Asterisk SIP Trunk Configuration on Linux

I need a reliable SIP trunking solution running on Asterisk under Linux. At the moment I’m unsure whether the server already in place can be salvaged or if it will be cleaner to start from scratch, so the first step will be a quick audit of what’s there. From that point we can move straight into installing / re-installing Asterisk, pointing it at my chosen SIP trunk provider, and fine-tuning the dial-plan so inbound and outbound calls flow smoothly.

Security hardening (fail2ban, TLS / SRTP where supported), codec optimisation for solid call quality, and basic call-flow features such as voicemail, IVR and call recording should all be factored in. I’ll need everything documented well enough that I can tweak extensions or add trunks later without starting over.

Acceptance criteria
• Asterisk running on Linux with the SIP trunk registered and passing both inbound and outbound calls.
• Dial-plan, voicemail and any IVR or recording rules functioning as agreed.
• Clear documentation of key configuration files, passwords stored separately, and a brief hand-off session so I understand what was done.

If that scope is clear and achievable for you, let me know your timeline and any additional information you require to get started.