SIP Trunking (BYOC)
Already have a phone system or PBX? SIP Trunking lets you connect your existing setup to Akol without switching carriers or porting numbers. This is sometimes called BYOC — Bring Your Own Carrier.
What Is SIP Trunking?
SIP Trunking connects your existing phone system (PBX, call center software, etc.) directly to Akol. Instead of buying new numbers through Akol, you route calls from your current setup to your AI agent.
Why use it:
- Keep your existing phone numbers and carrier
- No need to port numbers or change providers
- Connect office PBX systems to your AI agent
- Route specific lines through Akol while keeping others on your current system
Creating a SIP Trunk
Go to Phone Numbers
Navigate to Dashboard > Phone Numbers and click the SIP Trunks button, then click Create SIP Trunk.
Configure Your Trunk
Fill in the details:
| Field | Description | Default |
|---|---|---|
| Trunk Name | A friendly name (e.g., “Office PBX”) | Required |
| Max Channels | How many simultaneous calls (1–1,000) | 10 |
| Codec | Audio encoding format | PCMU |
Save and Get Your Credentials
Click Create and you’ll receive three credentials:
- SIP Domain — The server address (e.g.,
sip.telnyx.com) - SIP Username — Your unique login
- SIP Password — Your authentication password
Save your password immediately. The SIP password is only displayed once when you create the trunk. If you lose it, you’ll need to delete the trunk and create a new one.
Supported Codecs
Choose the codec that matches your phone system:
| Codec | Best For | Quality |
|---|---|---|
| PCMU (G.711 µ-Law) | Most systems in North America — the safe default | Standard |
| PCMA (G.711 A-Law) | Systems in Europe, Asia, and Africa | Standard |
| G722 | HD voice — clearer audio if your system supports it | Wideband |
| Opus | Modern systems — best quality and efficiency | Excellent |
Not sure which to pick? PCMU works with almost everything.
Managing Your SIP Trunks
Editing Settings
Click on any trunk to open its detail page. You can update:
- Trunk name
- Max channels
- Codec
Click Edit, make your changes, and Save.
Verifying Connection
After configuring your PBX with the credentials, click Verify to test the connection. The status will update to:
| Status | Meaning |
|---|---|
| Active | Connected and ready to receive calls |
| Pending | Waiting for your PBX to register |
| Failed | Connection test failed — check your PBX settings |
Assigning Phone Numbers
Once your trunk is active, assign phone numbers to route through it:
- Open the trunk detail page
- Click Assign Number
- Select a phone number from the dropdown
- Calls to that number will now route through your SIP trunk
You can unassign numbers at any time to revert them back to standard routing.
Deleting a Trunk
Click Delete on the trunk detail page. This will:
- Remove the SIP connection
- Unassign all phone numbers (they’ll revert to standard routing)
- This action cannot be undone
Setting Up Your PBX
After creating a trunk in Akol, configure your PBX or phone system with the credentials you received:
- SIP Domain — Enter as the SIP server/registrar address
- SIP Username — Enter as the authentication username
- SIP Password — Enter as the authentication password
- Codec — Make sure your PBX uses the same codec you selected in Akol
The exact steps depend on your phone system. Check your PBX documentation for how to add a SIP trunk.
Need help connecting your specific phone system? Contact our support team and we’ll walk you through it.