LaunchTelephony

Call Transfer to SIP

Overview

Call Transfer to SIP enables your Synthflow AI agents to hand off live calls to SIP endpoints such as PBX systems, SIP phones, or softphones. It bridges the gap between automation and human support — giving you the flexibility to route calls from AI to live agents when escalation is required.

Why It Matters

  • Seamless Escalation – Effortlessly move callers from AI to a human agent.
  • Flexible Integration – Works with any SIP-compatible PBX, trunk, or softphone.
  • Context Preservation – Optionally send a conversation summary before transfer.
  • Professional Experience – Provide warm, human-like handoffs.
  • Scalable and Reliable – Supports enterprise-grade routing and high call volumes.

Setup

Prerequisites

Before enabling Call Transfer to SIP, ensure:

  • Your SIP endpoint (PBX, SIP server, or softphone) is reachable from Synthflow.
  • Supported audio codecs are enabled: G.711, G.722, or Opus.
  • Proper network routing and firewall access are configured.

Step 1. Configure Your SIP Endpoint

1

Allow inbound SIP calls from Synthflow’s IPs or proxy.

2

Enable SIP authentication if required.

3

Open and forward SIP (UDP/TCP 5060) and RTP port ranges.

4

Confirm audio codec compatibility (G.711, G.722, or Opus).

Step 2. Add a Call Transfer Action in Synthflow

1

Go to Agents → [Your Agent] → Actions.

2

Under During the call, click Add Action → Call Transfer.

3

Enter your SIP URI (for example, sip:user@domain.com).

4

Click Add Action to save.

Step 3. Configure Transfer Settings

  • Transfer Mode
    • Warm transfer with message – Play a short message before connecting.
    • Warm transfer with context summary – Send a brief call summary.
    • Blind transfer – Instantly connect without preamble.
  • Timeout – How long Synthflow waits for the SIP endpoint to answer.
  • Optional: Configure pre/post message pauses for better timing control.

Usage

How It Works

  1. The caller requests to speak with a human.
  2. The AI agent detects the intent (e.g., “transfer me to an agent”).
  3. Synthflow sends a SIP INVITE to your configured SIP URI.
  4. Once the SIP endpoint answers, the caller is bridged directly.

Supported SIP Endpoints

  • On-premise or cloud PBX (Asterisk, Kamailio, FreePBX, etc.)
  • SIP phones and softphones
  • SIP trunks and SIP service providers
  • SIP-enabled mobile or web apps

Supported SIP URI Formats

  • sip:user@domain.com
  • sip:user@domain.com:5060
  • sips:user@domain.com (TLS)
  • sip:+1234567890@provider.com
  • sip:1001@pbx.company.com

Testing Call Transfer to SIP

1

Go to Agents → [Your Agent] → Test Agent.

2

Start a test call.

3

Ask for a transfer, e.g.:

  • “I’d like to speak to a human.”
  • “Can you transfer me to support?”
4

Confirm the call transfers and audio quality are good.

Troubleshooting

Debug Tips

  • Review call logs in the Synthflow Dashboard.
  • Enable SIP tracing on your PBX or SIP server.
  • Verify network latency and packet loss.
  • Confirm audio codec negotiation in SIP INVITE/200 OK.

FAQs

What SIP systems are supported?

Any SIP-compatible endpoint — including PBX systems, SIP phones, softphones, or SIP trunks.

What’s the difference between warm and blind transfer?

Warm transfer includes a pre-message or context summary; blind transfer connects instantly without any introduction.

Can I transfer to multiple destinations?

Yes. Create multiple transfer actions for different departments or agents, and route them through Flow Designer.

How do I test transfers?

Use the Test Agent feature in Synthflow. Request a transfer during a test call to confirm setup and audio quality.

What should I check if transfers fail?

Confirm SIP endpoint reachability, authentication details, and open ports. Review Synthflow call logs and your SIP traces for details.