Setup Linux Telecom Server

@farooqtahir2013

As an experienced Network, Cybersecurity, VoIP and System Engineer, I'm confident in my ability to turn your fresh Linux box into a powerful telecom server. With over a decade working with leading names like Cisco, VMware, IBM, and Fortinet, I understand the importance of well-commented config files stored in a version-controlled repo and clean source-based/package installs that we will use in your project to ensure both security and functionality. My past includes extensive work with various VoIP platforms, including FreePBX, 3CX, Vodia, Asterisk and Cisco CUCM/CME, which are in line with the systems you need. Moreover, I have keen experience in system administration on both Microsoft and Linux servers. In this light, setting up the software stack you require using systemd will not be a challenge. What sets me apart is my commitment to delivering projects timely and effectively. I'll also provide clear documentation for any future reference and run a handover session to walk you through the deployment. My skills include excellent networking knowledge for maintenance and upkeep post-installation too. I am ready to bring my expertise to your project for its timely deployment for all your telecommunication needs!

@AZTURK1

Hi there, I read your spec carefully—you need ejabberd, Kamailio, FreeSWITCH and Class 4/5 softswitch layers on a fresh Linux box with full systemd boot, working SIP/PSTN calls, XMPP presence and video, all open-source and documented. I’ve built these exact stacks several times. What I'll do: ✅ Install and configure ejabberd, Kamailio, FreeSWITCH with Class 4 and Class 5 routing, and wire everything to start at boot. ✅ Set up SIP/PSTN trunks, XMPP messaging/presence, and HD video calling; test with softphones and clients; hand over via SSH walkthrough with version-controlled docs. ✅ I’ll sandbox-test the whole setup and stage the rollout so production misconfigs are avoided. Skills for this project: ✅ Linux sysadmin (Debian/Ubuntu/CentOS – your choice) ✅ VoIP / SIP / Kamailio / FreeSWITCH ✅ XMPP / ejabberd ✅ Telecommunications engineering & softswitch (Class 4/5) ✅ Microsoft® Certified: MCSA | MCSE | MCT ✅ 300+ projects delivered, 280+ five-star reviews Why me: I’m available 24/7, reply fast, and provide unlimited revisions until you’re satisfied. Quick question: Which Linux distro do you prefer for the build? I can deliver in 1 day for $124 and start right now. Let’s get it live.

@engkarimismail

As an accomplished IT professional, my expertise in server administration, networking, and system implementation makes me the ideal candidate for your Linux Telecom Server setup. Over the years, I have honed my skills in deploying virtualization frameworks such as VMWare ESXi, Hyper-V, and Proxmox VE, and gained extensive experience with Linux operating systems, including Debian. These proficiencies are the foundation that drives efficient execution when working with tools like ejabberd for XMPP-based messaging, Kamailio for SIP routing-registrar duties, FreeSWITCH facilitation of multimedia handling among others. Essentially, this project falls right within my wheelhouse. To deliver your ideal solution, I will guarantee clean source-based or package installs (based on your preference) that comply with the best security practices. Furthermore, I'll provide well-commented configuration files that will be saved in a version-controlled repository. This approach ensures future scalability and simplified troubleshooting procedures. Given my command of the tools needed for this task —right from DNS/DHCP solutions to VoIP/IP telephony—I am well-equipped to implement a stable platform with impeccable call quality, flawless messaging functionalities and reliable media processing capabilities as required.

@android2014213

With my extensive 10+ years of experience as a Linux/Windows System Administrator, Devops, and Cloud Engineer, I am uniquely positioned to tackle your project. Installing and configuring the software stack you require - eJabberd, Kamailio, FreeSWITCH, and more - is my everyday toolkit. Reliable communication entails clear audio, video, and data sharing which are sensitive aspects of your project that I am acutely aware of. My proficiency in using open-source components aligns with your preference for an exclusive reliance on them during this project. Not only can I build a Linux server from scratch but also deliver secure source-based or package installs adhering to best security practices – a critical factor for your unified communications platform. Additionally, maintaining well-commented configuration files stored in a version-controlled repository is part of my workflow by default. This ensures not only a smooth handover but also simplifies the addition of new tenants or dial plan rules in your system. Timing is important especially given the complexity of the project; let's collaborate effectively to get your platform live as soon as possible while maintaining excellent service quality through diligent system monitoring and optimization. Let's connect!

@gotucat

Hi, I reviewed the project and the main requirement is to turn a fresh Linux box into a full-featured telecom platform supporting VoIP, XMPP messaging with presence, video calling, and multimedia handling using open-source components. With solid System Administration discipline, I’ll install and configure ejabberd for XMPP-based messaging, Kamailio as SIP registrar/router, FreeSWITCH as the media server, and both softswitch layers to handle retail and carrier routing. I’ll apply security best practices, keep well-commented configs in a version-controlled repo, and ensure all services boot via systemd. The stack will support VoIP, PSTN, presence, and HD video sessions, tested with popular clients. Expect quality, responsiveness, clean work, and easy management through well-documented steps and maintainable configs. Let’s discuss here now.

@softsolution2000

I am a highly experienced telecom engineer with a strong background in Linux and open-source communications platforms. I am confident in my ability to set up and configure the required software stack including ejabberd, Kamailio, FreeSWITCH, and Class 5/4 softswitch layers on a fresh Linux server. I will ensure all services start at boot, conduct thorough testing to ensure seamless communication, provide detailed documentation, and offer a comprehensive SSH hand-over session for deployment walkthrough. I am ready to discuss timelines and milestones to get the platform live. See the above links please.

@shivamhardia

Deploying a telecom platform goes far beyond simply installing software, it requires ensuring that every call, message, and media session performs reliably in real-world conditions. That is precisely where I can add value. I have hands-on experience with Linux-based VoIP infrastructure, SIP routing, FreeSWITCH, Kamailio, and secure server deployments. I'll deliver a production-ready environment with clean configurations, systemd-managed services, version-controlled configs, comprehensive documentation, and complete validation of SIP/PSTN calling, XMPP messaging, presence, video, and IVR workflows before handover. Let's discuss your deployment goals, timeline, and milestones to get this platform running the right way from day one. Thanks!

@nemanjam61

Hello, I’ll script systemd unit files with watchdog timers so each service—ejabberd, Kamailio, FreeSWITCH—restarts automatically and survives reboots, while hardening configs via SELinux and fail2ban on a fresh Debian install. For PSTN I’ll set up DAHDI with proper timing drivers and map them into the Class 5 softswitch so inbound/outbound calls work out‑of‑the‑box, avoiding the silent‑failure edge case many miss. Do you prefer source compilation for all components or packages where available? Looking forward to working with you. Nemanja M.

@fmateo05

Hello. I would install your PBX setup as requested; I would use 2600hz/kazoo platform. It has most of the features that you request. It includes kamailio, freeswitch, rabbitmq, and 2600hz/kazoo. If you desire I have a live demo available

@Hrolgar

Do you already have the SIP trunk, DNS records, TLS certs, and any tenant numbering plan ready, or should I treat those as part of the setup work? I'd approach this as a production-readiness build, not just package installation. First I'd lock down Debian/Linux basics, systemd restart behaviour, firewalling, Kamailio registration/routing, ejabberd XMPP presence, and FreeSWITCH media tests. Then I'd wire PSTN, LCR, IVR, video, tenant admin, dial plans, git-tracked configs, and the SSH handover notes. M1: SIP/XMPP/media foundation, services boot cleanly and basic tests pass, $100, 3 days. M2: PSTN/LCR/IVR/video/tenant workflows, docs, git repo, and handover, $150, 4 days. The $250 is based on the brief; trunk/provider rules and tenant complexity are the bits that could move it. M1 is the right first step to confirm scope before the rest.

@teodorgeorge

Hello, My Linux telecom work includes SIP routing, FreeSWITCH media services, Kamailio registrar setups, ejabberd presence, and Asterisk PBX interoperability. I understand the need for Class 4 and Class 5 layers, systemd boot handling, secure configs, and clean documentation. I can install the open-source stack, store commented configs in Git, and test calls, XMPP presence, HD video, IVR, and media tasks with common clients. I will also prepare tenant and dial plan notes, then walk you through the handover over SSH. Best regards, Teo

@ayesha86664

Hi there! I understand you need a complete open-source telecom platform setup with VoIP, messaging, video, and carrier-grade routing. The main challenge is configuring multiple real-time services together with strong stability, security, and production readiness. I have experience with Linux server administration, VoIP systems, SIP, Asterisk, XMPP, FreeSWITCH, Kamailio, and telecom infrastructure. I have worked on communication platforms, SIP routing, PBX setups, server configurations, and real-time service deployments. My focus is building secure and reliable telecom solutions. My approach is to install and configure all required components, set up SIP routing, media handling, XMPP messaging, Class 4 and Class 5 features, and secure Linux services. I will test calls, messaging, video sessions, and system recovery after reboot. I will also provide documentation and a proper handover session. check our work https://www.freelancer.com/u/ayesha86664 Do you already have preferred Linux distribution and carrier SIP trunk details, or should I prepare the complete environment from scratch? Let me know if you’re interested & we can discuss it. Best Regards Ayesha

@nintysix96

Treat the bid amount and timeline above as rough ballpark figures, we will work out real numbers once we've talked through the scope. This is a full telecom stack build, not just a server setup. You need ejabberd, Kamailio, FreeSWITCH and both Class 4 and Class 5 softswitch layers all talking to each other cleanly on a single Linux box, with PSTN connectivity, XMPP presence, HD video and IVR working end to end. The finish line is an SSH handover where you can pick up a softphone and place a real call without touching a config file. Here's how we'd approach this: - Base OS and hardening: Start with a clean Debian install, lock down SSH, configure firewall rules for SIP/RTP port ranges, and set up the systemd service structure before any telecom software touches the box. - Kamailio + FreeSWITCH core: Deploy Kamailio as the SIP registrar and router, wire it to FreeSWITCH over the event socket so registration and call legs hand off correctly. This is the heart of the platform. - ejabberd XMPP layer: Install and configure ejabberd for messaging and presence, with S2S and client-to-server TLS, tied into the same user directory as the SIP side where possible. - Class 4 and Class 5 softswitch: Build the Class 5 retail feature layer on top of FreeSWITCH (voicemail, IVR, hunt groups), then configure the Class 4 carrier routing and LCR on Kamailio with dial plan rules for PSTN termination. - Docs and version control: All configs go into a Git repo with comments explaining every non-obvious setting. Install steps, tenant onboarding and dial plan changes all get written up so you're not dependent on us for day-to-day ops. Once we've had a quick scope call to confirm your PSTN provider, expected call volume and any existing infrastructure, we'll send a proper proposal with milestones, timeline and a firm price. Want to jump on a short call this week to walk through the architecture before we quote firm numbers? Best, 96 Studio

@jonasdedzo

Hi, Jonas here, from Ghana, to help you. Stitching together ejabberd, Kamailio and FreeSWITCH only stays stable in production if the SIP routing and softswitch layers are proven under real call load before you rely on them, otherwise everything works fine in testing until the first busy hour exposes a gap. I noticed you want documentation covering how to add new tenants or dial plan rules, that usually means you're planning to scale this beyond initial setup, so I'd structure the configs and version-controlled repo with multi-tenancy in mind from the start rather than a single-tenant quick build. I'd install and configure ejabberd for messaging, Kamailio as SIP router, FreeSWITCH as the media engine, and layer in Class 4/5 softswitch functionality, all systemd-managed with version-controlled configs and full documentation. I've worked with similar open-source VoIP and telecom stack deployments before. A few questions to better understand this project: Q1: Do you have PSTN connectivity already arranged through a SIP trunk provider, or does that need setting up too? Q2: Roughly how many concurrent calls or tenants should the platform handle at launch? Q3: For video calling, is WebRTC through FreeSWITCH sufficient, or do you need support for specific mobile SDKs? Looking forward to hearing from you.

@ramoneduardo73

Getting a telecom server running reliably comes down to the choice between Asterisk or FreePBX based on your scaling needs, combined with strict SIP trunking security to prevent toll fraud. I’d start by hardening the kernel and configuring Fail2Ban with custom Jails specifically for your SIP ports to kill unauthorized traffic before it hits the daemon. For the stack, if you're looking for high concurrency, I'd suggest implementing Kamailio as a lightweight SIP router in front of your Asterisk core to manage load balancing efficiently. As a pro-tip, we should also script a cron job that monitors your packet loss and jitter in real-time and logs the output to a simple CSV or SQLite DB—it makes troubleshooting call quality issues 10x faster later on. I’m ready to dig into the shell and get this architecture locked down. If you have the server specs ready, drop me a message in the chat with your preferred distro and we can get moving. Ah by the way, hi! How are you doing? haha

@saifali012

Hi! I understand you need a production-ready open-source telecom platform on Linux that combines SIP routing, media processing, XMPP messaging, PSTN connectivity, and both Class 4 and Class 5 softswitch capabilities into a secure, maintainable deployment. I can deploy and configure Kamailio, FreeSWITCH, ejabberd, and the required Class 4/Class 5 softswitch architecture following telecom best practices. The platform will include secure SIP routing, XMPP presence and messaging, IVR and media services, LCR, multi-tenant support, PSTN integration, systemd service management, hardened Linux configuration, Git-managed configuration files, and comprehensive documentation covering installation, dial plans, tenant provisioning, and ongoing administration. Before handover, I'll validate end-to-end , video, and media workflows using standard SIP and XMPP clients to ensure the system is production-ready. Open chat now and let's discuss the project requirements in detail.

@mojjammil

Hello, I understand this is a telecom infrastructure build where the focus is not only installing individual open-source components, but making the complete communication stack work reliably together across SIP, XMPP, media handling, and carrier routing. Similar deployments require careful planning around signaling flows, security hardening, interoperability testing, and operational handover. I’d start by preparing the Linux environment, mapping the required call and messaging flows, then install and configure each service with version-controlled configuration files. After integration, I’d validate SIP registration, PSTN routing, messaging presence, video sessions, IVR flows, and reboot recovery before documenting the final setup. A key challenge will be making the Class 4/5 switching layers, SIP routing, and media services cooperate cleanly while maintaining performance and troubleshooting visibility. Which Linux distribution and server specifications will be used for the deployment, and is this intended for a single tenant initially or a multi-tenant carrier setup? Do you already have SIP trunk providers and PSTN connectivity available, or should the routing and carrier integration layer be prepared as part of this project? If we're aligned, I can outline the implementation phases before kickoff. Best regards, Mojjammil

@millz

The part that actually breaks a stack like this isn't any single component — it's Kamailio and FreeSWITCH racing each other on call state. Route everything through Kamailio blindly and you get double-answered legs or orphaned dialogs on an unexpected re-INVITE; Kamailio should stay purely SIP-aware (registrar, routing, LCR) while FreeSWITCH owns all media and B2BUA state, never the reverse. I'd stand up ejabberd first since it's self-contained, then Kamailio with a dispatcher module for the Class 4 LCR layer, then FreeSWITCH's mod_sofia handling Class 5 retail features (voicemail, IVR, conferencing) — each on systemd with health-check units so a crash restarts the process instead of silently dropping calls. Fail2ban plus rate-limiting on the SIP registrar from day one is non-negotiable; an open Kamailio registrar is the fastest way these boxes get toll-frauded. For proof of production-readiness, I'd run concurrent SIP, PSTN, XMPP, and video sessions together, not isolated single-call tests, since that's where NAT traversal and resource contention actually surface. I've built SIP/XMPP stacks like this and can start on a milestone plan. What's terminating PSTN — a SIP trunk or physical E1/T1/FXO hardware? That changes whether FreeSWITCH needs a hardware gateway module.

@mobasherreza10

As an experienced technology partner with extensive knowledge and skills in system administration, I am confident that I'm the best fit for your project to set up a Linux Telecom Server. Although my background focuses mainly on branding and design, my repertoire goes far beyond that. With my additional expertise in software development & DevOps and AI, data & automation, I possess a comprehensive understanding of the tools and techniques required for this task. I have hands-on experience with nearly all the softwares you've listed, including configuring ejabberd, Kamailio, FreeSWITCH, etc., as part of various telecommunications platforms in the past. Being a seasoned telecom engineer, I’ve mastered secure-based installations and package installations of these open-source components while following industry's best security practices. My inclination towards clean work comes with the added benefit of documenting everything meticulously - every installation step, key config settings and even instructions on how to add new tenants or dial plan rules.