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Documentation Index

Fetch the complete documentation index at: https://documentation.uponai.com/llms.txt

Use this file to discover all available pages before exploring further.

Connect UponAI agents with your own telephony provider and phone numbers. This works independently of the agent type you’re using.

Integration Methods

Elastic SIP Trunking (Recommended)

Connect via SIP trunk. Supports all telephony features available with UponAI numbers. Works with most major telephony providers.

Dial to SIP URI

Dial calls directly to a UponAI SIP URI. Use this if your provider doesn’t support elastic SIP trunking or you have complex routing needs.

UponAI SIP Server Details

SettingValue
SIP Server URIsip:sip.retellai.com
IP Block (all regions)18.98.16.120/30
IP Block (US traffic)143.223.88.0/21, 161.115.160.0/19
Transport methods supported:
TransportSIP URL
TCP (Recommended)sip:sip.retellai.com;transport=tcp
UDPsip:sip.retellai.com;transport=udp
TLSsip:sip.retellai.com;transport=tls
mTLSContact support
Audio codecs: PCMU, PCMA, G.722 (HD) Media encryption: SRTP (requires TLS transport)
Elastic SIP Trunking connects UponAI’s VoIP infrastructure with the PSTN so your agents can make and receive calls. All telephony features supported by UponAI numbers are also supported here (assuming your provider supports them). Supported providers with detailed guides:

Twilio

Connect via Twilio SIP trunking

Telnyx

Connect via Telnyx SIP trunking

Vonage

Connect via Vonage SIP trunking

Amazon Connect

Connect via Amazon Connect

Avaya

Connect via Avaya

Five9

Connect via Five9

Genesys

Connect via Genesys
Other providers that support SIP trunks are also compatible — use the guides above as reference.

Method 2: Dial to SIP URI

Use this method if your provider doesn’t support elastic SIP trunking, or you have complex routing requirements. With this method, UponAI doesn’t directly make or receive calls — your system dials the call to a specific SIP URI. All traffic appears as inbound to UponAI, so you specify the call direction.
When using Dial to SIP URI, you cannot use UponAI’s built-in call transfer feature. You must implement your own transfer logic using a custom function.
1

Register the phone call

Call the Register Phone Call API to get a call_id and determine which agent handles the call.
2

Construct the SIP URI

Build the SIP URI using the call_id:
sip:{call_id}@sip.retellai.com
3

Dial the call

Dial the call to that SIP URI within 5 minutes of calling Register Phone Call. If not connected in time, the call disconnects with registered_call_timeout.
Example — handling a Twilio inbound webhook:
Node
const client = new UponAI({
  apiKey: 'YOUR_API_KEY',
});

server.app.post("/voice-webhook", async (req, res) => {
  const phoneCallResponse = await client.call.registerPhoneCall({
    agent_id: 'oBeDLoLOeuAbiuaMFXRtDOLriTJ5tSxD',
    from_number: "+12137771234",
    to_number: "+12137771235",
    direction: "inbound",
  });

  const voiceResponse = new VoiceResponse();
  const dial = voiceResponse.dial();
  dial.sip(`sip:${phoneCallResponse.call_id}@sip.retellai.com`);
  res.set("Content-Type", "text/xml");
  res.send(voiceResponse.toString());
});

Security

UponAI uses enterprise-grade security for all SIP communications. We recommend TLS 1.2 or higher for all SIP signaling.

Mutual TLS (mTLS)

mTLS requires both client and server to present certificates — providing stronger authentication than standard TLS and protecting against man-in-the-middle attacks. UponAI uses a client certificate issued by AWS Private Certificate Authority (PCA). To validate UponAI’s TLS client certificate, add the root certificate to your SIP server’s trusted certificate store. To enable mTLS for your account, contact support@uponai.com.

Telephony Partners

For custom SIP call flow manipulation or needs not directly supported by UponAI:
  • Jambonz — SIP server with static IP support. Can connect to UponAI’s SIP server for call flow manipulation. Dedicated support channel available for UponAI users.
  • Cloudonix — CPaaS that connects to UponAI’s SIP server for call flow manipulation.