# Changelog

Versions follow [semver](https://semver.org). 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.

## v0.0.7 — 2026-10-23

**Before you update:** this version changes the database (call reports, contacts); the api backs
up first, by itself. Portainer: change both tags to `0.0.7` and redeploy. Compose:
`sudo scripts/update.sh v0.0.7`. Contacts start in Simple mode with empty address books;
Contrack is optional and off until you set it up (admin guide: [Contacts](docs/admin-guide/contacts.md)).
Desk phones pick up their new directory at their next daily check (or reboot them).

- **Contacts** (spec 11, admin guide: [Contacts](docs/admin-guide/contacts.md)): each person's own
  contacts and the company's, in My Phone → Contacts and Admin → Contacts. Import `.vcf` files
  (iPhone, Android, Google, Outlook, Apple) or CSV, with a preview; duplicates are skipped. Share
  contacts with colleagues or everyone: they Accept or Deny each one, and nothing lands in their
  contacts without it. Contact names show in call history, voicemail, Home, the browser phone,
  and the call report email, with **Add to contacts** on calls and voicemails from numbers that
  aren't contacts yet.
- **Caller names on desk phones while they ring**: the caller's name from the ringing person's
  own contacts (or the company's), per phone, so nobody sees a colleague's private contact names;
  ring group and queue prefixes ("Sales:") stay in front.
- **Desk phone directories** now have the phone owner's own contacts, after the extensions and
  before the company's.
- **Contrack** (optional): the setup wizard (new step 8, Contacts) and Settings → Contacts offer
  [Contrack](https://github.com/arvarik/contrack), a self-hosted contact and relationship manager,
  as a second container (`deploy/contrack/compose.yml`, pinned to 2.0.2). Each person connects
  their own Contrack account in My settings → Contacts, and the company has one everyone sees.
  Calls go on the contact's Contrack timeline, optionally with the call summary and its action
  items (the first becomes a follow-up); contacts from before Contrack can be copied over.
- **Call report email** (My settings → Call report): your calls every hour, day or week, at the
  time you pick (company time zone), with whether each was recorded and, optionally, each
  recorded call's summary. No calls, no email.
- **Voicemail search**: search your messages by what was said (the transcript) or by the caller.
- Database changes: call report columns on `extensions`; `contact_shares`, `contrack_call_log`,
  and Contrack columns on `users` (the api backs up first).

## v0.0.6 — 2026-10-21

**Before you update:** this version changes the database (call transcripts); the api backs up
first, by itself. Portainer: change both tags to `0.0.6` and redeploy. Compose:
`sudo scripts/update.sh v0.0.6`. Call transcripts are off until you turn them on in Settings → AI;
calls recorded before that get no transcript.

- **Call transcripts and summaries** (Settings → AI, admin guide: [Local AI](docs/admin-guide/ai.md)):
  recorded calls are written out a minute or two after they end, caller and other side apart, with
  a summary and action items from the language model. The Recordings page shows them and searches
  what was said; the people on a call can read its transcript. Database change: four columns on
  `recordings` (the api backs up first).

## v0.0.5 — 2026-10-21

**Before you update:** this version changes the database (voicemail transcripts); the api backs
up first, by itself. Portainer: change both tags to `0.0.5` and redeploy. Compose:
`sudo scripts/update.sh v0.0.5`. The AI is optional and off until you set it up: a second stack
from `deploy/ai/compose.yml`, then Settings → AI (admin guide: [Local AI](docs/admin-guide/ai.md)).
After updating, phones set up automatically pick up the dial plan at their next daily check (or
reboot them).

- Admin guide: [Analog phones (ATAs)](docs/admin-guide/ata.md), 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.
- New internal numbers can no longer start with 911 or 9911, so 911 can go out at once (numbers
  you already have keep working).
- **Local AI** (admin guide: [Local AI](docs/admin-guide/ai.md)): 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).