OpenSIPS TLS, Asterisk & REST
Budget: $250 – $750 USD
I need a complete yet streamlined deployment of OpenSIPS secured with TLS and fully integrated with Asterisk. The goal is to deliver a working voice platform that lets me:
• Route calls through OpenSIPS
• Handle voicemail
• Support user-to-user calls and call forwarding
On the web layer I want clean, well-documented REST endpoints so external services can create, monitor, and terminate calls or fetch voicemail status. You’re free to choose the most stable 3.x OpenSIPS release (3.1 LTS, 3.2 Latest, or 3.0 LTS) as long as you justify the decision and ensure future maintainability.
Scope I expect
1. Install and harden OpenSIPS on my VPS, enabling TLS for SIP traffic (certificate provisioning, cipher suite tuning, SRTP hand-off to Asterisk).
2. Install or reuse an Asterisk build and wire it to OpenSIPS for the listed features, providing clear dial-plan samples.
3. Expose a REST API layer (native OpenSIPS REST interface or an external micro-service) that lets me perform basic call control and voicemail queries.
4. Produce concise documentation: architecture diagram, annotated config files, and step-by-step deployment notes so I can reproduce the setup.
5. Run a demo session proving end-to-end encrypted calls, voicemail drop/retrieve, call forwarding, and a successful REST API call.
If you have prior experience with OpenSIPS + Asterisk pairings, especially under TLS, let me see it—speed and reliability matter to me.
• Route calls through OpenSIPS
• Handle voicemail
• Support user-to-user calls and call forwarding
On the web layer I want clean, well-documented REST endpoints so external services can create, monitor, and terminate calls or fetch voicemail status. You’re free to choose the most stable 3.x OpenSIPS release (3.1 LTS, 3.2 Latest, or 3.0 LTS) as long as you justify the decision and ensure future maintainability.
Scope I expect
1. Install and harden OpenSIPS on my VPS, enabling TLS for SIP traffic (certificate provisioning, cipher suite tuning, SRTP hand-off to Asterisk).
2. Install or reuse an Asterisk build and wire it to OpenSIPS for the listed features, providing clear dial-plan samples.
3. Expose a REST API layer (native OpenSIPS REST interface or an external micro-service) that lets me perform basic call control and voicemail queries.
4. Produce concise documentation: architecture diagram, annotated config files, and step-by-step deployment notes so I can reproduce the setup.
5. Run a demo session proving end-to-end encrypted calls, voicemail drop/retrieve, call forwarding, and a successful REST API call.
If you have prior experience with OpenSIPS + Asterisk pairings, especially under TLS, let me see it—speed and reliability matter to me.