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Mentor PBX
NewMENTOR on 00, aliases and a line in every country

Calls, meetings and chat, rewired for the web.

Mentor is a cloud PBX, HD video meetings and team chat in one multi-tenant platform — built on SIP, WebRTC and open standards, with a softphone that runs right in your browser and on your iPhone.

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Built on
  • SIP
  • WebRTC
  • Opus
  • SRTP
  • ICE

The product

Every conversation, one mentor.

Calls, meetings, chat and the contact center share one identity, one presence model and one API — so a ring, a meeting link and a message are all part of the same conversation.

01Calls

A full softphone that lives in a tab.

Place and answer calls straight from the browser over SIP on WebSockets — no plug-ins, no desk phone required. Hold, transfer and keypad work the way people expect, with live call-quality scores on every call. The same extension rings on your iPhone, even when the app is closed.

  • HD voice with G.722 wideband audio, encrypted with DTLS-SRTP
  • Blind and attended transfer, hold, mute and DTMF (RFC 4733)
  • Browsers, SIP desk phones and the iPhone app ring on the same extension
  • Forwarding, do not disturb and a voicemail greeting of your own
  • Park, pickup, intercom, paging and conference rooms on 21 feature codes

02SBC & media

An edge that keeps media honest.

Every call enters through a Kamailio session border controller that authenticates, rate-limits and routes per tenant. rtpengine anchors the media and bridges browser DTLS-SRTP to plain RTP for the PBX, so NAT, encryption and codecs are handled before a call reaches your dial plan.

  • Registrar, digest auth, flood limits and scanner bans at the edge
  • DTLS-SRTP ↔ RTP bridging with ICE, rtcp-mux and BUNDLE
  • SIP over TLS with SRTP for desk phones and the iPhone app, which a push wakes for a call
  • Tenant-aware dispatch to Asterisk with automatic failover
  • Calls between organisations by alias, extension or account number

03Meetings

Meetings that pick up where calls leave off.

HD video rooms run on a LiveKit SFU with simulcast and an active-speaker spotlight. Turn any phone call into a video meeting in one click, admit guests from a lobby, let people dial in from a regular phone, and bring a camera or OBS in as a stream.

  • Active-speaker layouts, screen share and recording
  • Lobby with admit and deny, plus host controls
  • Call-to-video escalation and SIP phone dial-in
  • Streaming in from an encoder over WHIP (RFC 9725)

04IVR builder

Design call flows the way you sketch them.

Wire menus, business hours, HTTP lookups and transfers together on a visual canvas. Every publish is a new version, so you can test, ship and roll back call routing without editing a dial plan.

  • Menus, gathers, schedules, conditions and timers
  • HTTP request nodes to route on CRM data
  • Versioned publishing with one-click restore
  • An AI-agent step that hands callers to MENTOR and carries on with its answer

05Queues & ACD

Queues your agents actually like.

Route callers to the right agent with longest-idle, round-robin, ring-all or skills-based strategies. Position announcements, wrap-up time and a real-time wallboard keep agents, supervisors and callers in the loop.

  • Atomic agent reservation: never two calls offered to one agent
  • Agent states, wrap-up and automatic away after missed offers
  • SLA, speed of answer and abandon rate on the wallboard

06Team chat

Chat that knows who's on a call.

Channels, DMs and threads with presence that reflects real call and meeting state, so you know who is free before you ring. Mentions, reactions, read receipts and files arrive instantly over Centrifugo.

  • Presence derived from calls, meetings and devices
  • Threads, quoted replies, mentions, reactions and read receipts
  • Full-text search across every conversation

07API & webhooks

Everything in the app is an API call.

The REST API is described by an OpenAPI 3.1 spec, errors are RFC 9457 problem details, and organization API keys authenticate with a single header. Subscribe to call, queue, meeting and chat events with signed Standard Webhooks.

  • OpenAPI 3.1 reference with interactive docs
  • Scoped organization API keys via x-api-key
  • Signed, retried Standard Webhooks deliveries
  • An MCP server, so AI assistants can run a workspace with your permissions

What's new · October 2026

New on the platform.

The latest additions, all live today: an AI agent on 00 in every workspace, aliases that reach you anywhere, a line in every country, Live Radio, on-server transcripts and toll-fraud controls.

  • MENTOR on 00, in every workspaceAI agent

    Dial 00 from any phone, or open a chat, and MENTOR — the AI agent — answers. It is seated on extension 00 in every workspace automatically, and IVR flows can hand callers to it.

  • Aliases and calls between organisationshandle@mentoragent.ai

    Everyone can have an alias such as maya@mentoragent.ai that reaches them wherever their extension is. Call other organisations by alias or extension — each one chooses whether to take part.

  • A line in every country249 countries · E.164

    A workspace for every country, with its dial plan, time zone and emergency numbers set. Mentor Mobile users get a permanent 16-digit number that only their friends can call.

  • Live Radio on *5Opus end to end

    Dial *5 and a station code to hear the world’s radio stations live. Service routes keep Opus untouched from the phone to the station, so music sounds like music.

  • Transcripts on your own serverWhisper

    Voicemail to text, transcripts of recorded calls and live captions, all transcribed on the platform’s own server — no audio leaves it. Summaries with action items arrive after the call.

  • Toll-fraud controls and a dial testerCalling policy

    Cap concurrent calls, calls per minute and daily spend, block premium and international numbers, lock dialling after hours — and see exactly what any number would do before anyone dials it.

  • Mentor Mobile for iPhoneBeta · TestFlight

    Your extension on your iPhone, over SIP-TLS with encrypted audio. The platform wakes the app with a push, so calls ring even when it is closed.

  • A live map of the platformHealth · alerts

    Every service on one animated map — call audio, signalling and data paths, with latency — plus nightly backups proven by a restore, and critical alerts pushed to administrators’ iPhones.

Platform

Seven servers. One platform.

Battle-tested open-source engines do the real-time heavy lifting; typed TypeScript services carry the product logic around them. No proprietary SIP stack, no black boxes.

  • PBX

    Asterisk 22 LTS · ARI

    Call routing with dial plans for 249 countries, extensions, ring groups, park, pickup, paging, conferences, voicemail, stereo recording and toll-fraud limits — driven over ARI by a typed router.

  • SBC

    Kamailio

    The SIP edge: WebSocket and UDP/TCP/TLS, registrar and auth, NAT handling, scanner bans, per-tenant routing, aliases and calls between organisations, and the push that wakes a phone.

  • Media

    rtpengine

    Anchors every edge stream, bridges DTLS-SRTP, SRTP and RTP, converts Opus and G.729 for mobile, and handles prompts, hold music and recordings.

  • Meeting

    LiveKit SFU

    Video rooms with lobby, host controls, recording, phone dial-in, streaming in over WHIP and one-click call escalation.

  • IVR

    Flow engine on ARI

    Runs versioned call-flow graphs: menus, DTMF, schedules, conditions, HTTP lookups, transfers and a hand-off to the MENTOR AI agent.

  • Queue / ACD

    Valkey-backed dispatcher

    Hold music, position announcements, routing strategies, agent states, wrap-up and live statistics.

  • IM

    Centrifugo

    DMs, channels, threads, reactions and presence, plus the bridge that streams events to every browser.

  • Control plane

    Fastify API · Next.js app

    Tenants, roles, provisioning, the public REST API, webhooks and an MCP server for AI tools — and the app you're looking at.

Underneath:Postgres · Valkey · NATS JetStream · S3-compatible storage · Whisper speech-to-textContainerised with Docker Compose for Ubuntu

Standards

Built on open standards.

Mentor interoperates because it implements the specs, not a proprietary dialect. Here is what ships today.

SIP signaling

The same SIP your desk phones speak — over UDP, TCP, TLS and WebSockets from the browser.

  • SIPRFC 3261
  • Offer/answerRFC 3264
  • Symmetric response (rport)RFC 3581
  • Reliable provisional (PRACK)RFC 3262
  • UPDATERFC 3311
  • Reason headerRFC 3326
  • INFORFC 6086
  • REFER (transfer)RFC 3515
  • ReplacesRFC 3891
  • SIP MESSAGE (texts)RFC 3428
  • Session timersRFC 4028
  • SIP OutboundRFC 5626
  • SIP over WebSocketRFC 7118
  • WebSocketRFC 6455

SDP & RTP

Session description and real-time transport.

  • RFC 8866SDP
  • RFC 3550RTP / RTCP
  • RFC 3551A/V profile
  • RFC 4585RTCP feedback
  • RFC 5761RTP/RTCP mux
  • RFC 5506Reduced-size RTCP
  • RFC 9143BUNDLE

Media security

Browser, phone and app media is encrypted end to the edge.

  • RFC 3711SRTP
  • RFC 4568SDES keys
  • RFC 5763DTLS-SRTP framework
  • RFC 5764DTLS-SRTP
  • RFC 8827WebRTC security

NAT traversal

Connectivity that survives real networks.

  • RFC 8445ICE
  • RFC 8838Trickle ICE
  • RFC 8489STUN
  • RFC 8656TURN

WebRTC

Standards-track browser real-time media.

  • RFC 9429JSEP
  • RFC 8825–8835WebRTC suite
  • RFC 8853Simulcast

Codecs

Wideband voice, scalable video and out-of-band DTMF.

  • RFC 6716Opus
  • RFC 7587Opus over RTP
  • ITU-TG.711
  • ITU-TG.722 wideband
  • ITU-TG.729
  • RFC 4733DTMF events
  • VideoVP8 · H.264
  • VideoVP9 / AV1 SVC

Web & API

The product surface, held to the same bar.

  • W3CWCAG 2.2 AA
  • RFC 9457Problem details
  • OASOpenAPI 3.1
  • SpecStandard Webhooks
  • SpecModel Context Protocol
  • ITU-TE.164 numbering
  • ISO3166 · 249 country dial plans

Desk phones

Lamps and message lights, the way phones expect them.

  • RFC 6665Event notification
  • RFC 3842Message waiting
  • RFC 4235Dialog events (BLF)

Recording & streaming

Calls out to your recorder, encoders into your meetings.

  • RFC 7866SIPREC
  • RFC 7865Recording metadata
  • RFC 9725WHIP

Sign-in

Single sign-on, passkeys and one-time codes.

  • SpecOpenID Connect
  • W3CWebAuthn passkeys
  • RFC 6238TOTP

On the roadmap: STIR/SHAKEN caller ID verification and attestation, SFrame end-to-end encryption for meetings (RFC 9605), T.38 fax and next-generation emergency calling (NG911).

Engineering commitments

  • Interaction budget<150 msINP target for every marketing page, and under 100 ms for chat and the dialpad.
  • Accessibility baselineWCAG 2.2 AABoth themes, full keyboard paths, and non-drag alternatives for every drag interaction.
  • Encrypted by defaultDTLS-SRTPBrowser calls and meetings carry encrypted media from the very first hop.
  • Self-hostableUbuntu + DockerEvery server ships as a container. Run Mentor in our cloud or on your own hardware.

Get started

Give your team a mentor.

Create a workspace, invite your team, claim extensions and place your first browser call in minutes — no desk phones required.

Explore the platform