# Updated via Autotest Editor - 2026-05-27T09:41:49.884Z
# Source of truth: this YAML file
# Note: Expected results estimated based on step actions

id: pbx-calls-check-call_waiting_notification_and_switching
module: pbx
group: pbx-calls
title: Verify call waiting notification and switching between active and waiting calls
description: Verify call waiting notification and switching between active and waiting calls
priority: Medium
tags: []
references: []
dependencies:
  - label: Admin Portal
    value: https://pbx.local:4443 (configure via $BASE_URL)
  - label: Repository URL
    value: ${BASE_URL}/downloads or $REPO_URL
  - label: Protocol
    value: HTTPS + SIP (UDP/TCP/TLS) + RTP/SRTP
  - label: SIP Endpoints
    value: 'At least 2 registered SIP clients (deskphone or softphone: Zoiper, MicroSIP, Yealink, etc.)'
  - label: Network
    value: SIP trunk or local Asterisk/FreeSWITCH simulator for PSTN simulation
  - label: Credentials
    value: Admin user with sufficient privileges (portal + SSH where needed)
  - label: Test Data
    value: Clean test tenant or lab PBX instance (recommended isolated environment)
preconditions: []
postconditions: []
steps:
  - action: Open admin portal or repository URL in browser context
    expected: >-
      The admin portal or repository page loads successfully without errors, security warnings, or
      redirects.
  - action: Ensure SIP endpoints / softphones are registered
    expected: All required SIP endpoints show as registered with the correct status in the PBX.
  - action: Initiate or receive call using the required scenario
    expected: >-
      The call is successfully established between the endpoints with ringing and connection
      confirmed.
  - action: Verify audio path, signaling, and media parameters
    expected: >-
      Two-way audio is clear and bidirectional. Signaling (SIP) is correct and media parameters
      (codec, RTP/RTCP) match expectations.
automation_status: manual
file: tests/pbx/calls.spec.ts
custom_fields:
  original_action: check
  manualReason: >-
    Requires concurrent call sessions on the same extension with physical/softphone capable of
    call-waiting
automatable:
  - Most UI-driven configuration and verification steps via Playwright
notYet:
  - >-
    Requires concurrent call sessions on the same extension with physical/softphone capable of
    call-waiting
  - Requires external SIP endpoints or physical phones for media/audio validation
  - Real RTP/SRTP media path and audio quality verification (human ear or PESQ/MOS tools)
  - External SIP trunk / PSTN connectivity and DID numbers