Our dialer is compatible with the latest verions of Asterisk PBX, and, with help of PJSIP module, gives ability to integrate it with WebRTC, and other SIP clients.
You can use any number of VoIP providers and distribute your calls across them based on you needs.
No need for agents to spend their time on greetings. Simply record the message and it will be played during connection with th agent.
Ability to transfer a call to another agent
or phone number
Manager can be added to a call with ability to whisper only for the agent
Using our custom answering machine detection system you will increase performance of your agents
Using configuration templates you can easily add new agents to the system
Catch call events and attach them to triggers in you CRM or other systems
Dialer and other modules of the system can be run in your Docker or Kubernetes clusters
Our system is more accurate comparing with modules which use only volume level and voice duration analysis.
Excluding background noises, analysing utterances using our own set of trained Machine Learning models we maximise probability of correct identification of an answering machine.
To get more accurate system we are continuously train our system with new datasets.
Compatible with Asterisk PBX using EAGI with ability to run locally
Analysis is performed using own trained system for speech recognition
Average call time for analysis 3.2 seconds
Customization using module parameters