50 lines
2.0 KiB
YAML
50 lines
2.0 KiB
YAML
# Updated via Autotest Editor - 2026-05-25T21:52:12.026Z
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# Source of truth: this YAML file
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id: pbx-calls-check-caller_id_presentation_restriction_and_pai
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module: pbx
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group: pbx-calls
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title: Verify Caller ID presentation, Anonymous restriction (CLIR), and P-Asserted-Identity header
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description: Verify Caller ID presentation, Anonymous restriction (CLIR), and P-Asserted-Identity header
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priority: Medium
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tags: []
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references: []
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dependencies:
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- label: Admin Portal
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value: https://pbx.local:4443 (configure via $BASE_URL)
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- label: Repository URL
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value: ${BASE_URL}/downloads or $REPO_URL
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- label: Protocol
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value: HTTPS + SIP (UDP/TCP/TLS) + RTP/SRTP
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- label: SIP Endpoints
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value: 'At least 2 registered SIP clients (deskphone or softphone: Zoiper, MicroSIP, Yealink, etc.)'
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- label: Network
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value: SIP trunk or local Asterisk/FreeSWITCH simulator for PSTN simulation
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- label: Credentials
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value: Admin user with sufficient privileges (portal + SSH where needed)
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- label: Test Data
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value: Clean test tenant or lab PBX instance (recommended isolated environment)
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preconditions: []
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postconditions: []
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steps:
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- action: Open admin portal or repository URL in browser context
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expected: ''
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- action: Ensure SIP endpoints / softphones are registered
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expected: ''
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- action: Initiate or receive call using the required scenario
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expected: ''
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- action: Verify audio path, signaling, and media parameters
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expected: ''
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automation_status: manual
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file: tests/pbx/calls.spec.ts
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custom_fields:
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original_action: check
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manualReason: Requires active call sessions and SIP packet capture to verify PAI header values
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automatable:
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- Most UI-driven configuration and verification steps via Playwright
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notYet:
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- Requires active call sessions and SIP packet capture to verify PAI header values
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- Requires external SIP endpoints or physical phones for media/audio validation
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- Real RTP/SRTP media path and audio quality verification (human ear or PESQ/MOS tools)
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- External SIP trunk / PSTN connectivity and DID numbers
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