SIP Jobs

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’m looking for a seasoned QA professional to craft a full-scale test strategy for our new self-serve IVR platform. The document must tell a complete story—from high-level objectives down to step-by-step execution—so that my internal QA team can pick it up and run without guesswork. Scope and focus The strategy has to address every angle of the experience: the customer-facing user interface, voice recognition accuracy, and all back-end system integrations. My end goal is threefold: prove day-one usability for callers, demonstrate rock-solid performance and reliability under load, and confirm seamless compatibility with the rest of our customer-care stack. Key features that must receive priority coverage are call routing and transfers, natural-language voice command re...

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 IPRN switch brought online fast, fully wired for billing integration and ready to carry live traffic. The core signalling will run over SIP, so every module you build or configure must interoperate cleanly with SIP endpoints and the upstream carrier trunks I already have in place. Billing is the priority: once a call lands on the switch the CDRs must flow straight into our existing rating platform without manual touches. I am open to whether you plug in a ready-made mediation layer or write custom logic—what matters is that usage records appear in real time and reconcile correctly at the end of each day. You will get SSH access to a fresh cloud instance plus the credential set for my billing server. I expect you to: • Deploy or compile the soft-switch software, enab...

I run a growing small business that relies on Microsoft 365 and a cloud-based VoIP phone system. I’m looking for a dependable partner who can step in as our day-to-day IT resource, keeping both environments running smoothly while I focus on the business itself. Most of the work is remote and ongoing. Typical tasks include adding or removing users, managing licences, tightening security policies, monitoring storage limits, resolving sync or mail-flow problems, and fine-tuning Teams, OneDrive, SharePoint, and Exchange whenever needed. On the voice side, I’ll call on you for new handset or softphone rollouts, call-flow changes, SIP or trunk tweaks, quality-of-service checks, and the occasional deep dive when call quality drops. I value quick response times, clear communicati...

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...