Telephony

SIP Integration with Cisco Webex Calling

SIP trunking is only available on the Synthflow Enterprise Plan.

This guide covers connecting Cisco Webex Calling to Synthflow via a SIP trunk (Local Gateway). Webex routes calls to a Synthflow AI agent through the trunk, and Synthflow can transfer calls back to live Webex agents or Cisco CUCM queues.

Prerequisites

RequirementDetail
Cisco WebexActive Webex Calling organization with Admin access; Local Gateway (PSTN) feature enabled
SynthflowEnterprise Plan with a configured AI assistant (v2.0+)
Phone NumberAt least one DID provisioned in Webex Calling (E.164 format)
NetworkFirewall allows SIP (UDP/TCP 32681 or TLS 32682) and RTP media between Webex and Synthflow

Webex Calling Configuration

All configuration is done in the Webex Control Hub (admin.webex.com).

Create the SIP Trunk (Local Gateway)

  1. In Control Hub, navigate to Calling → PSTN → Manage.
  2. Select your location and click Add PSTN Connection.
  3. Choose Cloud-Connected PSTN or Premises-based PSTN depending on your setup.
  4. For a direct SIP trunk to Synthflow, select Local Gateway under Premises-based PSTN.
  5. Configure the trunk:
FieldValue
NameSynthflow AI
IP/FQDNsip.synthflow.ai
Port32681 (UDP/TCP) or 32682 (TLS)
TransportUDP/TCP or TLS
AuthenticationIP-based (whitelist Synthflow IPs) or Digest
UsernameFrom Synthflow — only required if using Digest auth
PasswordFrom Synthflow — only required if using Digest auth
  1. Click Save.

Webex Calling Local Gateway supports both IP-based and Digest authentication. For IP-based, add Synthflow’s signaling IPs to the allowed list (contact Synthflow support for the IP list). For Digest, enter the SIP credentials from the Synthflow Configuration section below.

Configure Inbound Routing (Caller → Webex → Synthflow)

To route inbound calls to Synthflow:

  1. In Control Hub, go to Calling → Call Routing → Auto Attendant (or Hunt Group).
  2. Create or edit the auto attendant/hunt group handling your inbound DID.
  3. Add a Transfer to External Number or Forward action and set the destination to the Synthflow-associated DID.
  4. Select the Synthflow AI Local Gateway trunk as the PSTN route for this destination.

For overflow or after-hours routing, use Webex Calling’s Business Hours and Holiday Schedule settings in the auto attendant to conditionally route to Synthflow.

Configure Outbound Routing (Webex → Synthflow / Synthflow → PSTN)

  1. In Control Hub, navigate to Calling → Outbound Calling → Dial Plans.
  2. Create a dial plan that matches the DID range used by Synthflow.
  3. Assign the Synthflow AI trunk as the route for this dial plan.
  4. Save and verify the dial plan is active.

Configure SIP Access Control

  1. Note Synthflow’s SIP signaling IP addresses. Contact Synthflow support for the current list.
  2. Add these IPs to your network firewall and to the Webex Local Gateway allowed signaling list.
  3. Ensure RTP media ports (8000–48000) are open bidirectionally between Webex media gateways and Synthflow.

Do not open all inbound SIP traffic. Whitelist only Synthflow’s signaling IPs to prevent unauthorized calls.

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 configured in Webex Calling).
  4. Fill in the SIP connection details:
FieldValue
Termination URIYour Webex Local Gateway FQDN or SBC IP address
SIP UsernameYour SIP authentication username
SIP PasswordYour SIP authentication password
Outbound Proxy (if required)Webex SBC proxy address (if required by your deployment)

The Termination URI is your Webex Local Gateway or SBC address — this is where Synthflow routes outbound calls into your Webex environment.

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 → Webex Calling → Auto Attendant → Local Gateway → Synthflow AI Agent
AI transfers to live agentSynthflow AI → SIP transfer → Local Gateway → Webex → Hunt Group → Live Agent

Verification

TestWhat to check
Inbound callCall the Webex DID from an external phone. Verify it routes through auto attendant to Synthflow AI. Check two-way audio.
Transfer to live agentTrigger a transfer during a Synthflow call. Confirm the call lands in the correct Webex hunt group with caller ID intact.
Control Hub trunk statusIn Webex Control Hub, verify the Local Gateway shows as active and registered.

Quick Troubleshooting

IssueFix
Registration failsVerify Termination URI is the Webex SBC/Local Gateway address. Check credentials and firewall (UDP/TCP 32681 or TLS 32682).
Calls not reaching SynthflowConfirm the Auto Attendant or Hunt Group transfer action points to the Synthflow DID via the correct trunk.
One-way or no audioEnsure RTP ports are open bidirectionally. Confirm G.711 (PCMU/PCMA) codec is negotiated.
Transfer failsVerify the transfer destination SIP URI is correct and reachable via the trunk.
Caller ID missingVerify the From header is set to the caller’s E.164 number. Check Webex Calling caller ID settings for the trunk.

Reference

  • Webex Calling Local Gateway Setup: help.webex.com (search “Local Gateway”)
  • Webex Control Hub Calling Configuration: admin.webex.com
  • Cisco SIP Trunk Interoperability: cisco.com/go/uc-interop
  • Synthflow SIP Overview: docs.synthflow.ai/about-sip
  • Synthflow X-EI SIP Header: docs.synthflow.ai/x-ei-sip-header