Troubleshooting SIP Registration

Registration Fails (401/403)

A 401 Unauthorized or 403 Forbidden response means the SIP server rejected your credentials.

Checklist

1. Verify credentials. Go to your AIVO dashboard > AIVO Connect > SIP Trunking > your connection. Copy-paste the username and password directly into your PBX. Do not retype manually (avoids typos).
2. Check the SIP server address. It should be sip.aivo.bz (or a regional endpoint). Common mistake: using https:// prefix or adding a port when it is not needed.
3. Check the auth type. If you chose IP authentication, your PBX's public IP must match what you entered in AIVO. Run curl ifconfig.me from your PBX server to check.
4. Check for password special characters. Some PBX platforms have trouble with certain characters in passwords. If your password contains #, &, or @, try regenerating a new password from the AIVO dashboard.
5. Check registration method. Some PBX systems default to peer-based auth. AIVO requires standard digest authentication.

Still Failing?

• Check your PBX's SIP debug log for the exact error message.
• In FreePBX: asterisk -rvvv then pjsip set logger on.
• In 3CX: Check the event log under Dashboard > System Status.
• Send the relevant log snippet to hello@aivo.bz for help.

Calls Not Routing

Your trunk is registered but calls do not go through.

Inbound Calls Not Arriving

1. Check DID mapping. In AIVO, verify the phone number is assigned to the correct connection. In your PBX, verify the inbound route matches the DID.
2. Check the format. Some PBX systems expect DIDs in different formats (e.g., 15551234567 vs +15551234567 vs 5551234567). Try all variations.
3. Check your PBX is reachable. If using IP auth, verify your public IP has not changed. If using credential auth, verify the trunk still shows as registered.
4. Check call routing logs. In your PBX, look at the inbound call log to see if the call arrives but gets rejected or misrouted.

Outbound Calls Not Connecting

1. Check outbound routes. Verify your PBX has an outbound route that uses the AIVO trunk.
2. Check dial patterns. Ensure the pattern matches the number format you are dialing. E.164 format (+1XXXXXXXXXX) is recommended.
3. Check channel limits. If you have reached your maximum simultaneous calls, additional calls will fail. Check your plan's channel limit.
4. Listen for error tones. A fast busy signal usually means a routing error. A normal ring followed by nothing suggests a remote-side issue.

One-Way Audio

You can hear the remote party but they cannot hear you (or vice versa). This is the most common SIP issue.

Root Cause: NAT

In almost all cases, one-way audio is caused by incorrect NAT (Network Address Translation) configuration.

Fix for FreePBX

1. Go to Settings > Asterisk SIP Settings.
2. Under NAT Settings:

• External Address: Enter your public IP (or hostname that resolves to it).
• Local Networks: Enter your LAN subnet (e.g., 192.168.1.0/24).

1. In the trunk's Advanced settings:

• Enable Force rport.
• Enable Rewrite Contact.

1. Apply config and test.

Fix for 3CX

1. Go to Dashboard > Firewall.
2. Run the Firewall Checker to verify ports are open.
3. In the SIP Trunk settings, ensure Outbound Proxy is set to sip.aivo.bz.

General Fixes

1. Open RTP ports (10000-20000 UDP) on your firewall.
2. Disable SIP ALG on your router.
3. If behind double NAT (router + ISP router), configure port forwarding on both devices.

DTMF Not Working

Callers press keys but the system does not detect them.

Check DTMF Mode

AIVO Connect uses RFC 4733 (also called RFC 2833) for DTMF. Ensure your PBX is set to the same mode.

Other Causes

• Inband DTMF: Some PBXs send DTMF as audio tones instead of SIP events. AIVO does not support inband DTMF reliably. Switch to RFC 4733.
• SIP INFO DTMF: Less common but sometimes used. Switch to RFC 4733 for compatibility.
• Codec issues: DTMF tones can be mangled by low-bitrate codecs. Use G.711 or G.722.

Codec Negotiation Issues

If calls fail during setup with a 488 Not Acceptable Here or similar error, the PBX and AIVO could not agree on a codec.

Supported Codecs

Ensure your PBX offers at least one of these:

Codecs to Disable

These codecs are not supported by AIVO Connect:

• G.729 (requires licensing)
• iLBC
• GSM
• Speex

Disable them in your PBX to prevent negotiation conflicts.

Checking Codec Negotiation

In your PBX's SIP debug log, look for the SDP (Session Description Protocol) section in the INVITE and 200 OK messages. The m=audio line lists offered codecs. If none match AIVO's supported codecs, the call will fail.