# Updated via Autotest Editor - 2026-05-25T21:52:12.026Z # Source of truth: this YAML file id: pbx-calls-check-caller_id_presentation_restriction_and_pai module: pbx group: pbx-calls title: Verify Caller ID presentation, Anonymous restriction (CLIR), and P-Asserted-Identity header description: Verify Caller ID presentation, Anonymous restriction (CLIR), and P-Asserted-Identity header 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: '' - action: Ensure SIP endpoints / softphones are registered expected: '' - action: Initiate or receive call using the required scenario expected: '' - action: Verify audio path, signaling, and media parameters expected: '' automation_status: manual file: tests/pbx/calls.spec.ts custom_fields: original_action: check manualReason: Requires active call sessions and SIP packet capture to verify PAI header values automatable: - Most UI-driven configuration and verification steps via Playwright notYet: - Requires active call sessions and SIP packet capture to verify PAI header values - 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