Changelog

Versions follow semver. Each release's container images are built from its tag (08 §3.1): selfphoneng-api, selfphoneng-asterisk and selfphoneng-probe, on ghcr.io and on the Gitea registry. From v0.0.3 the release also has the probe as downloads for machines without Docker.

Unreleased

Admin guide: Analog phones (ATAs), setting up a Grandstream HT802 V2 by hand.
Dial plan: a user's Credentials tab shows a dial plan for analog phone adapters, made from your outbound routes and internal number lengths: 911 and full outside numbers go out the moment they're dialed instead of after a pause. Yealink desk phones get the same as dial-now rules.
Internal numbers can no longer start with 911 or 9911, so 911 can go out at once.
Local AI (admin guide: Local AI): Settings → AI sets a language model, speech to text and text to speech, each with a Test, on this server or the local network only (anything else is refused). deploy/ai/compose.yml is a second stack with Speaches (Whisper and Kokoro) on the GPU. AI_* in .env sets them on a new install.
Voicemail transcripts: with AI on, a new message's words are in the voicemail email and on the voicemail page. Database change: voicemail_messages.transcript (the api backs up first).

v0.0.4 — 2026-10-21

Updating: no database changes. Portainer: change both tags to 0.0.4 and redeploy. Compose: sudo scripts/update.sh v0.0.4.

Settings → Security has the SIP port and Accept SIP over TCP (5060 and on by default, as before). Both were settings with no field. Another port, such as 5080, helps when a router or the ISP keeps UDP 5060 for itself: a Complete network test shows UDP getting through while SIP on UDP 5060 doesn't. Changing it needs a telephony restart, a new router forward and setting the phones up again; trunks keep working. A port that another part of SelfPhoneNG listens on is refused. The admin guide (Network) explains when to change it.

v0.0.3 — 2026-10-21

Before you update: this version changes the database (music-on-hold order). The api backs up first, by itself (see the admin guide, Updating). Compose installs from v0.0.2 get scripts/update.sh with the release: git fetch --tags && git checkout v0.0.3, then sudo scripts/update.sh v0.0.3. Portainer: change both tags to 0.0.3 and redeploy. Optional: RTP_PORT_START/RTP_PORT_END on the api, then open that range instead of 10000–20000.

Any internal number: users, ring groups, queues, digital receptionists, conference rooms and paging groups take any free number of 1 to 6 digits, for example 1 for the first user (the setup wizard asks for it too). Settings → Numbering sets where Next free looks. A one-digit number and a menu key of a digital receptionist that lets callers dial numbers can't be the same. Ring groups, queues, voicemail, pickup (**), queue login (*45), forwarding (*72) and intercom (*80) now work with one-digit numbers.
Music on hold order: a set plays in the order shown on Sounds & music (move files up and down; new uploads go last), or Shuffle for random order. Before, the order was Asterisk's.
Reports: totals now include answered calls, talk time and outside minutes (in and out, roughly what voip.ms bills), with outside minutes per day, and the voip.ms balance of each trunk that has the API login.
System → Diagnostics: checks what usually keeps phones off the network (public IP, DNS, local networks, the phone network limit, SIP listening, free audio ports), with how to fix each. A Quick or Complete network test checks the ports from the internet through a network test probe you run outside your network; set its URL in Settings → Network. The probe comes as a Docker image (selfphoneng-probe), and, for a machine without Docker, as a single Linux executable (x64 and arm64) attached to each release, with a systemd unit.
The call audio port range is configurable: RTP_PORT_START and RTP_PORT_END (default 10000–20000).
Updating (Docker only): a new version backs up before it changes the database, and stops instead if the backup fails. scripts/update.sh updates a compose install to a release, checks the result and goes back by itself if the check fails. The admin guide's "Updating" section covers compose and Portainer.
Container logs are capped (3 × 10 MB) in docker-compose.yml.
Security leftovers from the review (08 §6), all fixed:
- Asterisk no longer loads the modules that can run host commands (app_system, func_shell, app_disa, res_agi, app_externalivr).
- Turning two-step sign-in on asks for your password again.
- *97/*98 voicemail logins: one at a time per phone, 10 an hour.
- Phone setup files: an unknown MAC gets the same answer as a known one.
- The login limiter no longer keeps every email ever tried until a restart.

v0.0.2 — 2026-10-21

Fix: calls to a user with Linphone on an iPhone registered dropped at once, for every device of that user. Its push parameters hold &, which split the dial string.

v0.0.1 — 2026-10-20

The first tagged build, for testing. Not for production yet: the v1.0 definition of done (docs/spec/09-roadmap.md) still needs a week on a real server with real calls, and the open real-account checks.

Phones, users and extensions; calls between desk phones, softphones and the browser phone.
voip.ms trunks (registration or IP auth, UDP/TCP/TLS + SRTP), phone numbers, outbound rules with calling permissions, 911 with E911 caller ID and alerts.
Voicemail with email, digital receptionists, office hours, ring groups, queues, parking, pickup, conference rooms, paging and intercom, music on hold.
Call history, recordings, the live dashboard (listen, whisper, barge), reports.
Desk phone provisioning (Yealink, Grandstream), company phonebook, Linphone setup QR.
Security: two-step sign-in for admins, phone network limit, limits on outside calls at once, the M11-T8 security review (08 §6); backups and restore; background watches; webhooks and API tokens; the admin guide.
Not in this release: SMS (parked until wanted), scripts/update.sh (before v1.0).

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