An ATA adapter lets an analog telephone use a SIP service. The HT801 V2 has one FXS phone port; the HT802 V2 has two. This guide uses the V2 administration interface. Check the label on your adapter before following the menus; first-generation HT801/HT802 devices use a different layout.
Prepare the IllyVoIP line
Open SIP Lines and obtain the SIP username, SIP password and server for the line you intend to connect. These are not your Portal email and login password. This walkthrough uses password authentication. If the existing line uses IP authentication, do not switch modes without checking which devices depend on it.
A separate device can use a separate SIP line; review any subscription charge before confirming. Keep a note of which device uses each line, but do not put passwords in shared documents. When replacing a working phone, keep its configuration privately so you have a way back if the new setup needs troubleshooting.

Connect the cables before entering credentials
Connect the network port to your trusted router or switch and the analog telephone to the adapter’s FXS phone port. Use the correct power supply. Do not connect an FXS port to an active public telephone line or unknown building telephone wiring. Start with one directly connected telephone so a cabling problem cannot be confused with a SIP problem.
Configure the port that the phone actually uses
Find the adapter’s local IP using its network information or the router’s device list, then open its administration page. On V2, go to Port Settings → FXS PORT x → General Settings → Account Registration. Enable Account Active and enter the values below. On an HT802, check whether the cable is in port 1 or port 2 before editing.
| Field | What to enter |
|---|---|
| Primary SIP Server | IllyVoIP SIP server for the line |
| SIP User ID | SIP username |
| Authenticate ID | SIP authentication ID |
| Authenticate Password | SIP password |
| Name | A local label for this port |
Save and apply. Follow any restart prompt only when the adapter is not carrying a call. Check Status → Port Status → Registration for the configured port. A password field can be displayed blank after saving for security; do not repeatedly overwrite it if registration already succeeds.
A dial tone is not the complete test
Lift the receiver and check the physical phone connection, but verify SIP registration and an actual call separately. If no dial tone is heard, check power, cable and the selected physical port first. If you hear tone but no call reaches IllyVoIP, check the adapter’s digit collection/dial plan; it may be waiting for more digits rather than sending the call. Do not replace the whole dial plan with an Internet example without checking the formats you need.
Using the second port
Treat port 2 as a separate configuration decision. Do not assume copying port 1 gives both phones independent incoming routing or extra simultaneous-call capacity. If they need different behavior, use the appropriate separate line and configure the purchased number’s destination accordingly. Test each port separately before handing the adapter to staff.
Ringing, telephone menus and specialist devices
If registration and outgoing calls work but the analog phone does not ring, check the number’s destination, the FXS port used, Do Not Disturb and the handset itself. If a telephone menu cannot hear key presses, report that specific DTMF symptom rather than changing authentication settings. Fax machines, alarm panels and emergency equipment need separate compatibility checks; successful voice calling is not proof they will work safely through the adapter.
Match signaling and audio settings
Use the server, destination port and transport supplied for your line. The device’s local IP opens its administration page; it is not the IllyVoIP server address. SIP signaling and audio encryption are separate choices. The Portal audio mode must match the device’s supported configuration: Standard RTP, SDES-SRTP or DTLS-SRTP. Do not change encryption just because an unrelated tutorial uses another setting.
Keep the administration page on a trusted network. Do not expose it to the Internet or disable a firewall to make a test work. If your model does not offer a required option, ask about that exact model and firmware instead of changing several unrelated settings at once.
Verify the result one step at a time
- Registration: check the device’s account status. A familiar name on its screen is not proof that it is registered.
- Outgoing call: select the IllyVoIP line and call a number you control in international format. Check balance and destination rates first; the test may be chargeable.
- Audio and identity: speak in both directions and check the outgoing caller ID. Editing a local display name does not grant permission to use another number.
- Incoming call: route an active phone number to this SIP line and call it from another device. SIP registration does not automatically choose a number’s incoming destination.
What to check before contacting Support
If registration fails, check the line details before changing the network. If registration succeeds but a call is rejected, check number format, balance, routing package and permitted caller ID. If the call connects without audio, report that distinction: it is a different problem from failure to register.
Include the device model, firmware version, time and timezone, visible error and public call reference where available. The SIP response-code guide helps explain errors. Never send passwords, unmasked configuration exports or screenshots containing customer data.
Documentation and scope
Device configuration reference. Menu paths were checked against documentation and Portal steps against IllyVoIP. This is not physical-device certification; the Portal screenshot is not a screenshot of the phone administration interface.