Open-source-first VoIP SaaS

Sufi Online

A production-oriented communication platform foundation for teams, contacts, realtime chat, document sharing, and future SIP interconnects.

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Operations cockpit

Live-ready

28

Contacts

39

Chat

Setup

SIP setup

61

Audit

SIP trunk status

Credentials required before PSTN calls are enabled.

Real workspace identity

Register, create organizations, manage users, teams, permissions, and audit trails.

Realtime communication

Search users, approve contacts, chat one-to-one, share files, and publish presence events.

VoIP management base

Persist SIP trunks, DIDs, routing rules, normalization, and engine event APIs without fake call data.

Mobile-ready architecture

Expo app scaffold and API contracts are prepared for native app-to-app calling flows.

VoIP future

Asterisk, Kamailio, RTPengine, and FreeSWITCH hooks are modeled without pretending calls work before trunks exist.

Identity first

Face-secured patient access and protected admin consultations are built into the platform.

Files and records

Uploads, sharing metadata, notifications, and audit logs are real backend records from day one.