The SIP Debug page shows all inbound and outbound SIP messages
directly in the web UI – no SSH, no Asterisk CLI, and no external tools like
sngrep or Wireshark.
Features
- Toggle capture – Enable/disable recording with a switch
- Call list – All SIP dialogs grouped by Call-ID with timestamp, sender, receiver, and method
- SIP timeline – Click a call to view messages chronologically
- Direction arrows – → Sent (blue) and ← Received (green)
- Color coding – Methods blue, 2xx responses green, 4xx/5xx red, 1xx yellow
- Expandable raw SIP text – Full SIP headers and body in monospace
- Auto-refresh – Updates every 10 seconds when capture is active
Screenshots
SIP messages grouped by Call-ID with expandable raw SIP text
How does it work technically?
SIP Debug uses Asterisk’s built-in res_pjsip_history module.
When enabled, SIP history is activated in Asterisk.
The backend polls every 3 seconds via AMI (Asterisk Manager Interface)
for new entries and loads the full SIP text for each one.
Messages are kept in memory – no database, no log files:
- Max 2 hours retention – older messages are removed automatically
- Hard cap: 10,000 messages – prevents memory overflow under high traffic
- No disk I/O – everything in RAM, no performance impact on the PBX
Typical use cases
| Problem | What to check |
|---|---|
| Trunk does not register | REGISTER → 401/403 responses, verify credentials |
| Inbound call does not ring | INVITE → 100 Trying / 180 Ringing sequence |
| One-way audio | SDP body in INVITE/200 OK (media addresses, NAT) |
| Wrong caller ID | From header and P-Asserted-Identity in outbound INVITE |
| Call drops after 30s | Missing ACK or session-timer headers |
Access
The SIP Debug page is available via the navigation in the GonoPBX web UI (bug icon between FAQ and Settings). It is admin-only – users with the “User” role will not see the menu item.
Available since version 1.7.0.
Update via Settings → Server → Update or manually with
git pull && docker compose up -d --build.