Transferring Calls

Transfer calls from Glassix to your telephone number

Overview

After your AI agent handles a call, you can transfer the caller to your internal team, another phone number, or any SIP destination. There are two types of transfers:

  1. PSTN Transfers - Transfer to regular phone numbers (mobile, landline, etc.)
  2. SIP Transfers - Transfer to SIP addresses (internal PBX extensions, SIP endpoints, etc.)

Transfers to PSTN Phone Numbers

Transfer calls to any real phone number through your telephony provider.

Prerequisites

  • At least one phone number configured with credential-based authentication
  • The number must be verified and active with your provider
  • Your provider must allow outbound/termination calling on the trunk
  • Caller ID requirement: You must specify one of your registered phone numbers as the caller ID

How PSTN Transfers Work

  1. A user calls your Glassix number (e.g., +1-555-0100)
  2. Glassix AI agent handles the call
  3. Based on your voice flow logic, our AI Agent decides to transfer
  4. Glassix sends a SIP INVITE to your provider to dial the destination number
  5. You must specify which registered number to use as caller ID (e.g., +1-555-0100 or +1-555-0200)
  6. Your provider validates the caller ID is one of your authorized numbers
  7. The provider completes the call to the destination
  8. Both parties are connected

Configuring PSTN Transfers in Voice Flows

  1. Navigate to SettingsChatbot and AIChatbot Flows Editor → Your Flow.

  2. Add an AI Agent Card.

  3. Add a Transfer to External Voice Agent action:

  4. Configure:

    • Transfer Type: Phone Number (PSTN)
    • Destination Number: +15551234567 (E.164 format recommended)
    • Caller ID: Select from your registered numbers (required)
  5. Add fallback actions if transfer fails

Important Notes for PSTN Transfers

  • Caller ID is mandatory: You cannot transfer without specifying a valid, registered caller ID
  • Use E.164 format: Always use international format (e.g., +15551234567)
  • Verify number ownership: The caller ID must be a number you own and have configured in Glassix.
  • Provider restrictions: Some providers require explicit authorization for caller ID numbers
  • Costs apply: PSTN transfers incur per-minute charges from your provider

Transfer to SIP Addresses

Transfer calls directly to SIP endpoints (PBX extensions, conference bridges, voicemail systems, etc.).

Prerequisites

  • A SIP endpoint or PBX that can receive calls
  • Glassix IP addresses whitelisted on your SIP server (see IP Whitelisting section above)
  • Your SIP server must be reachable from the internet (or use VPN/private connection)
  • Optionally: SIP authentication credentials if your server requires them

How SIP Transfers Work

  1. A user calls your Glassix number
  2. Glassix AI agent handles the call
  3. Based on your voice flow, Glassix transfers to a SIP address
  4. Glassix sends a SIP INVITE directly to your SIP URI (e.g., sip:[email protected])
  5. Your SIP server validates the source IP (must be whitelisted)
  6. Your SIP server routes the call to the destination extension/endpoint
  7. Both parties are connected

Configuring SIP Transfers in Voice Flows

  1. Navigate to SettingsChatbot and AI → ** Chatbot Flows Editor** → Your Flow.

  2. Add an AI Agent Card.

  3. Add a Transfer to External Voice Agent action:

  4. Configure:

  5. Add fallback actions if transfer fails

Important Notes for SIP Transfers

  • IP whitelisting is mandatory: Your SIP server will reject calls from non-whitelisted IPs
  • Direct connection: Calls go directly to your SIP infrastructure, not through your provider
  • No provider costs: SIP-to-SIP transfers don't incur provider charges
  • Firewall configuration: Ensure SIP (5060/5061) and RTP (16384-32768 UDP) ports are accessible
  • Domain/IP: You can use domain names or IP addresses in SIP URIs

Provider Requirements for PSTN Transfers

Twilio:

  • Enable Termination on your SIP trunk
  • Add authorized caller ID numbers in trunk settings
  • Ensure sufficient account balance

Vonage:

  • Link phone numbers to your application
  • Enable outbound permissions
  • Configure caller ID policies

Other Providers:

  • Verify trunk has bidirectional calling enabled
  • Confirm caller ID validation rules
  • Check for number porting or ownership verification requirements