Freeswitch PBX System & API Integration
Budget: $1,500 – $3,000 CAD
Overview
We are looking for a qualified vendor or engineering team to set up a production-ready Freeswitch PBX server with advanced SIP and IVR capabilities, integrated with an API capable of handling 100+ concurrent calls per minute. This system is intended for high-volume automated calling and must be stable, scalable, and API-driven.
Scope of Work
1. System Setup
Deploy and configure a Freeswitch PBX server (on-prem or cloud-based, depending on recommendations)
Implement SIP trunking (we’ll provide trunk details if needed)
Enable support for high call concurrency (minimum 100 calls per minute, simultaneously)
2. API Requirements
Build or expose a fully functional REST API that supports:
Initiating outbound calls
Receiving webhooks or callbacks for call events (answered, failed, DTMF input, hang-up)
Injecting calls into specific IVR workflows
Real-time call status tracking
API documentation must be included
3. IVR Capabilities
Design and implement dynamic IVR flows
Support for:
Custom messages
DTMF-based input collection
Call forwarding or routing to agents/external numbers
Error handling / fallback logic
4. Performance & Resilience
Ensure system can sustain load testing of 100+ concurrent calls
Include failover or recovery plan (e.g. multiple nodes, containerized deployment, or clustering if needed)
Monitoring and basic alerting recommended
5. Deliverables
Configured Freeswitch server instance (with access credentials)
SIP and IVR configuration files
API endpoints and documentation
Load testing results with recommendations
Basic support for go-live phase (bug fixing and minor adjustments)
Timeline
We are looking to begin the setup as soon as possible, with a target go-live date of 2 days.
Next Steps
Please provide:
A proposal with estimated timeframes
High-level architecture and hosting recommendations
Budget or cost estimate
Any questions or clarifications you may need from our side
We are looking for a qualified vendor or engineering team to set up a production-ready Freeswitch PBX server with advanced SIP and IVR capabilities, integrated with an API capable of handling 100+ concurrent calls per minute. This system is intended for high-volume automated calling and must be stable, scalable, and API-driven.
Scope of Work
1. System Setup
Deploy and configure a Freeswitch PBX server (on-prem or cloud-based, depending on recommendations)
Implement SIP trunking (we’ll provide trunk details if needed)
Enable support for high call concurrency (minimum 100 calls per minute, simultaneously)
2. API Requirements
Build or expose a fully functional REST API that supports:
Initiating outbound calls
Receiving webhooks or callbacks for call events (answered, failed, DTMF input, hang-up)
Injecting calls into specific IVR workflows
Real-time call status tracking
API documentation must be included
3. IVR Capabilities
Design and implement dynamic IVR flows
Support for:
Custom messages
DTMF-based input collection
Call forwarding or routing to agents/external numbers
Error handling / fallback logic
4. Performance & Resilience
Ensure system can sustain load testing of 100+ concurrent calls
Include failover or recovery plan (e.g. multiple nodes, containerized deployment, or clustering if needed)
Monitoring and basic alerting recommended
5. Deliverables
Configured Freeswitch server instance (with access credentials)
SIP and IVR configuration files
API endpoints and documentation
Load testing results with recommendations
Basic support for go-live phase (bug fixing and minor adjustments)
Timeline
We are looking to begin the setup as soon as possible, with a target go-live date of 2 days.
Next Steps
Please provide:
A proposal with estimated timeframes
High-level architecture and hosting recommendations
Budget or cost estimate
Any questions or clarifications you may need from our side
Related categories:
Linux
Android
Cloud Computing
Asterisk PBX
VoIP
API Development
SIP
System Administration