FreePBX with Asterisk provides VoIP telephony, auto-attendant, voicemail, call recording, and CTI integration with SuiteCRM.
Category: communications · Phase: 3 · Server: lab-pbx1
Ports: 5060 (SIP), 5061 (SIP TLS), 80 (Admin HTTP), 10000-20000/udp (RTP)
# Clone and run standalone lab
git clone https://github.com/it-stack-dev/it-stack-freepbx.git
cd it-stack-freepbx
make test-lab-01| Lab | Name | Duration | Purpose |
|---|---|---|---|
| 01-standalone | Standalone | 30–60 min | Basic functionality in isolation |
| 02-external | External Dependencies | 45–90 min | Network integration, external services |
| 03-advanced | Advanced Features | 60–120 min | Production features, performance |
| 04-sso | SSO Integration | 90–120 min | Keycloak OIDC/SAML authentication |
| 05-integration | Advanced Integration | 90–150 min | Multi-module ecosystem integration |
| 06-production | Production Deployment | 120–180 min | HA cluster, monitoring, DR |
See $repo.yml for full module metadata.
See CONTRIBUTING.md and the organization guide.
Apache 2.0 — see LICENSE.