selfphoneng/docs/admin-guide/troubleshooting.md

Troubleshooting

Run the commands on this page on the server, from the folder you cloned SelfPhoneNG into.

Looking inside Asterisk

Ask Asterisk directly:

docker compose exec asterisk asterisk -rx "pjsip show registrations"   # the trunk: Registered?
docker compose exec asterisk asterisk -rx "pjsip show endpoints"       # phones and trunks
docker compose exec asterisk asterisk -rx "pjsip show contacts"        # who is signed in, from where
docker compose exec asterisk asterisk -rx "core show channels"         # calls right now
docker compose exec asterisk asterisk -rx "queue show"                 # queues and agents

Seeing the SIP messages themselves

Open Asterisk's console and turn on the SIP logger:

docker compose exec asterisk asterisk -rvvv
pjsip set logger on

Make the failing call and read what scrolls by. To stop:

pjsip set logger off
exit

The logger prints passwords' hashes and phone numbers; don't paste it publicly without editing.

Logs

What	Where
Asterisk notices, warnings, errors	data/asterisk-logs/messages.log
Failed sign-ins (fail2ban reads this)	data/asterisk-logs/security.log
The api (web app, config changes, emails)	docker compose logs api
Everything, live	docker compose logs -f

One-way audio, or no audio

The call connects but one side (or both) hears nothing. It's almost always NAT (Network).

1. SIP ALG must be off on every router in the path. This is the first suspect.
2. Public IP: Settings → Network must show the server's current public IP. If it changed, update it and restart telephony (docker compose restart asterisk).
3. Audio ports: UDP 10000–20000 must be forwarded to the server and open on its firewall.
4. Local networks (setup wizard, Network step) must list your office networks, and only them.
5. Calls drop after about 30 seconds: same causes. The answer never reaches the other side, so it gives up.

The trunk won't register

Trunks shows Rejected or Not registered.

• Rejected: voip.ms said no.
  • The username must be the sub-account (123456_pbx) and the password its password, not your voip.ms portal password.
  • Retype the password on the trunk and save.
• Not registered / Unreachable: Asterisk gets no answer.
  • Check the server name (chicago1.voip.ms, no sip: or port).
  • Check the server can reach it: ping chicago1.voip.ms.
  • Check outbound UDP 5060 isn't blocked by a firewall.
  • If you limited SIP to voip.ms with nftables, the POP's address may have changed (Firewall).
• IP authentication: there's nothing to register.
  • The trunk shows Reachable when voip.ms answers.
  • The sub-account's allowed IP must be your server's current public IP.
• TLS:
  • TLS must be turned on in Settings → Security, with a valid certificate.
  • Encryption must also be on at the voip.ms sub-account.
  • Port 5061.

Then run pjsip show registrations, and use the SIP logger to see voip.ms's answer.

Incoming calls don't arrive

The trunk is Registered and outgoing calls work, but calls to your number fail or reach the wrong place.

1. At voip.ms (DID Numbers → Manage DIDs), the number must be routed to SIP/IAX → your sub-account, on the same POP the trunk uses. A different POP is the most common cause.
2. In SelfPhoneNG, the number must be on Phone numbers with a destination. Imported numbers hang up until you route them.
3. Office hours or a holiday may be sending the call elsewhere: check the number's schedule.
4. Still nothing? Run the SIP logger, call from your mobile, and check an INVITE arrives.
   • No INVITE: the problem is at voip.ms or the firewall.
   • An INVITE, then an error: see what Asterisk replied.

The browser phone won't connect

On My phone, the phone stays "connecting" or calls fail.

• Use the address with HTTPS and the real host name. Browsers block the microphone on plain HTTP or an IP address.
• Microphone permission: the browser must be allowed to use it. Look for the camera/mic icon in the address bar.
• The reverse proxy must pass /ws to 127.0.0.1:8088 with WebSocket upgrades.
  • The bundled Caddy and deploy/nginx/selfphoneng.conf.example both do.
  • On your own proxy, check: curl -i https://pbx.example.com/ws should answer 426 Upgrade Required with Server: Asterisk, not 404 or the web app.
• Certificate: the browser must trust the HTTPS certificate (no warning page).
• No audio on browser calls: see one-way audio. Browsers use the same audio ports.

Voicemail emails (or alerts) aren't sent

1. Settings → Email → Send test to your own address. The error shown says what the mail server answered.
2. Check the server, port and security setting match your provider:
   • 587 with STARTTLS;
   • or 465 with TLS from the start. Many providers need an app password.
3. The user's email address must be right, and voicemail-to-email on for them.
4. The api's log has every send error: docker compose logs api | grep -i mail.

"SIP port or transport changes need a telephony restart"

Changes to the public IP, SIP ports or TLS only take effect when Asterisk restarts. Run this when nobody is on a call:

docker compose restart asterisk

The banner goes away a few seconds after Asterisk is back.

Locked out

• Forgot the admin password: another admin can reset it on Users & Extensions.
• Lost the phone with the two-step codes: use a recovery code, or have another admin turn two-step sign-in off for you (Users & Extensions → the user → turn off two-step sign-in).
• The only admin is locked out: keep two admins so this can't happen. If it does, the api makes a new admin from .env when no admin is left:

  1. Make the locked-out account a plain user:

     docker compose exec postgres psql -U selfphoneng \
       -c "update users set role = 'user' where email = 'you@example.com'"
     

  2. In .env, set INITIAL_ADMIN_EMAIL to an email no account uses yet and INITIAL_ADMIN_PASSWORD to a temporary password.

  3. Run docker compose up -d --force-recreate api.

  4. Sign in with the new email; you're asked to change the password.

  5. On Users & Extensions, make your old account an admin again, reset its password (and two-step sign-in), sign in with it, and delete the temporary admin.