Asterisk PBX Jobs

Mam na własnym VPS zainstalowaną najnowszą wersję GoAutoDial 4. Potrzebuję specjalisty, który poprawnie skonfiguruje połączenie z moim dostawcą SIP trunk – Zadarma – tak, aby system bezbłędnie realizował zarówno połączenia wychodzące, jak i odbierał przychodzące. Zakres prac: • dodać i zweryfikować trunk Zadarma, • utworzyć odpowiednie dial-plany i reguły routingu, • przeprowadzić testowe połączenia w celu potwierdzenia jakości oraz stabilności, • wprowadzić ewentualne korekty konfiguracji serwera Asterisk wchodzącego w skład GoAutoDial. Środowisko obejmuje jeden VPS, więc konfiguracja dotyczy pojedynczego serwera. Po zakończeniu proszę o krótką instrukcję krok-po-kroku, abym mógł samodzielnie zarządzać podstawowymi zmiana...

I’m looking to partner with a reliable call-centre operation that can dedicate 1 – 5 agents exclusively to inbound customer support over the phone. All interactions will be in English, so clear diction and a neutral accent are essential. The agents will be the first point of contact for my customers, answering questions, resolving basic issues, and routing complex cases to my in-house team when needed. Average volume is modest right now, but I need a team that can scale quickly if call traffic grows. Core expectations • Live phone coverage during the agreed shift window • Courteous, solution-focused conversations that follow my brand guidelines • Real-time ticket or CRM updates so I always know the status of each call • Daily call logs and a conc...

Project Title: Full Issabel 5 PBX Deployment: Installation, Trunks & Extensions Project Overview I am seeking a VoIP specialist to perform a complete installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architecture, including SIP trunks for external connectivity and internal extensions for users. Detailed Scope of Work 1. Server Installation & Hardening Perform a clean installation of Issabel 5 on AlmaLinux 8 using the official net-install script. Configure Fail2ban and firewall rules to block unauthorized SIP and SSH attempts. Set secure passwords for the Linux root, MariaDB, and Issabel web admin. 2. SIP Trunk Configuration Connect the PBX to my chosen VoIP provider using a SIP trunk....

I don't have FusionPBX installed on any server yet. You will install and setup fusionpbx on a server I will provide credentials access into. I want to be able to present a specific caller ID on every outbound call placed from my softphone. The softphone I will be using is Linphone, so everything you build must be fully tested from that app before we wrap up. What I need you to do • Configure the relevant SIP profile(s) and outbound route so that the number I supply appears consistently, regardless of the device extension that initiates the call. • Make any dial-plan or gateway adjustments required for a typical commercial SIP trunk. I have not fixed on one provider yet, so keep the setup provider-agnostic and document the few fields I will have to tweak once I decide. ...

I need an outbound voice AI that can pick up the phone, speak naturally with customers, and lock in service appointments without human involvement. The bot should dial leads from a list, confirm the customer’s need for our service, offer available slots pulled from an online calendar, secure the booking, and then push the confirmed appointment back to our CRM while sending the customer a confirmation SMS or email. Core flow • Call initiation through a voice gateway such as Twilio • Speech recognition and NLU (Dialogflow CX, Amazon Lex, Rasa, or similar) to understand the caller’s intent, handle clarifying questions, and gracefully manage “I didn’t catch that” moments. • Text-to-speech that sounds human; Google WaveNet, Amazon Polly or Micro...

I’m building a voice-driven receptionist that greets visitors or callers, listens to their questions, and replies with a natural-sounding voice. The core flow is straightforward: • Speech-to-text: Accurate recognition of English only is required for now. • NLP: Classify the intent and pull the correct answer when a caller asks about our business hours or location. • Text-to-speech: Respond with friendly, human-like audio. • Fallback: whenever the system is unsure, it should politely ask for more details rather than handing the call off or giving a canned line. I need the full stack—STT, intent handling, response generation, and audio playback—wrapped in a module that I can drop into my existing website widget today and expand to a phone line ...

I need a streamlined website whose sole function is to host and manage a voice-based AI call-center. The site will act as the control panel and public-facing entry point for customers who dial in, while giving me an admin area to configure and monitor activity. Core requirements • Automated responses: callers must hear natural-sounding, AI-generated answers that pull from a knowledge base I can update myself. • Live call routing: when the AI decides human help is needed, it should hand the caller off to a live agent instantly and log the transfer. • Voice channel only: chat or email may be added later, but right now the system must excel at telephone interactions. Technical expectations – Cloud telephony integration (Twilio, Plivo, or similar) for SIP/VoIP ha...

I’m building a voice-driven receptionist that greets visitors or callers, listens to their questions, and replies with a natural-sounding voice. The core flow is straightforward: • Speech-to-text: Accurate recognition of English only is required for now. • NLP: Classify the intent and pull the correct answer when a caller asks about our business hours or location. • Text-to-speech: Respond with friendly, human-like audio. • Fallback: whenever the system is unsure, it should politely ask for more details rather than handing the call off or giving a canned line. I need the full stack—STT, intent handling, response generation, and audio playback—wrapped in a module that I can drop into my existing website widget today and expand to a phone line ...

I need a fully-functional Google Voice account for 10 Users supported, registered from India but tied to a U.S. phone number, so I can place and receive personal calls to and from the States. The account itself does not yet exist and I have not picked a number, so part of the job is guiding me through—or handling—the selection of an available U.S. number that feels natural to me. Because Google Voice is officially unsupported here, you’ll have to take care of the usual hurdles (VPN, verification SMS to a real U.S. mobile, billing profile if required) and then hand over a clean login that I can secure with my own 2-step authentication. I plan to use the service interchangeably on both my smartphone and my laptop, so please configure and test calling on the Google Voice mo...

Necesito conectar nuestro PMS Cloud con la centralita Asterisk para cubrir funcionalidades de gestión de hotel. El objetivo es que el personal pueda manejar desde el PMS: • Registro de huéspedes: que cada check-in o check-out actualice automáticamente el estado de la extensión telefónica asignada. • Asignación de habitaciones: que al cambiar una habitación en el PMS se reprograme la extensión correspondiente en Asterisk sin pasos manuales. • Facturación y pagos: que las llamadas salientes e internas se registren en la cuenta del huésped y se reflejen en la factura final. Ya contamos con un software específico de gestión hotelera en la nube; requiero que el integrador trabaje sobre sus A...

I already have a Contabo server standing by and simply need a clean, production-ready ViciDial stack on it. The job covers: • Installing the latest stable ViciDial (with Asterisk and its dependencies) from scratch. • Optimising the underlying Linux distro you feel is most reliable for call-centre workloads. • Hardening the box with a well-tested firewall configuration—CSF, UFW, iptables, or a similar solution is fine—as long as only the ports Vicidial, SSH and web administration actually require remain open. • Verifying that the web interface, database, and telephony services all start automatically after a reboot and that calls can be placed through a demo campaign. • Supplying a concise hand-off sheet: all commands run, credentials created, a...

I’m looking to have a fully-functional AI receptionist built that can pick up every inbound call, hold a natural conversation with the caller, and then route the call to the right person or department purely on the basis of what the caller actually says. The key outcomes I need are: • Advanced conversational AI – the system must go well beyond simple keyword spotting and understand context, intent, and follow-up questions so the interaction feels human. • Dynamic query-based routing – once the caller’s need is clear, the AI should transfer the call to the appropriate extension or external number automatically. • Clean hand-off – when the call is routed, the receiving party must get a short, accurate summary of the caller’s request...

My Vicidial server is already installed, secured, and reachable from the public internet; what’s missing is the practical know-how to make calls and launch the very first blended (inbound + outbound) campaign. Here is what I need you to walk me through and, where necessary, configure directly on the box: • Register my SIP-trunking provider inside Vicidial, confirm two-way audio, and run a quick live test call. • Create one blended campaign with at least a test list, DID routing for inbound, and the appropriate dial plan entries for outbound. • Show me, via screenshare or concise step-by-step notes, how to add new agents, upload lead lists, record custom greetings, and monitor live calls so I can repeat the process after you leave. Acceptance criteria – S...

I have a WebRTC soft-phone built with JsSIP that needs to register to an Asterisk 18 server over WSS. SIP credentials are confirmed correct, yet the browser console shows an authentication failure. The signalling path is protected with TLS certificates, so the problem is somewhere in the certificate handling or the way Asterisk presents the challenge. Your job is to trace and eliminate the registration failure, then hand back a clean configuration and proof that the client can successfully register and place a test call. Environment details you will touch: – Asterisk 18 (pjsip stack enabled) – JsSIP running in a standard browser (wss://) – TLS with server and client certificates already issued Acceptance criteria: • JsSIP completes REGISTER without 401/403 r...

I need a custom panel for my 3CX phone system that listens for DTMF tones during live calls, captures every numerical input the caller enters, and turns those digits into real-time reports I can view instantly in a browser. The key for me is accuracy and speed: as soon as the caller finishes entering a number, the panel should refresh the on-screen report so supervisors can act on the information while the call is still active. I am not looking to store the data long-term or trigger downstream workflows—just fast, on-the-spot visibility of what the caller keyed in. Because everything must sit cleanly inside (or alongside) 3CX, please base your solution on technology that plays well with the 3CX Call Control API, SIP messages, or any proven method you have used before for intercep...