Route PBX calls to a SIP domain using FONE API interface. You can also enable connection over SRTP/TLS and set up your own safety and security guidelines.
Using FONE API, setup your VoIP infrastructure to direct incoming calls on a specific number to your new SIP domain. Your web servers will receive a callback via HTTP every time FONE API gets a call.
Configure your application web server to call numbers from the SIP-enabled domain. It will then communicate with UML to determine which client username or SIP username to direct the call.
UML users sign up using a token which specifies the client or SIP username. FONE API SDKs include functions that are initiated once a call comes in.
Every WebRTC-enabled agent should have its unique phone number for FONE API to accurately pick up calls from your contact center.
Send every call that your call center normally assigns to a WebRTC agent to your new FONE API number. The moment YC receives these calls, your web server receives a webhook.
Your app arranges the number with the WebRTC agent, and answers with UML to identify the username to call.
FONE API users sign up using a token which specifies the client or SIP username. FONE API SDKs include functions that are initiated once a call comes in.
Automated failover ensures that you have 99.95% uptime SLA without the need for a maintenance window.
Use existing apps to new markets by configuring features for compliance and localization.
Use a single platform for voice, SMS, video, authentication, chat and more.
Get free support, have the freedom to scale your business, market faster with pay-as-you-go.