Implementation of Asterisk-Based Call Center System
Budget: $3,000 – $5,000 USD
## 1. Project Scope
The service provider will design, install, and configure a production-ready VoIP Call Center system using Asterisk on a Linux-based environment. The scope includes:
* Server Environment: Installation and hardening of the latest stable Asterisk version on a cloud-based or local Linux server (Ubuntu/Debian).
* Trunking & Connectivity: Integration of [X] number of SIP Trunks for external inbound and outbound calling.
* Extension Management: Creation and configuration of [X] agent extensions supporting PJSIP/SIP protocols.
* Call Flow Logic (Dialplan):
* Development of a multi-level IVR (Auto-Attendant) menu.
* Implementation of time-based routing (Business Hours vs. After Hours).
* Queue Management (ACD): Setup of call queues with specific distribution strategies (e.g., Round Robin, Least Recent) and custom "Music on Hold."
* Security: Implementation of a VoIP-specific firewall, Fail2Ban, and SIP port obfuscation to prevent unauthorized access.
* System Features: Enablement of Call Recording, Voicemail-to-Email, and Call Transfer capabilities.
*
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## 2. Project Deliverables
1. Fully Operational PBX: A live, tested Asterisk system capable of handling [X] concurrent calls.
2. Configuration Documentation: A technical document detailing the IP addresses, SIP credentials, extension list, and Dialplan logic.
3. Admin Operations Guide: A "How-To" manual for common tasks (e.g., adding/removing agents, changing IVR recordings, accessing call logs).
4. Security Audit Report: Confirmation of active security protocols and a list of closed vulnerabilities.
5. UAT (User Acceptance Test) Report: Successful demonstration of inbound/outbound call quality, queue functionality, and recording retrieval.
The service provider will design, install, and configure a production-ready VoIP Call Center system using Asterisk on a Linux-based environment. The scope includes:
* Server Environment: Installation and hardening of the latest stable Asterisk version on a cloud-based or local Linux server (Ubuntu/Debian).
* Trunking & Connectivity: Integration of [X] number of SIP Trunks for external inbound and outbound calling.
* Extension Management: Creation and configuration of [X] agent extensions supporting PJSIP/SIP protocols.
* Call Flow Logic (Dialplan):
* Development of a multi-level IVR (Auto-Attendant) menu.
* Implementation of time-based routing (Business Hours vs. After Hours).
* Queue Management (ACD): Setup of call queues with specific distribution strategies (e.g., Round Robin, Least Recent) and custom "Music on Hold."
* Security: Implementation of a VoIP-specific firewall, Fail2Ban, and SIP port obfuscation to prevent unauthorized access.
* System Features: Enablement of Call Recording, Voicemail-to-Email, and Call Transfer capabilities.
*
------------------------------
## 2. Project Deliverables
1. Fully Operational PBX: A live, tested Asterisk system capable of handling [X] concurrent calls.
2. Configuration Documentation: A technical document detailing the IP addresses, SIP credentials, extension list, and Dialplan logic.
3. Admin Operations Guide: A "How-To" manual for common tasks (e.g., adding/removing agents, changing IVR recordings, accessing call logs).
4. Security Audit Report: Confirmation of active security protocols and a list of closed vulnerabilities.
5. UAT (User Acceptance Test) Report: Successful demonstration of inbound/outbound call quality, queue functionality, and recording retrieval.