Asterisk PBX Jobs

## 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 (...

A FusionPBX (FreeSWITCH) installation and custom call flow configuration will be performed on a cloud server (Debian-based). The technical requirements of the project are as follows: 1. Server and Panel Installation: Installation of an up-to-date and stable FusionPBX on a Debian operating system. Configure firewall and Fail2Ban settings. 2. SIP Trunk and Incoming Call Management: Definition of SIP Trunk information (e.g., Telnyx / Verimor) provided by me. Creation of an enterprise welcome menu (IVR). Directing callers to the relevant extensions via the voice menu. 3. Dynamic Mobile Phone Routing (Critical Point): Incoming calls to extensions will be routed directly to their mobile phones. Important: The mobile numbers to which extensions are routed will not be fixed. Each extensio...

I have a cloud-hosted 3CX instance, an IP-based SIP trunk and a fresh VPS ready for deployment. Your task is to stand up an auto-dialer on that server, connect it to my SIP, and feed answered calls straight into my 3CX queue so agents see the customer’s real caller-ID. Core flow I need 1. I upload a CSV with thousands of numbers. 2. The dialer calls each number, plays a text-to-speech message, then waits: “To be connected, press 1.” 3. When the contact presses 1 the call is transferred, in real time, to my 3CX queue and rings the next available agent. 4. Agents answer inside 3CX and the customer’s number displays as the caller ID. Admin panel requirements • Switch between multiple SIP trunks on the fly. • Upload or edit the TTS IVR prompt and...

Χρειάζομαι έναν/μια έμπιστο/η γραμματέα για part-time συνεργασία, αποκλειστικά εξ αποστάσεως. Η κύρια αρμοδιότητα είναι η τηλεφωνική υποστήριξη πελατών στα Ελληνικά: θ&al...

Title: AI Automation Guide / Operator (Vapi, Twilio, n8n, Go High Level) – Long Term Description: I’m looking for someone to help guide me through the technical side of my AI-based business. I already have someone building my voice agents. What I need now is someone who understands the tools and can help me set things up properly, explain what’s going on, and make sure everything runs smoothly. I’m not looking for a big agency or someone charging very high rates. I’m looking for someone who has learned by experimenting, building, and testing things themselves, and is now looking for consistent work and a long-term collaboration. What I need help with: * Setting up accounts properly (Vapi, Twilio, n8n, Go High Level) * Making sure everything is secure (AP...

Project Overview: We are looking for a reliable freelancer based in Bangladesh to support us with Robi SIM card procurement and optionally assist with mobile network testing activities. This project can be structured in two ways depending on your availability: SIM Procurement Only (one-time task) SIM Procurement + Testing Support (short-term or ongoing engagement) Scope Option 1: SIM Procurement (Required) Procure Robi prepaid SIM cards (initial batch of at least 4, with potential for more) Ensure all SIM cards are: Legally purchased Properly registered in line with local regulations Activated and ready for standard mobile usage (SMS and voice) Share SIM details and confirm readiness Note: The SIM cards will be used strictly for network performance testing purposes for telecommunicatio...

Job Title Robi SIM Procurement & SMS Testing Support (Bangladesh) Project Overview We are looking for a reliable freelancer based in Bangladesh to support us with Robi SIM card procurement and optionally assist with short-term and ongoing SMS routing testing on the Robi network. This project has two possible scopes depending on your availability: SIM Procurement Only (one-time task) SIM Procurement + Testing Support (short-term or up to 1 month+ engagement) Scope Option 1: SIM Procurement (Required) Procure Robi SIM cards (minimum 4, potentially more) Ensure SIMs are: Properly registered and activated Able to receive SMS and voice Share SIM details and confirm readiness for testing Important: The SIM cards will be used only for Robi network testing of incoming traffic (OTP/SMS/v...

I'm seeking an expert to troubleshoot and resolve persistent sound issues in my home recording studio. I'm experiencing sound cutting out and distorting all the time, regardless of the session type. I use a combination of: - Microphones and audio interfaces - Mixers and monitors - Computers and software Ideal Skills and Experience: - Expertise in audio engineering - Experience with troubleshooting audio equipment - Proficiency in recording software and hardware setups - Strong problem-solving skills Please provide a detailed approach and previous experience.

Click-to-call, Click-to-hang-up using Asterisk Manager api (AMI) In click-to-call (in outbound) api first call hit the agents number (not using softphone or hardphone) after receiving call in agents mobile then customer number will dial. Also handling the Inbound PBX – Call is hitting in a virtual number after that call will forward in agent mobile number. WebRTC – HTTP/WebSocketSecure (WSS), PJSIP with DTLS encryption. Sngrep – to monitor the sip calls. Webhooks – Send real time call data Using Dial plan, AGI, or AMI. Web Sockets – Web RTC based browser phone. STIR/SHAKEN – Identify the source of a call. Current Project – IVR Integration with Chat bot System using asterisk 13 and above version with FreePBX....

Multilingual Voice Recording Project – Code-Switching Conversations Project Name: BV Project Type: Remote | Ongoing (Limited Slots per Locale) Project Overview Project BV is a multilingual speech data collection initiative designed to enhance Automatic Speech Recognition (ASR) systems for high-value multilingual call center scenarios, including financial services, healthcare, and telecommunications. The project focuses on collecting natural code-switching conversational audio, where speakers alternate between two languages within a single conversational turn. Scripts will be provided; however, natural delivery, fluency, and context-appropriate language switching are essential. Language Requirements Primary Language (Native level – one required): Finnish (fi-FI) ...

Multilingual Voice Recording Project – Code-Switching Conversations Project Name: BV Project Type: Remote | Ongoing (Limited Slots per Locale) Project Overview Project BV is a multilingual speech data collection initiative designed to enhance Automatic Speech Recognition (ASR) systems for high-value multilingual call center scenarios, including financial services, healthcare, and telecommunications. The project focuses on collecting natural code-switching conversational audio, where speakers alternate between two languages within a single conversational turn. Scripts will be provided; however, natural delivery, fluency, and context-appropriate language switching are essential. Language Requirements Primary Language (Native level – one required): Swedish (sv-SE) ...