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import { test } from '@playwright/test';
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import { markManual } from '../../helpers/manual';
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// All tests in this group require active SIP endpoints, PSTN access, or real telephony
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// infrastructure. They cannot be driven by Playwright + MCP alone.
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test.describe('pbx-calls', () => {
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test('pbx-calls-check-internal_extension_to_extension_audio_call', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires two active SIP endpoints (phones or softphones) and human verification of two-way audio quality');
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testInfo.annotations.push({ type: 'description', description: 'Verify an internal extension-to-extension audio call connects, audio is bi-directional, and CDR is generated' });
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// Manual steps:
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// 1. Register extension 1001 on Phone A, extension 1002 on Phone B.
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// 2. From Phone A, dial 1002.
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// 3. Verify Phone B rings and shows Caller ID of 1001.
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// 4. Answer on Phone B. Verify two-way audio on both handsets.
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// 5. Hang up. Verify both phones return to idle state.
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// 6. In admin portal → Logs → CDR, verify a CDR record is created with correct extension, duration, direction.
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// 7. Record: call setup time (ms), audio quality (MOS if tool available), CDR accuracy.
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});
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test('pbx-calls-check-inbound_call_from_sip_trunk_pstn', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires live SIP trunk connectivity to PSTN, a real DID number, and a phone to receive the call');
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testInfo.annotations.push({ type: 'description', description: 'Verify an inbound call from a SIP trunk (PSTN) routes correctly to the configured destination' });
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// Manual steps:
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// 1. Ensure a SIP trunk is registered and active in admin portal.
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// 2. Configure an inbound route for the DID to an extension or ring group.
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// 3. Dial the DID from an external PSTN number.
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// 4. Verify the call rings the configured destination.
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// 5. Answer and verify two-way audio.
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// 6. Check CDR shows inbound direction, correct DID, and trunk name.
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});
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test('pbx-calls-check-outbound_call_to_sip_trunk_pstn_with_caller_id', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires live SIP trunk, ability to dial a PSTN number, and verification of Caller ID as seen by the receiving party');
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testInfo.annotations.push({ type: 'description', description: 'Verify outbound call via SIP trunk completes and presents the correct Caller ID on the receiving end' });
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// Manual steps:
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// 1. From a registered extension, dial an outbound number (e.g., 9 + number).
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// 2. Verify the call routes through the configured outbound rule and SIP trunk.
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// 3. Answer on the receiving PSTN phone. Verify Caller ID matches configured outbound CID.
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// 4. Confirm two-way audio.
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// 5. Review CDR for outbound direction, trunk used, and Caller ID value.
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});
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test('pbx-calls-check-caller_id_presentation_restriction_and_pai', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires active call sessions and SIP packet capture (Wireshark) to verify P-Asserted-Identity header values in SIP INVITE messages');
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testInfo.annotations.push({ type: 'description', description: 'Verify Caller ID presentation, Anonymous restriction (CLIR), and P-Asserted-Identity (PAI) header in SIP INVITE' });
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// Manual steps:
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// 1. Test CLIR: dial *67 + number from a registered extension; verify receiving end sees Anonymous/Unknown.
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// 2. Capture SIP traffic with Wireshark during a call with PAI configured.
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// 3. Inspect INVITE packets: verify P-Asserted-Identity header contains the correct identity.
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// 4. Test outbound CID override: verify configured CID appears in From and PAI headers.
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// 5. Document: PAI value observed, From header value, CLIR behaviour.
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});
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test('pbx-calls-check-call_hold_resume_and_music_on_hold', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires an active two-party call; Hold/Resume is a mid-call feature and Music on Hold requires audio monitoring with actual SIP phone hardware');
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testInfo.annotations.push({ type: 'description', description: 'Verify call hold, resume, and Music on Hold (MoH) playback work correctly' });
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// Manual steps:
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// 1. Establish a call between two extensions.
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// 2. Press Hold on Phone A. Verify Phone B hears Music on Hold.
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// 3. Verify hold music is the configured MoH file (recognisable audio).
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// 4. Press Resume on Phone A. Verify two-way audio is restored instantly.
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// 5. Record: hold activation time, MoH audio quality, resume time.
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});
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test('pbx-calls-check-call_waiting_notification_and_switching', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires concurrent call sessions on the same extension with a physical or softphone capable of handling call-waiting signals');
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testInfo.annotations.push({ type: 'description', description: 'Verify call waiting notification is presented and the user can switch between active and waiting calls' });
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// Manual steps:
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// 1. Establish an active call on extension 1001.
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// 2. While 1001 is in-call, have a second caller dial 1001.
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// 3. Verify extension 1001 hears call-waiting tone / sees notification.
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// 4. Switch to the waiting call (flash / hold key). Verify first caller is put on hold.
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// 5. Switch back. Verify first call is resumed and second caller is on hold.
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// 6. Hang up both calls. Record behaviour.
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});
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test('pbx-calls-check-three_way_conference_and_multi_party_bridge', async ({ page }, testInfo) => {
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testInfo.annotations.push({ type: 'automationStatus', description: 'manual' });
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testInfo.annotations.push({ type: 'testType', description: 'check' });
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markManual('Requires three or more active SIP endpoints and a working conference bridge; audio verification for all participants requires human listeners');
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testInfo.annotations.push({ type: 'description', description: 'Verify three-way conference call and multi-party conference bridge functionality' });
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// Manual steps:
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// 1. Extension 1001 calls 1002. Both are in active audio call.
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// 2. 1001 initiates a transfer/conference to 1003.
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// 3. Verify all three parties hear each other clearly.
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// 4. Verify admin portal shows the conference with 3 participants.
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// 5. One participant drops. Verify remaining two continue their call.
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// 6. Test via dedicated conference room: dial conference number, enter PIN, verify all hear each other.
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});
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});
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