List all available WITH_* feature flags used in OpenSIPS configuration, providing a Kamailio-compatible vocabulary to support engineers migrating between SIP server projects.
Apache 2.0
340,324 tools. Last updated 2026-07-30 07:19
"Kamailio" matching MCP tools:
- Build a flat OpenSIPS configuration from WITH_* flags by inlining only enabled blocks, eliminating nested #!ifdef clutter for a clean, validated output.Apache 2.0
- Evict a SIP registration binding for a trunk credential. First find the contact ID with trunk_list_credential_registrations.MIT
- Update a dispatcher endpoint on an owned trunk by providing host, port, transport, or a raw SIP URI. Include trunk ID and endpoint ID to modify priority, weight, or enabled status.MIT
- List live SIP registrations for a credential's AOR to see registered devices or find a contact_id for revoking registrations.MIT
- Add a SIP endpoint to an owned trunk using structured host/port/transport or a raw URI. Handles deferral warnings when Kamailio sync is pending.MIT
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- Delete a SIP credential from a trunk group, with automatic removal from Redis and Kamailio usrloc. Idempotent request that returns 204 when already deleted.MIT
- Remove a dispatcher endpoint from a trunk group using its trunk ID and endpoint ID. Requires prior lookup with trunk_list_endpoints.MIT
- Add an IP/netmask to a trunk's access control list and sync to Kamailio, enabling that source IP to send 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, no persistence] Take two SIP messages (typically the same request observed at two adjacent hops - e.g. the INVITE leaving FreeSWITCH and the INVITE arriving at Kamailio) and surface a structured per-header diff: `added`, `removed`, `mutated` (with old/new value), `duplicated` (single header → many), `de-duplicated`, `whitespace-only-change`, `parameter-reorder` (Via params, From tag), and `body-changed`. SDP bodies on both sides are delegated to `compareSdp` for codec / DTLS / ICE diffs. Use FIRST when the user has two captures or two log lines that should be carrying the same message and wants to know what an intermediate proxy / SBC / B2BUA changed. Far more reliable than visual inspection. Pair with: `parse_sip_message` to inspect either side in isolation; `lint_sip_request` if the diff reveals the downstream side became malformed; `search_sip_docs(vendor=<intermediate>)` once you know which hop's behavior is the source of the change.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