SIP Trunking
Connect your existing PBX or phone system to Voquii via native SIP trunk support. Route inbound calls from your current infrastructure directly to the AI voice pipeline without changing phone numbers or providers.
What is SIP Trunking?
SIP (Session Initiation Protocol) trunking allows you to connect your existing PBX, IP phone system, or call routing infrastructure directly to Voquii. Instead of routing calls through Twilio or Telnyx webhooks, your PBX sends calls to Voquii via a SIP trunk — keeping your existing phone numbers and call routing in place.
This is ideal for agencies whose clients already have established phone systems and want to add AI voice capabilities without disrupting their existing telephony setup.
When to Use SIP Trunking
Use SIP Trunking When:
- Your client has an existing PBX (Asterisk, FreeSWITCH, 3CX, etc.)
- You want to keep the current phone number and carrier
- You need custom call routing (e.g., AI handles overflow, after-hours, or specific menu options)
- You want to integrate AI voice into an existing IVR or call flow
Use Twilio/Telnyx Instead When:
- Starting fresh with a new phone number
- No existing PBX infrastructure to integrate with
- Want the fastest, simplest setup (5-minute Twilio/Telnyx webhook config)
See the Telephony Setup guide for Twilio/Telnyx BYOK configuration.
SIP Trunk Configuration
Get your SIP endpoint
In your Voquii dashboard, go to Phone → SIP Trunking. You'll receive a SIP URI and credentials for your trunk:
sip:agent-id@sip.voquii.comConfigure your PBX
In your PBX or phone system, create a new SIP trunk pointing to the Voquii SIP endpoint. Configure the trunk with the authentication credentials provided.
Route inbound calls
Set up your PBX dial plan to route calls to the SIP trunk. You can route all inbound calls, specific DID numbers, or overflow/after-hours calls.
Test the connection
Place a test call through your PBX to verify the SIP trunk is connected. You should hear the voice agent's greeting message within 375ms.
Supported Codecs
| Codec | Status | Notes |
|---|---|---|
| PCMU (G.711 mulaw) | Supported | Standard telephony codec, recommended |
| PCMA (G.711 alaw) | Supported | Common in European telephony systems |
Compatible PBX Systems
Asterisk / FreePBX
Add a SIP trunk in the Trunks module with the Voquii SIP URI and set up an inbound route.
FreeSWITCH
Configure a SIP gateway in your FreeSWITCH dial plan pointing to the Voquii endpoint.
3CX
Add a SIP trunk under SIP Trunks settings with the provided credentials and configure inbound rules.
Other SIP-Compatible Systems
Any system that supports standard SIP trunking can connect. Contact us for specific PBX setup help.