AI VoIP Dialer with Billing
Budget: $30 – $250 USD
1. Project Summary
We are seeking a qualified developer or team to build and implement a robust and scalable telephony (VOIP) system. The system must be capable of originating and managing calls, interacting with users via an AI-powered IVR (Interactive Voice Response), and accurately capturing responses via DTMF (keypad tones).
The critical requirement of this project is its full integration with our existing administration panel and user login system. The telephony system must query and debit credits from each user's account in real-time, based on the duration and cost of each call placed.
2. Project Scope & Key Features
The project will be divided into three main components:
A. Core Telephony Module (VOIP/CPaaS)
Call Origination: Ability to initiate outbound calls to provided phone numbers.
Call Management: Full control over the call lifecycle (initiate, answer, transfer, terminate).
Quality & Scalability: The system must be built on a reliable foundation (e.g., Asterisk, FreeSWITCH, or a CPaaS API like Twilio, Vonage, etc.) to ensure high audio quality and the ability to scale call volume.
B. Interaction Module (AI & DTMF)
Intelligent IVR (AI): The system must allow for the configuration of call flows. This includes playing pre-recorded audio, using Text-to-Speech (TTS) for dynamic prompts, and ideally, using basic AI for voice recognition (Speech-to-Text) or Natural Language Understanding (NLU) to understand user intent (e.g., "Say 'Yes' or 'No'").
DTMF Capture: The system must be able to prompt the user to "press a key" (e.g., "Press 1 to confirm") and reliably capture the digit (DTMF) entered.
Flow Logic: The system must make decisions based on the DTMF input or AI response (e.g., If 1 is pressed, transfer the call; If 2 is pressed, hang up).
C. Integration & Billing Module (API)
This is the most critical part. The telephony system must function as a "service" that communicates with our existing admin panel via an API (likely REST/JSON).
Authentication: The telephony system must validate (via API) if the user requesting the call has permission in our system.
Balance Check (Pre-Call):
Before initiating any call, the system must make an API call to our backend to check the user's credit balance.
If the balance is insufficient, the call must be blocked.
Rating & Debiting (Post-Call):
The system must track the exact duration of every call (in seconds).
At the end of the call, the system must calculate the cost (based on rating rules we will define, e.g., $0.02/minute).
The system must immediately send a transaction to our API to debit the calculated amount from the user's account.
Webhooks/Callbacks: The system must send call status updates (e.g., "ringing," "answered," "failed," "completed") to our backend via webhooks, including the final duration and cost.
3. Technical Requirements (Suggestions)
Infrastructure: The candidate should suggest the best architecture (self-hosted with Asterisk/FreeSWITCH or using CPaaS APIs like Twilio).
Backend: The control logic (orchestrating calls and integration) should be developed in a robust language (e.g., Node.js, Python, PHP/Laravel, Go).
APIs: Integration will be primarily via REST/JSON APIs. The candidate will need to consume our admin API.
4. Expected Deliverables
Full Source Code: Functional, commented, and version-controlled (Git) source code for the entire telephony system and integration module.
Functional Environment: The system deployed and working in a staging environment for testing.
Documentation: Clear documentation on the architecture, how to configure call flows, and how the billing system interacts with our API.
Basic Reporting: An interface or logs that allow administrators to track call history, duration, cost, and DTMF inputs for each call.
5. Desired Qualifications
Proven experience in developing VOIP systems.
Proficiency with Asterisk, FreeSWITCH, or CPaaS APIs (Twilio, Vonage, etc.).
Strong experience in integrating RESTful (JSON) APIs.
Solid backend development experience (Node.js, Python, PHP, or similar).
Understanding of telephony protocols (SIP, RTP).
Familiarity with DTMF capture and IVR configuration.
(Desirable) Experience with AI-for-voice services (Google Dialogflow, Amazon Lex, etc.).
We are seeking a qualified developer or team to build and implement a robust and scalable telephony (VOIP) system. The system must be capable of originating and managing calls, interacting with users via an AI-powered IVR (Interactive Voice Response), and accurately capturing responses via DTMF (keypad tones).
The critical requirement of this project is its full integration with our existing administration panel and user login system. The telephony system must query and debit credits from each user's account in real-time, based on the duration and cost of each call placed.
2. Project Scope & Key Features
The project will be divided into three main components:
A. Core Telephony Module (VOIP/CPaaS)
Call Origination: Ability to initiate outbound calls to provided phone numbers.
Call Management: Full control over the call lifecycle (initiate, answer, transfer, terminate).
Quality & Scalability: The system must be built on a reliable foundation (e.g., Asterisk, FreeSWITCH, or a CPaaS API like Twilio, Vonage, etc.) to ensure high audio quality and the ability to scale call volume.
B. Interaction Module (AI & DTMF)
Intelligent IVR (AI): The system must allow for the configuration of call flows. This includes playing pre-recorded audio, using Text-to-Speech (TTS) for dynamic prompts, and ideally, using basic AI for voice recognition (Speech-to-Text) or Natural Language Understanding (NLU) to understand user intent (e.g., "Say 'Yes' or 'No'").
DTMF Capture: The system must be able to prompt the user to "press a key" (e.g., "Press 1 to confirm") and reliably capture the digit (DTMF) entered.
Flow Logic: The system must make decisions based on the DTMF input or AI response (e.g., If 1 is pressed, transfer the call; If 2 is pressed, hang up).
C. Integration & Billing Module (API)
This is the most critical part. The telephony system must function as a "service" that communicates with our existing admin panel via an API (likely REST/JSON).
Authentication: The telephony system must validate (via API) if the user requesting the call has permission in our system.
Balance Check (Pre-Call):
Before initiating any call, the system must make an API call to our backend to check the user's credit balance.
If the balance is insufficient, the call must be blocked.
Rating & Debiting (Post-Call):
The system must track the exact duration of every call (in seconds).
At the end of the call, the system must calculate the cost (based on rating rules we will define, e.g., $0.02/minute).
The system must immediately send a transaction to our API to debit the calculated amount from the user's account.
Webhooks/Callbacks: The system must send call status updates (e.g., "ringing," "answered," "failed," "completed") to our backend via webhooks, including the final duration and cost.
3. Technical Requirements (Suggestions)
Infrastructure: The candidate should suggest the best architecture (self-hosted with Asterisk/FreeSWITCH or using CPaaS APIs like Twilio).
Backend: The control logic (orchestrating calls and integration) should be developed in a robust language (e.g., Node.js, Python, PHP/Laravel, Go).
APIs: Integration will be primarily via REST/JSON APIs. The candidate will need to consume our admin API.
4. Expected Deliverables
Full Source Code: Functional, commented, and version-controlled (Git) source code for the entire telephony system and integration module.
Functional Environment: The system deployed and working in a staging environment for testing.
Documentation: Clear documentation on the architecture, how to configure call flows, and how the billing system interacts with our API.
Basic Reporting: An interface or logs that allow administrators to track call history, duration, cost, and DTMF inputs for each call.
5. Desired Qualifications
Proven experience in developing VOIP systems.
Proficiency with Asterisk, FreeSWITCH, or CPaaS APIs (Twilio, Vonage, etc.).
Strong experience in integrating RESTful (JSON) APIs.
Solid backend development experience (Node.js, Python, PHP, or similar).
Understanding of telephony protocols (SIP, RTP).
Familiarity with DTMF capture and IVR configuration.
(Desirable) Experience with AI-for-voice services (Google Dialogflow, Amazon Lex, etc.).