3CX SIP Trunk Setup
Connect your 3CX phone system to AIVO Connect for reliable inbound and outbound calling. This guide covers both credential and IP-based authentication.
Prerequisites
- 3CX v18 or v20 installed and accessible via the Management Console
- An active AIVO Connect SIP trunking account with credentials
- Your AIVO Connect SIP username and password (from the Connect dashboard)
- Admin access to the 3CX Management Console
AIVO Connect SIP Settings
SIP Server: sip.aivo.bz
Port: 5060 (UDP/TCP) or 5061 (TLS)
Codecs: G.722, G.711u, G.711a, Opus
DTMF: RFC 2833
Registration: Not required (IP auth) / Required (credential auth)
Auth Methods: Credential (username/password) or IP-based
Add a New SIP Trunk
Open the 3CX Management Console and navigate to SIP Trunks in the left sidebar.
Click Add SIP Trunk. When prompted to select a provider, choose Generic SIP Trunk since we will configure the settings manually.
Give the trunk a descriptive name such as AIVO-Connect.
Enter SIP Server & Port
In the trunk configuration screen, enter the following values:
Registrar/Server/Gateway Hostname: sip.aivo.bz
Outbound Proxy: sip.aivo.bz
Port: 5060
Number of SIM Calls: (set to your channel limit)
If your network supports TLS, use port 5061 and set the transport to TLS for encrypted signaling.
Configure Authentication
Under the Authentication tab, select your preferred method:
Option A: Credential Authentication
Authentication ID: (your AIVO Connect SIP username)
Authentication Password: (your AIVO Connect SIP password)
Registration will be enabled automatically when credentials are provided.
Option B: IP-Based Authentication
Leave the authentication fields blank. Ensure your public IP address is whitelisted in your AIVO Connect dashboard under SIP Trunking > IP Authentication. No registration is required.
Set Codec Priority
Navigate to the Codecs tab within the trunk settings. Set the codec priority order as follows:
1. G.722 (wideband, recommended)
2. G.711u (PCMU) (narrowband, universal)
3. G.711a (PCMA) (narrowband, international)
4. Opus (wideband, if supported by your 3CX version)
Under DTMF mode, ensure RFC 2833 is selected. This guarantees reliable keypad input during calls.
Set Outbound Rules
Go to Outbound Rules in the left sidebar and click Add New.
Create a rule that routes calls through the AIVO Connect trunk:
Rule Name: AIVO-Outbound
Calls to numbers starting with prefix: 0 (or leave blank for all calls)
Calls from extension(s): All Extensions
Route 1: AIVO-Connect (select the trunk you created)
For inbound calls, 3CX will automatically receive calls on any DID numbers assigned to your trunk. You can configure inbound routing under Inbound Rules to direct calls to specific extensions, ring groups, or IVR menus.
Save and Apply
Click OK to save the trunk configuration and outbound rule. 3CX will apply the settings immediately.
If using credential authentication, check the trunk status in the SIP Trunks list. You should see a green status indicator confirming successful registration.
Test Your Connection
Once the trunk is configured and registered, verify your setup:
- Check trunk status — In the 3CX Management Console, go to SIP Trunks and confirm the trunk shows as Registered (green).
- Place a test outbound call — From any extension, dial an external number. Confirm you hear ringing and can establish a two-way audio call.
- Test an inbound call — Call one of your AIVO Connect DID numbers from a mobile or external phone. Verify the call reaches the correct extension or ring group.
- Verify audio quality — During the test call, check for clear audio in both directions. If you hear choppy audio, verify your firewall allows UDP traffic on ports 10000–20000 for RTP media.
- Test DTMF — During a call, press keypad digits and confirm they are transmitted correctly (useful for IVR menus and voicemail access).
Troubleshooting
Trunk shows as “Unregistered”: Double-check your SIP username and password. Ensure your firewall allows outbound traffic on port 5060 (or 5061 for TLS).
One-way audio: This is typically a NAT/firewall issue. Ensure RTP ports (10000–20000 UDP) are open. If your 3CX is behind NAT, configure the STUN server in 3CX settings.
Calls drop after 30 seconds: Check that SIP re-INVITE is not being blocked by your firewall. Some SIP ALGs on routers interfere with signaling — try disabling the SIP ALG.
Need help? Contact our team at contact-sales or check the SIP network information page for full technical details.