AI Native Call Center is a self-hosted call center where a voice model answers the phone first and people step in only when the conversation actually needs one. Calls land on FreeSWITCH and connect straight to a speech-to-speech provider, and a flow engine steers each call through phases that carry instructions and the tools the model may use, with transitions firing on tool results. When a person is required, the call transfers into a real queue and the agent's browser screen has already popped with the caller's details and the bot-phase transcript ready to read. Agents handle presence, call control, and callbacks from the web interface, and a companion browser extension holds their SIP registration and audio. Supervisors watch a live wallboard with the queue view and the agent roster. Persona, rules, and voice are published and versioned together, and flows are plain documents you can author and validate before loading. Every conversation, bot-answered, handed over, and finished by an agent, lands in a single call record holding one transcript and one recording stored on a filesystem or an S3-compatible bucket. The interface and the bot work in English and Chinese, with provider choices for different regions including an option to run your own realtime gateway. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache-2.0 licensed.

A SIP phone system lives or dies on the unglamorous middle. Something has to route every call, know who is registered, wake phones that are asleep, and translate between networks that disagree. Flexisip is that middle, built as a modular server suite where you deploy only the pieces your service needs. The proxy routes calls and messages, relays media with ICE, STUN, and TURN support, and authenticates users by digest, TLS certificates, or OpenID Connect. Its push gateway speaks RFC 8599 to Apple and Google, so incoming calls ring phones whose app is closed, and it can lend that gift to an existing Asterisk or FreeSWITCH platform that lacks push of its own. The presence server tracks who is reachable and shares contact lists, the back-to-back user agent bridges SIP networks to PSTN gateways and trunks with codec transcoding and caller identity translation, and a registration event server notifies peer domains when users register. Instances cluster with load balancing for services that outgrow one box, the same suite shrinks onto a Raspberry Pi class intercom, and it has quietly powered the public linphone.org SIP service for over a decade. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. AGPL-3.0 licensed.