SIP Trunking with Calls Hero

Seamlessly connect your existing phone system with Calls Hero conversational AI agents using SIP trunking. Leverage advanced AI capabilities without changing your current telephony infrastructure.

Overview

  • Integrate your PBX or SIP-enabled phone system directly with Calls Hero AI agents.
  • Route calls to AI agents without altering your existing telephony setup.
  • Support for both inbound and outbound calls.

How SIP Trunking Works

  • Inbound calls: Your SIP trunk routes calls to Calls Hero using our origination URI.
  • Outbound calls: Calls initiated by Calls Hero are routed to your SIP trunk via your termination URI.
  • Authentication: Secure connections using digest authentication (username/password) or Access Control List (ACL) by signalling IP.
  • Signalling & Media: SIP signalling uses TCP; audio (RTP) streams over UDP for low latency.

Requirements

  • SIP-compatible PBX or telephony system
  • Administrator access to SIP trunk configuration
  • Appropriate firewall rules to allow SIP and RTP traffic

Setting Up SIP Trunking

  1. Navigate to Phone Numbers:
    Go to the Phone Numbers section in the Calls Hero dashboard.
  2. Import SIP Trunk:
    Click “Import a phone number from SIP trunk”, select the SIP trunk option. The system pre-fills the Calls Hero origination URI for inbound calls: sip:sip.rtc.callshero.com:5060;transport=tcp.
  3. Enter Configuration Details:
    • Label: Descriptive name
    • Phone Number: E.164 format (e.g. +441234567890)
    • Termination URI: Where Calls Hero sends outbound calls (your SIP system)
  4. Configure Authentication:
    Optionally enter SIP digest username and password, or use ACL authentication (allowlist IP addresses).
  5. Complete Setup:
    Click “Import” to finalise.

Assigning Agents to Phone Numbers

  1. Go to the Phone Numbers section in the dashboard.
  2. Select your imported SIP trunk number.
  3. Click “Assign Agent” and choose the AI agent to handle calls to this number.

Troubleshooting & Tips

  • Connection issues?
    • Check firewall allows TCP/UDP port 5060 to sip.rtc.callshero.com.
    • Verify SIP credentials and termination URI.
  • Authentication problems?
    • Double-check digest username/password if used.
    • With ACL, ensure correct allowlist for Calls Hero IP.
  • No or one-way audio?
    • Ensure firewall allows UDP RTP media ports (typically 10000–60000).
    • Check for NAT or network translation issues.
  • Audio quality issues?
    • Ensure at least 100 Kbps per call, low latency/jitter.
    • Check network congestion and packet loss.
    • Verify codecs are matched at both ends (Calls Hero supports PCM 16kHz).

FAQ

  • Can I use my existing phone numbers? Yes. SIP trunking enables you to connect your existing numbers to Calls Hero without porting.
  • What SIP trunk providers are supported? Calls Hero works with most SIP providers, including Twilio, Vonage, RingCentral, Sinch, Infobip, Telnyx, Exotel, Plivo, Bandwidth, and other standards-based SIP services.
  • How many concurrent calls can I have? This depends on your Calls Hero subscription plan.
  • Is call encryption available? Yes, encrypted SIP (SIPS) is supported for secure calls. Contact support for requirements.
  • Can I route calls to different agents? Yes, use your PBX rules to route calls to specific numbers or agents as needed.

Limitations & Considerations

  • Concurrent call limits depend on your plan.
  • Call recording and analytics may require further setup.
  • Outbound calling capabilities are subject to your SIP trunk provider.
  • Currently, only PCM audio formats (e.g., PCM 16kHz) are supported for SIP trunk connections.