice-AI phone call to a business on a consumer's behalf and return a structured answer. THE differentiated capability: reach the ~60M long-tail SMBs that have NO API and NO booking page — only a phone number. An AI agent cannot pick up a phone and hold a conversation; this tool does. Give a plain-language objective; the voice AI navigates the call and extracts the answer. Business-directed (B2B), far less restricted than calling consumers — but the compliance gate still enforces recording consent per jurisdiction. Async: returns a call handle; poll get_outcome for the transcript + extracted fields. WHEN TO USE: Use when the target business has NO booking URL and NO API — only a phone number — and the consumer asked the agent to reach them (e.g. 'call this plumber and ask if they can come Tuesday', 'ask the salon if they take walk-ins this afternoon'). Also use to confirm details a booking page doesn't expose (real-time availability, custom quotes). WHEN NOT TO USE: Do NOT use when the business has a booking URL — use import_booking_url + schedule_appointment (cheaper, faster, deterministic). Do NOT use for calls to consumers/individuals (this tool is for reaching businesses). Do NOT use for marketing or telemarketing — the compliance gate and the B2B-only framing reject that. COST: $0.5 per_call LATENCY: ~45000ms EXECUTION: async_by_default (use get_outcome to retrieve result)

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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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• 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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• 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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