Telephony

SIP Integration with Dixa

SIP trunking is only available on the Synthflow Enterprise Plan.

This guide covers connecting Dixa to Synthflow via SIP trunk so that Dixa can route calls to a Synthflow AI agent. Synthflow handles AI-driven conversations and transfers calls back to live Dixa agents when escalation is needed.

Prerequisites

RequirementDetail
DixaActive account with Admin access; telephony and SIP forwarding enabled
SynthflowEnterprise Plan with a configured AI assistant (v2.0+)
Phone NumberAt least one DID provisioned in Dixa (E.164 format)
NetworkFirewall allows SIP (UDP/TCP 32681 or TLS 32682) and RTP media between Dixa and Synthflow

Dixa Configuration

All configuration is done in the Dixa Admin panel (app.dixa.com) under Settings → Phone.

Configure SIP Forwarding to Synthflow

Dixa supports forwarding calls to external SIP destinations through its Flow Builder.

  1. In Dixa, go to Settings → Phone → SIP Settings.
  2. Note your Dixa SIP domain — it is in the format {yourcompany}.sip.dixa.com.
  3. Add Synthflow’s SIP signaling IPs to the IP Allowlist. Contact Synthflow support for the current IP list.

Dixa uses IP-based access control for inbound SIP. Synthflow’s signaling IPs must be allowlisted before calls can be received.

Create a Phone Flow to Route Calls to Synthflow

  1. In Dixa, navigate to Flows → Phone Flows.
  2. Create a new flow or edit the flow assigned to your inbound DID.
  3. Add a Transfer step and configure:
FieldValue
Transfer TypeSIP Address
SIP Address{synthflow_did}@sip.synthflow.ai:32681
UUI HeaderInclude phone_number={caller_e164} (required for caller identification)
  1. Set a timeout (recommended: 30 seconds) with a fallback to an agent queue if Synthflow does not answer.
  2. Save and activate the flow.

The UUI (User-to-User Information) header passes caller context to Synthflow. Use the phone_number key with the caller’s E.164 number as the value.

Configure Multi-Department Routing (Optional)

For organizations routing different call types to different Synthflow agents:

  1. Create separate Dixa phone flows per department.
  2. Assign different Synthflow DIDs per flow — one DID per Synthflow assistant.
  3. Each DID maps to a separate SIP number and assistant in Synthflow.

Configure Transfer Back to Dixa (AI → Live Agent)

To enable Synthflow to escalate a call to a Dixa agent queue:

  1. Note your Dixa SIP domain (e.g. support.sip.dixa.com) and the destination DID for the agent queue.
  2. Configure Synthflow’s transfer action to route to the SIP URI: {queue_did}@{yourcompany}.sip.dixa.com.
  3. Ensure the destination DID is mapped to an agent queue or ring group in Dixa.

Use SIP URI transfer only — do not transfer to a plain DID (telephone number). Dixa routes agent escalations via SIP address, not PSTN.

Synthflow Configuration

Add the SIP Number

  1. Go to the Synthflow Phone Numbers page.
  2. Click Add Number → Custom / SIP.
  3. Enter the phone number in E.164 format (same DID provisioned in Dixa).
  4. Fill in the SIP connection details:
FieldValue
Termination URIYour Dixa SIP domain (e.g. {yourcompany}.sip.dixa.com)
SIP UsernameYour Dixa SIP authentication username (if credential auth is used)
SIP PasswordYour Dixa SIP authentication password (if credential auth is used)
Outbound Proxy (if required)Leave blank unless Dixa specifies a proxy

Dixa primarily uses IP-based authentication. If credential-based auth is not required, leave SIP Username/Password blank and rely on IP allowlisting.

Be precise when copying SIP credentials. Incorrect values will cause registration failure.

Attach to the Assistant

  1. Go to Assistants and create or select an assistant (v2.0+).
  2. In the assistant settings, attach the SIP number you just created.
  3. Wait 1–2 minutes for SIP registration to complete.

The SIP trunk is created at the moment of attaching the SIP number to the assistant.

Call Flow Summary

ScenarioFlow
Inbound to AICaller → PSTN → Dixa → Phone Flow → SIP Transfer → Synthflow AI Agent
AI transfers to live agentSynthflow AI → SIP URI transfer → Dixa SIP domain → Agent Queue → Live Agent
Overflow / after-hoursDixa Flow (condition check) → Synthflow AI → handles call

Verification

TestWhat to check
Inbound callCall the Dixa DID from an external phone. Verify the Phone Flow routes to Synthflow AI. Check two-way audio.
Transfer to live agentTrigger escalation during a Synthflow call. Confirm the call lands in the correct Dixa queue with caller ID intact.
UUI header passthroughVerify Synthflow receives the phone_number UUI header with the caller’s E.164 number.

Quick Troubleshooting

IssueFix
Registration failsVerify Termination URI is the correct Dixa SIP domain. Confirm Synthflow IPs are in the Dixa allowlist.
Calls not reaching SynthflowCheck the Phone Flow transfer step is correctly configured with the Synthflow SIP address.
One-way or no audioEnsure RTP ports are open bidirectionally. Confirm G.711 codec is negotiated on both sides.
Missing caller contextVerify the UUI header is configured in the Dixa flow transfer step with the phone_number key.
Transfer back failsConfirm the Dixa SIP domain and agent queue DID are correct in the Synthflow transfer destination. Use SIP URI, not plain DID.

Reference

  • Dixa Phone Flow Documentation: help.dixa.com (search “Phone Flows”)
  • Dixa SIP Settings: help.dixa.com (search “SIP”)
  • Synthflow SIP Overview: docs.synthflow.ai/about-sip
  • Synthflow X-EI SIP Header: docs.synthflow.ai/x-ei-sip-header