cambios profundos con grok
This commit is contained in:
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# Updated via Autotest Editor - 2026-05-25T21:52:12.022Z
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# Source of truth: this YAML file
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id: pbx-calls-check-call_hold_resume_and_music_on_hold
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module: pbx
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group: pbx-calls
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title: Verify call hold, resume, and Music on Hold playback work correctly
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description: Verify call hold, resume, and Music on Hold playback work correctly
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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 an active two-party call; MoH requires audio monitoring with real SIP phone hardware
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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 an active two-party call; MoH requires audio monitoring with real SIP phone hardware
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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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@@ -0,0 +1,53 @@
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# Updated via Autotest Editor - 2026-05-25T21:52:12.025Z
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# Source of truth: this YAML file
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id: pbx-calls-check-call_waiting_notification_and_switching
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module: pbx
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group: pbx-calls
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title: Verify call waiting notification and switching between active and waiting calls
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description: Verify call waiting notification and switching between active and waiting calls
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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: >-
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Requires concurrent call sessions on the same extension with physical/softphone capable of
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call-waiting
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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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- >-
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Requires concurrent call sessions on the same extension with physical/softphone capable of
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call-waiting
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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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@@ -0,0 +1,49 @@
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# 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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@@ -0,0 +1,49 @@
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# Updated via Autotest Editor - 2026-05-25T21:52:12.027Z
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# Source of truth: this YAML file
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id: pbx-calls-check-inbound_call_from_sip_trunk_pstn
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module: pbx
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group: pbx-calls
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title: Verify inbound call from a SIP trunk/PSTN routes correctly to the configured destination
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description: Verify inbound call from a SIP trunk/PSTN routes correctly to the configured destination
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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 live SIP trunk connectivity to PSTN and a real DID number
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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 live SIP trunk connectivity to PSTN and a real DID number
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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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@@ -0,0 +1,49 @@
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# Updated via Autotest Editor - 2026-05-25T21:52:12.028Z
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# Source of truth: this YAML file
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id: pbx-calls-check-internal_extension_to_extension_audio_call
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module: pbx
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group: pbx-calls
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title: Verify internal extension-to-extension audio call connects with bi-directional audio
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description: Verify internal extension-to-extension audio call connects with bi-directional audio
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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 two active SIP endpoints and human verification of two-way audio quality
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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 two active SIP endpoints and human verification of two-way audio quality
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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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+49
@@ -0,0 +1,49 @@
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# Updated via Autotest Editor - 2026-05-25T21:52:12.029Z
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# Source of truth: this YAML file
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id: pbx-calls-check-outbound_call_to_sip_trunk_pstn_with_caller_id
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module: pbx
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group: pbx-calls
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title: Verify outbound call via SIP trunk presents the correct Caller ID on the receiving end
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description: Verify outbound call via SIP trunk presents the correct Caller ID on the receiving end
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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 live SIP trunk and verification of Caller ID at receiving PSTN party
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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 live SIP trunk and verification of Caller ID at receiving PSTN party
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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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@@ -0,0 +1,49 @@
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# Updated via Autotest Editor - 2026-05-25T21:52:12.030Z
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# Source of truth: this YAML file
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id: pbx-calls-check-three_way_conference_and_multi_party_bridge
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module: pbx
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group: pbx-calls
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title: Verify three-way conference calling and multi-party conference bridge functionality
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description: Verify three-way conference calling and multi-party conference bridge functionality
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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
|
||||
- label: Credentials
|
||||
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: []
|
||||
steps:
|
||||
- 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: ''
|
||||
- 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 three or more active SIP endpoints and audio verification by human listeners
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automatable:
|
||||
- Most UI-driven configuration and verification steps via Playwright
|
||||
notYet:
|
||||
- Requires three or more active SIP endpoints and audio verification by human listeners
|
||||
- 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
|
||||
Reference in New Issue
Block a user