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Difficulty: Easy · ~10 min

Yeastar SIP Trunk Setup

Connect your Yeastar S-Series or P-Series PBX to AIVO Connect for reliable SIP trunking. This guide covers the web admin panel configuration for both credential and IP authentication.

Prerequisites

  • Yeastar S-Series (S20/S50/S100/S300) or P-Series PBX with latest firmware
  • An active AIVO Connect SIP trunking account with credentials
  • Admin access to the Yeastar web admin panel
  • Your AIVO Connect SIP username and password (from the Connect dashboard)

AIVO Connect SIP Settings

SIP Configuration Reference

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

1

Navigate to VoIP Trunks

Log in to your Yeastar web admin panel. The URL is typically https://<PBX-IP>:8088.

S-Series: Go to Settings > PBX > Trunks.

P-Series: Go to Extension and Trunk > Trunk.

Click Add and select VoIP Trunk.

2

Enter SIP Server Details

In the trunk creation form, select Register Trunk as the trunk type (for credential auth) or Peer Trunk (for IP auth). Then fill in the following:

Trunk Name: AIVO-Connect

Hostname/IP: sip.aivo.bz

Domain: sip.aivo.bz

Port: 5060

Transport: UDP (or TLS with port 5061)

3

Configure Authentication

Set up authentication based on your chosen method:

Register Trunk (Credential Auth)

Username/Auth ID: (your AIVO Connect SIP username)

Password: (your AIVO Connect SIP password)

The PBX will register with the SIP server automatically.

Peer Trunk (IP Auth)

Leave username and password fields empty. Under your AIVO Connect dashboard, add your PBX's public IP address to the IP Authentication whitelist. No registration is needed.

4

Set Codec and DTMF Settings

Scroll down to the Codec section of the trunk settings. Select and order codecs as follows:

1. G.722 (wideband, recommended)

2. G.711u (ulaw) (narrowband, universal)

3. G.711a (alaw) (narrowband, international)

4. Opus (wideband, if supported by your firmware)

Move enabled codecs to the Selected column using the arrow buttons, and arrange them in the priority order shown above.

Set DTMF Mode to RFC 2833.

5

Create Outbound Route

S-Series: Navigate to Settings > PBX > Call Control > Outbound Routes.

P-Series: Navigate to Call Control > Outbound Route.

Click Add and configure the route:

Route Name: AIVO-Outbound

Dial Pattern: X. (matches any number; refine as needed)

Member Trunks: AIVO-Connect

Member Extensions: All Extensions (or specific extension groups)

For inbound calls, Yeastar will automatically match incoming DID numbers. You can configure inbound routes under Inbound Routes to direct calls to specific extensions, ring groups, or IVR menus.

6

Apply and Save

Click Save on both the trunk and outbound route configurations.

Yeastar will prompt you to Apply Changes — click the apply button to activate the new settings.

Once applied, return to the trunk list. For a Register Trunk, you should see the status change to Registered. For a Peer Trunk, the status will show Available.

Test Your Connection

After applying the configuration, verify your setup:

  1. Check trunk status — In the Yeastar admin panel, go to the trunk list and confirm the AIVO-Connect trunk shows as Registered or Available.
  2. Place a test outbound call — Pick up a phone connected to your PBX, dial an external number, and confirm two-way audio works.
  3. Test an inbound call — Call one of your AIVO Connect DID numbers from a mobile or external phone. Verify the call reaches the correct destination.
  4. Check call logs — Go to Logs > Call Logs in the Yeastar panel and confirm both outbound and inbound test calls appear with correct status and duration.
  5. Verify DTMF — During a test call, press keypad digits to confirm they are transmitted correctly (test with voicemail or an IVR).

Troubleshooting

Trunk shows “Unavailable”: Double-check the hostname, port, and credentials. Verify your firewall allows outbound UDP/TCP on port 5060 (or 5061 for TLS).

One-way audio: This is typically a NAT issue. In the Yeastar admin, go to Settings > PBX > General > SIP > NAT and configure your External IP and Local Network. Ensure RTP ports (10000–20000 UDP) are open on your firewall.

Calls drop after 30 seconds: Check that your router's SIP ALG is disabled. SIP ALGs often modify SIP headers and cause session timeouts.

No inbound calls: Verify your DID numbers are correctly assigned in the AIVO Connect dashboard and that an inbound route exists on the Yeastar PBX to handle the incoming DID.

Need help? Contact our team at contact-sales or check the SIP network information page for full technical details.