FS PBX Documentation

website·Indexed 32 days ago

https://www.fspbx.com

Documentation and release notes for FS PBX, a VoIP PBX system. Includes guides on call flow configuration, firewall setup for OPNsense, pfSense, and MikroTik, Yealink RPS phone provisioning, Postmark webhook integration, and system administration including language settings and troubleshooting.

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What's inside FS PBX

  1. Overview of FS PBX Phone Provisioning

    Phone provisioning in FS PBX automates the setup and management of physical VoIP phones. Instead of manual configuration via a web interface, FS PBX acts as a central server. When a phone boots or resyncs, it contacts the FS PBX server; the server identifies the phone by its MAC address and delivers a dynamically generated configuration file. This process automatically applies connectivity settings (SIP credentials), user interface layouts (BLF, line keys), system preferences (time zones, firmware), and network settings (VLAN, QoS).
  2. Overview of Yealink RPS with FS PBX

    Yealink RPS (Redirection and Provisioning Service) allows Yealink phones to automatically find their provisioning server upon initial startup or factory reset. Instead of manual configuration or DHCP Option 66, the phone contacts Yealink's cloud, which redirects it to the FS PBX /prov/ endpoint. FS PBX then matches the phone's MAC address and serves the appropriate configuration template.
  3. Overview of FS PBX Architecture and Capabilities

    FS PBX is an enterprise-grade VoIP and unified communications platform built on FreeSWITCH and PostgreSQL. It is a redesigned evolution of FusionPBX, utilizing a modern tech stack including Laravel, Vue.js, Tailwind, and modern APIs to provide a multi-tenant, white-label capable environment suitable for small offices, large enterprises, and VoIP carriers.
  4. Overview of FS PBX Messaging Capabilities

    FS PBX provides a messaging framework for sending and receiving SMS and MMS messages. Key capabilities include:

    • Multi-Provider Support: Ability to use different messaging carriers (e.g., Bandwidth, Sinch, Commio, Telnyx, ClickSend) for different phone numbers.
    • MMS Support: Support for exchanging media/picture messages in addition to plain text.
    • Multi-Channel Delivery: Messages can be managed via the FS PBX dashboard, delivered to the mobile app, or sent as read-only notifications to an email address.
    • Two-Way Communication: Users can send and receive replies via the dashboard or mobile app.
  5. Overview of Direct Inward System Access (DISA)

    DISA (Direct Inward System Access) allows authorized users to access the PBX system from an external phone line. After authenticating with a PIN, users can place outbound calls or access internal features as if they were dialing from an internal extension. This is primarily used by remote employees or administrators to maintain PBX control and billing while calling remotely.
  6. Overview of FS PBX Architecture and Features

    FS PBX is a multi-platform VoIP core leveraging FreeSWITCH. It is rebuilt from the ground up using Laravel for the backend and Vue.js with Tailwind CSS for the frontend to provide a responsive, snappy UI. Key technical features include:

    • Multi-tenancy: Supports hosting multiple client PBX instances on a single system, where each tenant is isolated in the database and dial-plan.
    • Unified Login: Unlike FusionPBX, it uses a unified login portal where users are routed to their correct tenant space based on their email address.
    • High Availability: Supports master-to-master (bi-directional) PostgreSQL database replication and file synchronization between servers for automatic failover and zero downtime.
    • Core Capabilities: Includes SIP trunk management, IVRs, conferencing, and voicemail.
  7. Overview of Scheduled Job Server ownership

    The Scheduled job server control determines which FS PBX server acts as the owner for coordinated background work. In a redundant installation, one server is the active owner while others remain on standby. This setting is shared across all accounts and affects LDAP directory synchronization and Contact Center callbacks (including cleanup).

    Note that this control does not affect all scheduled tasks: S3 recording archiving, scheduled announcements, and certificate renewal use their own separate selection mechanisms (per-server or DNS-based).

  8. Overview of Poly Zero Touch (ZTP) for FS PBX

    Poly Zero Touch (ZTP) is a cloud redirection service that allows supported Poly phones to discover the FS PBX provisioning server automatically after a factory reset or first boot. This is particularly useful for remote deployments where the administrator does not control the local DHCP server.

    The provisioning flow is as follows:

    1. The phone contacts Poly Zero Touch via the internet.
    2. Poly Zero Touch identifies the phone's MAC address and redirects it to the FS PBX provisioning destination.
    3. The phone contacts the FS PBX /prov/ endpoint using configured HTTP credentials.
    4. FS PBX matches the MAC address and returns the assigned device template and configuration.
    5. The phone applies the configuration and registers its SIP lines.
  9. Understand FS PBX scheduled job ownership

    In redundant deployments, FS PBX ensures that jobs performing database writes are executed by only one authorized server (the owner). This prevents duplicate work and conflicting database rows that could stall logical replication. If a server cannot prove it is the active owner, the job is skipped and the reason is recorded. Unclaimed work remains due for the next scheduler pass.
  10. Understand DIDs and Gateways in FS PBX

    In FS PBX, a DID (Direct Inward Dialing) is a public phone number used by external callers to reach a business. It is distinct from a Gateway, which is the actual connection (trunk) to the VoIP carrier.

    The call flow is: Caller → Carrier → Gateway → FS PBX → DID match → Destination.

  11. Use built-in call transcription and summarization in FS PBX

    FS PBX provides built-in call transcription capabilities utilizing speech-to-text AI providers. This allows users to automatically convert call recordings into text and generate summaries of conversations.