List all iTerm2 windows, tabs, and sessions, returning a tree with session IDs. Highlights the focused window, tab, and session with an asterisk.
MIT
136,059 tools. Last updated 2026-05-26 00:34
"Asterisk" matching MCP tools:
- Retrieve system metrics and statistics for automated phone call operations within the Asterisk S2S MCP Server to monitor and analyze telephony performance.MIT
- Retrieve a real-time list of active phone calls on the Asterisk S2S MCP Server for monitoring or managing ongoing conversations efficiently.MIT
- Assess and monitor the health status of your telephone system with this diagnostic tool, ensuring reliable performance for automated conversational phone calls using Asterisk and Speech-to-Speech capabilities.MIT
- Initiate automated phone calls with conversational capabilities using the Asterisk S2S MCP Server. Specify user, phone number, purpose, and additional context to streamline communications.MIT
- Retrieve system logs from the Asterisk S2S MCP Server for debugging. Filter logs by level (info, warn, error, debug) and component (mcp, phone, callback, client) to diagnose telephony system issues efficiently.MIT
- Retrieve the last processed result of a specific call using the call ID. Ideal for automated conversational phone systems powered by Asterisk S2S MCP Server.MIT
- Terminate an ongoing phone call by specifying the call ID, enabling users to stop conversations efficiently within the Asterisk S2S MCP Server environment.MIT
- Retrieve recent call conversation history from the Asterisk S2S MCP Server, specifying the number of conversations to fetch for efficient review and analysis.MIT
- Retrieve the current status of a phone call by providing the call ID using Asterisk S2S MCP Server's functionality for automated conversational calls.MIT
- [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.Connector
- [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.Connector
- [cost: free (pure CPU, no network) | read-only] Use this when the user asks 'review my config' or attaches a kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML profile, opensips.cfg, `res_fax.conf` / `udptl.conf` / `spandsp.conf` (fax-relay tuning), or a SIP-shaped source file from a repo. This tool: 1. Detects the vendor from filename + structural signatures (loadmodule, route blocks, [transport-*] sections, <profile name=>, KEMI calls). 2. Extracts a structured outline: loaded modules, modparams, listen lines, route blocks, profiles, gateways, dialplan extensions. 3. Surfaces risk flags - e.g. websocket loaded without TLS, nathelper without rtpengine, chan_sip used in modern Asterisk, AND the Kamailio/OpenSIPS lump-vs-subst race (`subst('/^From:.../...')` colliding with `KSR.hdr.append/remove` or `uac_replace_*` or `append_hf/remove_hf` on the same header - corrupts the buffer at serialization). 4. Returns a list of `suggestedQueries` for `search_sip_docs` so you can ground the actual review in vendor docs. Pair with: one or more `search_sip_docs` calls (cite returned `source_url` values verbatim instead of recalling vendor behavior from memory); `webrtc_sip_checklist` when the config is a WebRTC ↔ SIP bridge.Connector