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 +1,53 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.035Z
+# Source of truth: this YAML file
+
+id: pbx-codecs-check-codec_negotiation_g711_g729_opus_g722
+module: pbx
+group: pbx-codecs
+title: Verify successful codec negotiation for G.711, G.729, Opus, and G.722
+description: Verify successful codec negotiation for G.711, G.729, Opus, and G.722
+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/codecs.spec.ts
+custom_fields:
+  original_action: check
+  manualReason: >-
+    Requires SIP call traffic with codec-constrained endpoints and Wireshark packet capture to
+    verify SDP negotiation
+automatable:
+  - Most UI-driven configuration and verification steps via Playwright
+notYet:
+  - >-
+    Requires SIP call traffic with codec-constrained endpoints and Wireshark packet capture to
+    verify SDP negotiation
+  - 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 +1,53 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.036Z
+# Source of truth: this YAML file
+
+id: pbx-codecs-check-dtmf_transmission_rfc2833_inband_sip_info
+module: pbx
+group: pbx-codecs
+title: Verify DTMF digit transmission via RFC2833, in-band audio, and SIP INFO methods
+description: Verify DTMF digit transmission via RFC2833, in-band audio, and SIP INFO methods
+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/codecs.spec.ts
+custom_fields:
+  original_action: check
+  manualReason: >-
+    Requires active call sessions with DTMF generation, packet capture, and IVR system for
+    verification
+automatable:
+  - Most UI-driven configuration and verification steps via Playwright
+notYet:
+  - >-
+    Requires active call sessions with DTMF generation, packet capture, and IVR system for
+    verification
+  - 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 +1,49 @@
+# Updated via Autotest Editor - 2026-05-25T21:52:12.037Z
+# Source of truth: this YAML file
+
+id: pbx-codecs-check-t38_fax_passthrough_and_error_correction
+module: pbx
+group: pbx-codecs
+title: Verify T.38 fax passthrough and error correction (ECM) work correctly
+description: Verify T.38 fax passthrough and error correction (ECM) work correctly
+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/codecs.spec.ts
+custom_fields:
+  original_action: check
+  manualReason: Requires fax hardware or fax emulator (HylaFAX) and T.38 stack support
+automatable:
+  - Most UI-driven configuration and verification steps via Playwright
+notYet:
+  - Requires fax hardware or fax emulator (HylaFAX) and T.38 stack support
+  - 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