cambios profundos con grok

2026-05-26 00:42:42 +02:00
parent ee8cfa03ea
commit 19ea13dadd
99 changed files with 1447 additions and 2597 deletions
@@ -0,0 +1,54 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.031Z
+# Source of truth: this YAML file
+
+id: pbx-calls-security-check-acl_ip_whitelisting_and_rate_limiting
+module: pbx
+group: pbx-calls-security
+title: Verify ACL IP whitelisting rules and SIP rate limiting are configured and enforced
+description: Verify ACL IP whitelisting rules and SIP rate limiting are configured and enforced
+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: Log in to the admin portal (if required)
+    expected: ''
+  - action: Navigate to the relevant settings section
+    expected: ''
+  - action: Read current configuration values via UI or API
+    expected: ''
+automation_status: partial
+file: tests/pbx/calls-security.spec.ts
+custom_fields:
+  original_action: check
+  partialNotes: >-
+    ACL/rate-limit config via portal automated; functional enforcement testing requires
+    network-level testing from non-whitelisted IP
+automatable:
+  - Portal UI configuration and verification steps (already partially automated)
+  - Data extraction and reporting of current settings
+notYet:
+  - >-
+    ACL/rate-limit config via portal automated; functional enforcement testing requires
+    network-level testing from non-whitelisted IP
+  - 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
@@ -0,0 +1,54 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.032Z
+# Source of truth: this YAML file
+
+id: pbx-calls-security-check-srtp_sdes_or_dtls_media_encryption
+module: pbx
+group: pbx-calls-security
+title: Verify SRTP with SDES or DTLS-SRTP is enabled for media encryption on calls
+description: Verify SRTP with SDES or DTLS-SRTP is enabled for media encryption on 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: ''
+  - action: Log in to the admin portal (if required)
+    expected: ''
+  - action: Navigate to the relevant settings section
+    expected: ''
+  - action: Read current configuration values via UI or API
+    expected: ''
+automation_status: partial
+file: tests/pbx/calls-security.spec.ts
+custom_fields:
+  original_action: check
+  partialNotes: >-
+    Encryption config verified via admin portal; confirming live RTP is encrypted requires Wireshark
+    during active call
+automatable:
+  - Portal UI configuration and verification steps (already partially automated)
+  - Data extraction and reporting of current settings
+notYet:
+  - >-
+    Encryption config verified via admin portal; confirming live RTP is encrypted requires Wireshark
+    during active call
+  - 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
@@ -0,0 +1,47 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.033Z
+# Source of truth: this YAML file
+
+id: pbx-calls-security-check-toll_fraud_prevention_rules_and_alerting
+module: pbx
+group: pbx-calls-security
+title: Verify toll fraud prevention rules and alert notifications are configured
+description: Verify toll fraud prevention rules and alert notifications are configured
+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: Log in to the admin portal (if required)
+    expected: ''
+  - action: Navigate to the relevant settings section
+    expected: ''
+  - action: Read current configuration values via UI or API
+    expected: ''
+automation_status: automated
+file: tests/pbx/calls-security.spec.ts
+custom_fields:
+  original_action: check
+automatable:
+  - Full end-to-end execution via Playwright against the admin portal UI
+  - Navigation, form interaction, and assertions are scriptable
+notYet:
+  - Real RTP/SRTP media path and audio quality verification (human ear or PESQ/MOS tools)
+  - External SIP trunk / PSTN connectivity and DID numbers