Voice gateway, activation and deployment
Configure direct SIP-over-WSS, isolated switch output, activation snapshots, firewall/runtime publication, event bridge and live validation.
Guides in this workflow
Follow the guide that matches the exact action, filter, menu change or issue.
Understand direct SIP-over-WSS deployment
Use the supplied isolated voice runtime without enabling external trunks or PSTN.
Run the read-only voice audit
Inspect current services, profiles, ports, certificates and paths before changes.
Repair the internal SIP profile safely
Apply the supplied scoped profile correction only after audit evidence.
Configure xConnect and synchronise
Set WSS URL, domain and isolated switch path, then generate module-owned configuration.
Publish the WebRTC runtime
Expose the approved WSS/RTP runtime while keeping private SIP and event-control services protected.
Install switch synchronisation automation
Install the supplied controlled automation for queued/static configuration updates.
Install the event bridge
Configure the signed event service and secret file.
Open only required voice firewall paths
Permit the documented WSS/RTP traffic and keep SIP listeners/event control private.
Create an activation snapshot
Capture module settings and switch configuration evidence before activation changes.
Restore an activation snapshot
Use only a valid readable temporary snapshot under the allowed path and 128 MB limit.
Activate direct SIP
Apply the configured WSS/domain/path with rollback ownership and optional keep-paused controls.
Pause internal calling
Disable internal calling safely before maintenance or rollback.
Resume internal calling
Re-enable internal calling only after readiness checks pass.
Synchronise switch configuration
Generate, validate and install the current tenant/extension/route configuration.
Retain and use rollback evidence
Keep an activation/switch snapshot only for the controlled operation that owns rollback.
Verify WSS and TLS
Confirm secure WebSocket reachability and valid production certificates.
Verify browser media
Confirm allowed media paths and two-way audio.
Keep SIP and event control private
Do not expose internal SIP listeners or event-control interfaces publicly.
Run the live activation verifier
Complete the supplied live tests after static configuration passes.
Complete two-user UAT
Place a real two-browser call and confirm ringing, answer, audio, DTMF, hangup and event evidence.
Preserve the no-PSTN boundary
Do not add carrier trunks, public telephone numbers or external outbound routes.
Understand IVR operator fallback
Use 0/no-input fallback according to the generated account IVR configuration.