Uhandisi wa Mawasiliano ya Simu Jobs

I have an e-SIM that must complete its very first registration on Elisa’s network inside Finland before I can use it abroad. The profile is voice-only, and the carrier stipulates that activation happen in person on Finnish soil. Here is what I need from you: • Be physically in Finland with a handset that supports e-SIM. • Scan the QR I provide, let the profile attach to Elisa, and place a quick test call so the voice service shows as active. • Send me a screenshot (or short video) proving Elisa reception and an outgoing call. • Remove the profile from your device right after I confirm, so I can re-download it on mine. That’s the entire task. Ideally you can finish it within a day of accepting the job. If you have any questions about the QR, han...

I have been subjected to to “voice-to-skull” transmission, direct energy weapon and remote neural monitoring(remotely). My main concern right now is the physical discomfort it is causing—persistent headaches, muscle pain, fatigue, blurred vision,disorder sleep and many other things. I need help to stop it. You must have knowledge about RF wave, WIFI wave. Here is what I expect from you: • A brief consultation (voice or message) where you help me describe my symptoms and living environment in enough detail for an accurate assessment. • A written action plan that includes: – Practical counter-measures I can put in place immediately (shielding materials, RF detection equipment, grounding techniques, etc.). – Any recommended tools or devi...

Buy a readymade CRM Telecom Software consist of order management system, which goes through phase of order creation -> resource -> provisioning -> billing -> completed Have put not put actual amount of software. Need with Sales force and Spring boot

I need to link one incoming GSM leg with a SIP participant inside FreeSWITCH so both callers join the same live conference room without noticing any difference in media or signaling. The end result must feel like a single, seamless conversation. Key requirements • FreeSWITCH must act as the bridge and mixer. • The conference room has to support call recording (start automatically and save to disk) and allow me to mute or un-mute either participant from the console or an API. Current environment – A working GSM gateway already delivers calls to FreeSWITCH through SIP. – An existing SIP trunk is in place for the VoIP side. – Root access to the FreeSWITCH server is available. Deliverable 1. Step-by-step configuration (dial-plan snippets, conference profi...

We are looking for an experienced React Native developer to help build and integrate a VoIP calling SDK into an existing mobile application. This is not a basic app development task. We need someone who has real experience with VoIP / WebRTC / SIP / SDK-level work and can deliver quickly. ### Project Scope - Integrate and extend a React Native VoIP SDK - WebRTC calling using - Inbound & outbound call handling - Push notification based incoming calls - Background call handling - App-based configuration dialing logic - SDK-style packaging + documentation support Our team will handle: - OpenSIPS / SBC (Call logic) - SIP infrastructure - APIs You will handle: - React Native SDK layer - Mobile VoIP calling integration - Native module bridging if required - Call UI handling - Stabili...

I need help with a technical issue on Linphone, integrated with FusionPBX. I’m on a custom-built version of Linphone. Current Problem: - Calls do not come through at all when the app is in the background Requirements: - Expertise in VoIP, FusionPBX, and Linphone, especially with custom builds - Proven experience with similar issues Ideal Skills: - Strong background in mobile VoIP applications - Familiarity with both iOS and Android systems - Ability to troubleshoot and resolve background operation issues Please only bid if you have relevant experience.

I need a Japanese national DID or any other VOIP number that can reliably forward calls to my chosen destination. The line will serve both personal and business purposes, so stability, clear audio and the ability to register the caller ID on common soft-phone or SIP devices is essential. Please supply, activate and demonstrate the number working via simple call-forwarding within the shortest possible time; my preference is ASAP. If you already have an inventory of Japanese DIDs, even better—let me know the formats available and any documentation requirements. Acceptance is straightforward: once I can receive and place test calls through the number without drops or quality issues, the milestone is cleared.

Project Title: 5G NR RAN Protocol Stack Development (gNodeB – L1/L2/L3) Project Overview: This project focuses on the development, enhancement, and verification of 5G NR protocol stack components within the Radio Access Network (RAN). The engagement covers gNodeB feature development, MAC Scheduler optimization (3GPP Layer 2), and seamless integration across Layer 1, Layer 2, and Layer 3 to ensure performance, compliance, and scalability. Engagement Model: Full-Time Project Assignment Experience Band: 3–8 Years (Telecom/Wireless Domain) Scope of Work: Development and verification of 5G NR protocol stack (L1/L2/L3) MAC Scheduler implementation and enhancements (3GPP Layer 2) gNodeB feature development and Low-Level Design (LLD) contributions Cross-layer interface handling ...

I'm looking for a detailed simulation of 5G soft handover in Matlab for research and analysis purposes. Key Requirements: -Simulate system-level handover with integrated PHY and MAC layer modeling in MATLAB -Consider satellite, aerial (e.g., UAV/HAPS), and terrestrial segments within a SAGIN scenario -Support evaluation of handover performance using KPIs such as handover latency, throughput, packet loss, and connection stability Provide comprehensive analysis and visualization of results (e.g., KPI trends, CDFs, outage/interruption time, HO failure rate) Ideal Skills and Experience: - Proficiency in Matlab - Strong understanding of 5G network architecture and handover mechanisms - Experience in network simulation and performance analysis - Ability to present data clearly and concis...

I need a proven Airtel Black SIP configuration running on my Ubuntu machine and handled through Linphone. The goal is a clean, documented initial setup—no trial-and-error learning on my system, please. If you have already registered an Airtel SIP trunk on Linux, you’ll know the exact registrar format, the unusual port mapping Airtel uses, and the little tweaks that keep audio flowing both ways behind NAT. Here’s the workflow I’m expecting: • Install or verify the latest Linphone and any required dependencies on my Ubuntu box. • Register the Airtel Black SIP account, applying the correct proxy, authentication string, and codec priorities. • Prove the setup with at least one inbound and one outbound call (I can join you on a test call). •...