sip: TLS transport with a checked certificate, TLS + SRTP trunks
The certificate lives in data/tls (fullchain.pem + privkey.pem), mounted read-only into asterisk and the api. Asterisk quietly drops transport-tls when the files are missing, so the api won't turn TLS on unless they load and match, shows the certificate on Settings -> Security, and warns on the dashboard when it expires within 14 days. scripts/tls-cert.sh copies it from the bundled Caddy (or given paths) and reloads res_pjsip: Asterisk 24 re-reads a changed certificate on reload, calls stay up. transport-tls checks the provider's certificate (verify_server). Asterisk 24 checks a qualify's certificate against the IP instead of the name, which fails on voip.ms, so TLS trunks don't qualify; REGISTER and INVITE verify the name fine. Task: M11-T2
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@@ -16,7 +16,7 @@ trunk fallback (same fields, no voip.ms-specific help text or API features).
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- Allowed codecs: **G.711u (ulaw)**. Others optional.
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- DTMF mode: **RFC 2833**
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- NAT: **Yes**
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- Encrypted SIP traffic (TLS/SRTP): optional. If on, set our trunk transport to TLS, port 5061, SRTP on.
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- Encrypted SIP traffic (TLS/SRTP): optional. If on, set our trunk transport to TLS, port 5061, SRTP on. Needs Settings → Security → Accept SIP over TLS (a certificate in `data/tls`, 08 §2.1) first. voip.ms's certificate names every POP, so the trunk checks it (`verify_server`).
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- CallerID Number: your main DID.
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- International calling: leave **disabled** unless needed (toll-fraud protection on their side too).
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2. **Choose a POP server** close to the PBX (e.g. `chicago1.voip.ms`). The sub-account registers to this server.
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