Commit Graph
10 Commits
Author SHA1 Message Date
iamdoubz aaa294ae6a release: version 0.0.6
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2026-10-21 20:24:46 -05:00
iamdoubz d9f9c6e444 release: version 0.0.5
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2026-10-21 18:46:04 -05:00
iamdoubz 4a175a0d77 release: version 0.0.4
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release / images (release) Successful in 38s
2026-10-21 16:22:54 -05:00
iamdoubz f2868cf5d3 settings: the SIP port, with a check against other listeners
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2026-10-21 16:12:12 -05:00
iamdoubz 6893c55e1b settings: a switch for SIP over TCP
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2026-10-21 15:22:39 -05:00
iamdoubz 4143f6d064 probe: downloadable executables for machines without Docker
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Each release attaches selfphoneng-probe-linux-x64 and -arm64 (Node's single executable
applications: esbuild bundle + postject into the official Node binary, checked against
nodejs.org's SHASUMS256.txt), the bundle as selfphoneng-probe.cjs (Node 22+) and SHA256SUMS.
Built in binary.Dockerfile with BuildKit's local output, so Gitea's runner can build it too.
deploy/probe/selfphoneng-probe.service runs it under systemd, sandboxed.
2026-10-21 12:57:04 -05:00
iamdoubz 65a0438dde diagnostics: System → Diagnostics with a network test probe; RTP_PORT_START/END
The server checks itself (public IP, DNS, local networks, phone ACL, SIP transports, free audio
ports) and says how to fix each problem. A quick or complete network test asks the probe
(apps/probe, own image) to test this server from outside: HTTP/HTTPS, SIP over UDP/TCP/TLS, the
browser phone's /ws, sampled audio ports. The probe only tests the address that asks.
The audio port range comes from RTP_PORT_START/END (default 10000-20000).
2026-10-21 12:37:16 -05:00
iamdoubz f6a9b9a4b1 calls: limit outside calls at once, per extension and in total
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Owner's decision after the M11-T8 review: nothing capped how many calls a stolen password could
run at once. Settings → Security: calls.maxPerExtension (3) and calls.maxTotal (10), 0 = no
limit. Each sp-route-* counts the caller's extension (or whose forward it is: SP_LIMITEXT from
sp-sub-forward) with GROUP(spext) and every outside call with GROUP(spout); over a limit the
caller hears "all circuits are busy". 911 is never limited.

trunk-test.ts on a real Asterisk, limits 1 and 2: a 2nd call from 101's phone and a 3rd in total
are refused (with both 0 the same check fails). call-limits.test.ts covers the 0 combinations.
Also fixes the Security tab's hint: browser phones still work from anywhere.

Task: M11-T8
2026-10-20 13:14:20 -05:00
iamdoubz a1525088dc admin: System page with restart steps, Record calls on the Calling tab
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System shows whether telephony runs and needs a restart, and how to restart it
(docker compose restart asterisk); the dashboard's restart warning, the Network hint and
the public IP email point there. The user drawer's Calling tab gets the Record calls
select the spec listed (recordMode, already in the dialplan). e2e m11-system.mjs.

Task: M11-T7
2026-10-20 09:58:11 -05:00
iamdoubz 324f9d9fe2 docs: admin guide (install, voip.ms, network, E911, troubleshooting, updating)
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docs/admin-guide/: install and the setup wizard, voip.ms with screenshots, NAT and SIP
ALG, firewall (ufw, nftables limited to the voip.ms POP), fail2ban and CrowdSec, E911
duties, recording consent, troubleshooting (08 §3.3), updating, backup and restore,
monitoring and log rotation. scripts/update.sh as 08 §3.1 describes. The e2e checks
fail2ban's and CrowdSec's stock Asterisk rules against a real failed sign-in, takes the
screenshots and checks every link and anchor.

Task: M11-T7
2026-10-20 08:57:54 -05:00