Freeswitchtrabajos

...arquitectura. Se aceptan soluciones basadas en: * Kamailio * OpenSIPS * FreeSWITCH * Asterisk * RTPengine * Media Proxy * Servicios propios en Go, Python o C++ * Integración con motores de Machine Learning o Voice Activity Detection si aportan ventajas. El middleware deberá: * Interceptar sesiones SIP. * Procesar el flujo RTP. * Analizar el audio en tiempo real. * Detectar patrones característicos de buzón de voz. * Cortar automáticamente la llamada al detectar un buzón. * Funcionar tanto en: * Early Media (183 Session Progress). * Llamadas ya establecidas (200 OK). Deseable Se valorará experiencia previa en: * Answering Machine Detection (AMD) * SIP/RTP * Proveedores mayoristas de VoIP * Softswitches * FreeSWITCH...

...lógica a un entorno VoIP moderno basado en FreeSWITCH, manteniendo todo el manejo de llamadas y, de ser posible, mejorando su rendimiento. Busco a alguien que conozca a fondo FreeSWITCH y sus APIs (mod_dptools, ESL, Sofia-SIP, etc.), capaz de leer el código existente, analizarlo con apoyo de técnicas de IA y guiar la migración módulo por módulo. El objetivo inmediato es que las funciones esenciales de control de llamada se comporten igual que en el sistema original, pero sobre IP, sin depender ya del hardware NMS. Para asegurar que ambos estemos alineados, acepto el siguiente criterio de entrega: • Reporte de análisis del código C actual, señalando dependencias con la placa y equivalentes en FreeSWITC...

...antes de 200 OK Reglas basadas en: Diversion History-Info 181 Call Is Being Forwarded 302 Moved Temporarily patrones en Contact patrones en Request-URI headers específicos del carrier Logs de llamadas bloqueadas Motivo del bloqueo Estadísticas básicas Lista blanca / lista negra de carriers o prefijos Configuración editable por archivo o base de datos simple Fuera de alcance No AMD No Asterisk No FreeSWITCH No RTPengine No análisis de audio No grabación No transcodificación Entregables esperados Instalación de Kamailio en servidor ubuntu Archivo documentado Reglas anti-buzón configurables Pruebas con SIPp o tráfico real Documentación básica de operación Recomendación de servidor para...

Busco desarrollador VoIP + backend para montar una arquitectura con FreeSWITCH multi-cliente integrada a mi backend en NestJS y frontend ReactJS. Objetivo general Tener una PBX en FreeSWITCH multi-tenant donde: Cada cliente (tenant) tenga su propio domain (ej: ). Desde el backend pueda definir cuántas llamadas simultáneas permite cada cliente (ej: 3, 5, 10 canales). Todos los usuarios del cliente puedan llamar, pero respetando ese límite de canales. El frontend ReactJS se conecte vía WebRTC/SIP al servidor usando credenciales efímeras (no fijas). Stack tecnológico deseado FreeSWITCH (Ubuntu) configurado para: Multi-tenant (un dominio por cliente). Directorio dinámico vía mod_xml_curl (usuarios generados ...

Estamos buscando un experto en freeswitch para poder realizar un proyecto donde el voip es una parte importante. Buscamos talento para realizar consultoría.

Hola a todos estoy buscando alguien que me ayude con la terminación de mi bot de telegram solo nesecito terminar de configurar el sistema de llamada

Necesito Crear una interface Web C# mvc para recibir y Generar Llamados Utilizar WebRTC

buenas, e instalado astpp y le he echo todas las configuraciones, ahora tengo un problema y es cuando los clientes me mandan peticiones me cuelga el servidor, dejare la imagen anexa para que la vean

Somos una app para realizar llamadas internacionales y envío de mensajería de texto. Estamos en la búsqueda de un desarrollador o asesor que domine Ionic 4 para realizar integración para función de llamadas por internet con protocolo SIP y Freeswitch. La posición es para trabajar remotamente a tiempo parcial o brindar asesoría. Buscamos un profesional de alto nivel cuyo trabajo sea estructurado, en equipo y organizado. El trabajo será remoto, desde su casa, utilizando la herramienta de monitoreo de Hubstaff.

Requiero una persona con experiencia en el sistema de FreeSwitch "FusionPBX", para instalar en un server dedicado y configurar una seria de parámetros. Eventualmente cada mes se procederá a chequear que el server y los servicios están correctos.  Es indispensable conocer el FusionPBX ya que se necesita sobre todo activar la opción Multitenant, asi como colas de llamadas, IVR, BFL, y capturar llamadas.<br />Otro equipo llevará la getión diaria de los clientes, cuentas, y funcionamiento en general. <br />La finalidad de este proyecto es para la instalación y primeras configuraciones, asi como soporte eventual.

Instalación de Freeswitch-Fusionpbx en pcb Banana Pi, con Debian 8 ya instalado en tarjeta sdcard, sería a través de SSH ó teamviewer

, conocimiento u orientación profesional hacia el área de software libre y de código abierto (FOSS), administración de servidores Debian Linux, VoIP (Freeswitch), Moodle, desarrollo móvil, APIs, Think Client, herramientas de groupware, entre un profesional de tecnologías de información consciente de la importancia de la programación en Bash y herramientas orientadas a consola. En pocas palabras: amor por la "ventanita negra" o "línea de comandos" de UNIX o GNU/ de administración de servidores GNU/Linux (Debian), SSH, Shell y programación en Bash, aplicaciones en modo consola (vim, emacs, tmux, irssi, htop, etc.), configuración de redes y Thin Client, virtualización.4. ...

Necesito conectar una planta Asterisk a una Planta Avaya por medio de una troncal H323. En este momento tenemos problemas de señalizacion y se pueden generar llamadas via Asterisk -> Avaya pero no vie Avaya -> Asterisk. Si tiene conocimientos en el manejo de otra PBX como Freeswitch o Yate, se puede usar esta tecnologisa y subirlo como Gateway.

Tenemos una plataforma desarrollada en freeswitch que ofrece servicios de audioconferncia. Buscamos personas que nos apoyen en el mantenimiento de dicha plataforma.

Creacion de CRM usando Asterisk, Freeswitch, php y mysql.

...en servicios de comunicación de voz en mercados emergentes con un rápido crecimiento, ofreciendo a carriers calidad y alto rendimiento de la red a través de interconexiones directas. Estamos buscando a un Programador con experiencia que pueda crear aplicaciones para VOIP, conference calls, chat y streaming audio utilizando Asterisk y otras tecnologías de código abierto (por ejemplo OpenSIP, Freeswitch, SER, etc.) Responsabilidades: La persona elegida trabajará en un entorno dinámico y estimulante, desarrollando e implementando soluciones únicas y personalizadas para satisfacer las siempre cambiantes necesidades de la industria de las telecomunicaciones y a la vez aprovechar las oportunidades de mercado que se presentan. L...

...communications platform. The build must rely exclusively on open-source components and cover every real-time channel we use—VoIP, PSTN, instant messaging with presence, video calling and broader multimedia handling. Here is the software stack I want you to install, configure and prove in production readiness: • ejabberd for XMPP-based messaging and presence • Kamailio as the SIP registrar/router • FreeSWITCH acting as media server and application engine • A Class 5 softswitch layer for retail voice features • A Class 4 softswitch layer for carrier routing and LCR Everything will run on Linux. I expect clean source-based or package installs (your choice) that follow best security practices, plus well-commented config files stored in a versi...

I have a custom School Bell application that works perfectly on FusionPBX 4.x but no longer loads on the 5.x branch. The core job is to refactor whatever needs refactoring so the app installs cleanly and runs just as relia...between multiple schedules, and any new permissions needed • Proof-of-concept demo on a 5.x test instance (screen-share or short video is fine) Acceptance criteria: the app installs without errors, the original single schedule runs correctly, the new schedule selector works, and no existing dialplan entries are broken. If you’ve already migrated modules between FusionPBX versions or worked with FreeSWITCH APIs, that’s a bonus. Let’s keep the communication tight; once we agree on the approach, I’m ready to hand over repo access an...

I already run FusionPBX and have the full source code for a pre-existing school bell application that is known to work with it. Your task is to install that app on my live server, compile or otherwise build it where necessary, register it inside the FusionPBX menu, and confirm that scheduled bells trigger correctly through FreeSWITCH. Once the bell app is fully functional, replace every instance of the default FusionPBX logo (login page, top-left dashboard, and favicon) with the PNG I will supply, making sure cache issues are cleared so the new branding shows immediately for all users. Deliverables • Bell app installed, visible in the FusionPBX GUI and surviving reboots • Custom logo applied system-wide without layout issues • Short summary of the steps taken ...

I need a turnkey IPRN/VoIP switch built by someone who has done it before in a live environment. The project covers the full stack—Carrier SIP interconnections, dynamic routing with LCR, real-time billing and CDR, multi-level (admin / reseller / client) panels, plus SMS and voice interconnect. FreeSWITCH is fine, as is any similar carrier-grade platform you can prove you already mastered. Please include solid proof of your past work with an operational IPRN system—screenshots, a demo URL, or a brief portfolio spotlighting the project. Applications that lack demonstrable past work will be ignored. Timeline is flexible; stability and feature-completeness take priority over speed. Deliverables I will assess on hand-over: • Fully deployed IPRN switch with documente...

...SIP integration, native mobile wrappers (or React Native / Flutter, if you prefer), and a responsive web interface. Core functions I need delivered: • HD video calls with minimal latency • Built-in screen sharing during any call session • One-click call recording stored reliably in the cloud (with simple playback and download options) Everything—from the media servers (Kurento, Jitsi, FreeSWITCH, or another proven stack) through to the frontend UI—should be production-ready, secure, and scalable. I’ll rely on you for server configuration, NAT traversal, encryption (TLS / SRTP), user authentication hooks, and a clean API that lets me extend features down the road. If you have a proven track record making WebRTC or SIP endpoints talk flaw...

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...the way from concept to a production-ready product built on FreeSWITCH. The core platform must run reliably on both Windows and Linux, and I need full documentation that lets any future engineer understand, maintain, and extend the system. Key functionality I expect from day one includes IVR menus that can be edited without code changes, crystal-clear call recording with searchable storage, multi-party conference calling, high-volume voice broadcasting, and basic voice recognition for simple spoken commands. Whatever libraries or modules you choose inside FreeSWITCH are fine, as long as they are open-source friendly and you can explain why you picked them. Front end and Backend we have to team to work on it. Need support on Freeswitch integration. Because this is...

...technician who knows FreeSWITCH inside-out to set up an automated voice-broadcasting system that will deliver a single marketing message to a list of contacts I will supply. The list arrives clean and ready for import, so your focus is the server build, dial-plan logic, and campaign workflow, not data cleansing or list generation. Here’s what I’m after: the call engine must pick numbers from my CSV, play one pre-recorded promotional message, handle basic call outcomes (live answer vs. voicemail) and generate a concise completion report with call status statistics. No per-recipient personalisation is required; the same audio goes to everyone, keeping the logic straightforward. I already have a VPS available; you are welcome to recommend tweaks to its specs if Free...

My in-house contact-centre stack is almost ready: a React softphone running hands WebRTC traffic to FreeSWITCH, which in turn should hand the call to an Airtel SIP trunk for delivery to the PSTN. In the lab everything looks clean, yet the moment we hit real Indian numbers the call stalls—no ringing, no media—so we know the problem sits squarely in the SIP signalling with Airtel, not in registration or dialplan execution. I already have: • Stable WebRTC leg from the browser to FreeSWITCH • Dialplans and Lua scripts that work for internal tests • Airtel trunk registered and authenticated What I need is an engineer who can jump onto the server, fire up sngrep / sofia trace, spot what Airtel is rejecting (the call currently never connects), and p...

...1, then plays the second MP3 and ends the call. • When all numbers have been processed the bot returns a Telegram summary listing each number that was successfully called. Scope specifics you must cover • Fix the current issue: “Bot not making calls.” • Restrict outbound traffic to USA numbers only; no international dialing should be possible. • Any voice gateway (Twilio, Asterisk, FreeSWITCH, etc.) is fine so long as you wire it cleanly to the Telegram Bot API and document credentials/config in an env file. • Concurrency should be configurable so I can tune call throughput without hitting carrier limits. Deliverables 1. Fully working Telegram bot (source code) that follows the flow above. 2. Simple README covering setup, e...

...regulations (TRAI/DoT). 2. Objectives • Automate bulk calling campaigns with minimal manual intervention. • Deliver pre recorded or text to speech greeting messages. • Provide a simple dashboard for campaign management, monitoring, and reporting. • Ensure scalability and reliability for large call volumes. 3. Scope of Work • System Setup: o Deploy open source dialer software (e.g., Asterisk, FreeSWITCH, VICIdial, GoAutoDial) or propose alternatives. o Integrate with SIP trunks/VoIP providers. • Features Required: o Upload and manage contact lists (CSV/Excel). o Schedule and batch calls to avoid congestion. o Play greeting messages (audio file or AI TTS). o Real time dashboard for call status (answered, busy, failed). o Reporting and analytics (delive...

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 e...

...campaigns must dial through the same platform so agents work from a single interface, with those calls recorded in the same repository. Scope of work You’ll deploy, configure, and hand over a production-ready IVR. I expect guidance on menu design, prompt management, real-time monitoring and a clean dashboard where my team can adjust routing rules without touching code. I’m open to Asterisk, FreeSWITCH, Twilio, or another stack—just explain why it’s the right fit for reliability and scale. Acceptance criteria 1. Three-level menu tested end-to-end with live callers. 2. Call recordings saved in .wav or .mp3, downloadable from the dashboard. 3. Outbound dialler integrated and functional. 4. Admin user can add prompts, edit menus, and reroute ca...

I need a fully-functional auto dialer built for my company and I want it up and running fast. The core requirement is seamless VoIP ser...with the VoIP provider I’ll share once we start, including proper authentication and fail-over handling. • A clean, web-based interface where agents can log in, see their queue, and record basic call outcomes. • All source code plus clear deployment and user documentation so my in-house tech team can maintain it afterward. If you’ve built dialers before—especially with Twilio, Asterisk, FreeSWITCH, or similar stacks—let me know what framework you recommend, the timeline you can commit to, and any additional features you can add (call recording, automated messages, analytics, etc.). I’m ready to move q...

We are seeking an experienced development with experience in SIP, Kamilio, FS and other telephony environments, must be able to build custom CPaaS (Communications Platform as a Service) platform. The system will provide a complete voice infrastructure layer, custome...Kamilio, FS and other telephony environments, must be able to build custom CPaaS (Communications Platform as a Service) platform. The system will provide a complete voice infrastructure layer, customer portal, developer APIs, and SDKs enabling businesses and developers to build voice-enabled applications and manage telephony services. The platform will be built using Kamailio as the SIP proxy/SBC, FreeSWITCH as the media server, and a modern web application stack to deliver scalable, secure, and enterprise-grade voice...

I’m looking for a patient, technically-savvy trainer who can take me from my current beginner level to confidently performing advanced troubleshooting and maintenance on both FreeSWITCH and OpenSIPS. Scope of the masterclass • Begin with a quick foundation to be sure my basics are solid. • Move rapidly into deep-dive modules on: – Diagnostics and logging (call traces, SIP capture, fs_cli, Homer) – Security best practices (TLS, ACLs, antifraud rules, topology hiding) – Performance optimization (load balancing, caching, tuning Sofia/SIP and dispatcher settings) • Emphasise hands-on configuration: I want to work through real scenarios, replicate common failures, and fix them live. • Provide recorded sessions, editable lab...

IPRN voice switch FreeSWITCH Server/Dashboard setup Implementation of Systems CALLS only version Setup

I need to add real–time Australian driver’s-license verification to my existing VOIP platform immediately. Callers will enter or speak their licence details during the call flow; the system must then query the official Australian DVS (or a comparable API you recommend) and return an instant pass/fail result be...is maintainable in-house after hand-over. Deliverables • Fully integrated verification module with source code • End-to-end test script showing successful matches and rejection handling • Brief setup notes so my team can redeploy the service if needed Time is critical; please outline how quickly you can connect to the DVS, what libraries or SDKs you plan to use (e.g., Asterisk AGI, FreeSWITCH ESL, Twilio Voice, or ), and any prerequisit...

I have a fresh FusionPBX install that i need help to register with my SIP provider. I have not yet filled in any trunk settings because I am unsure which parameters the carrier needs and where each item belongs inside FusionPBX. ...• Tell me exactly which credentials and network details to request from the provider (user ID, auth name, password, proxy, outbound proxy, codecs, etc.). • Remotely configure the trunk in FusionPBX once those details are in hand. • Prove the registration is solid and that I can place and receive at least one test call without errors. Please work directly in the FusionPBX web interface (FreeSWITCH‐based) and keep a quick note of every change you make so I can replicate it later if needed. Once registration and test calls succeed, the...

We are building a real-time voice agent using Elevenlab's WebSocket API and need an expert in Freeswitch to help us bridge our current call flow to the Elevenlab's voice agent. Requirements:
 - Proven experience with Freeswitch core and module development
 - Experience with `mod_audio_fork` and `mod_audio_stream`
 - Deep understanding of SIP/RTP/media flows What you will do:
 - Connect our existing Freeswitch server with Elevenlab's WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured. 
 Enable seamless, real-time, bi-directional audio between the caller and the voice agent
. Stream audio to Elevenlab in real-time and handle incoming transcription/command messages. Maintain high availability and ...

...timings, status, due payments, support requests) • Make automated outbound calls (appointment confirmations, reminders, lead follow-ups, collections) • Transfer the call to a human agent when required This is NOT a keypad IVR. The assistant must understand natural spoken language from callers and respond with a natural-sounding voice. Technical Scope: • SIP/VoIP integration using Asterisk or FreeSWITCH • Real-time Speech-to-Text • Text-to-Speech voice responses • LLM/NLP based conversation handling • Call recording and call logs • Basic web admin panel Budget: The genuine project budget is as mentioned in the posting. It may be extended if required based on technical justification and developer recommendations, subject to management appr...

...Optimize OS for: Low-latency audio High concurrent calls Firewall configuration: SIP (TLS) RTP port range Internal API access only SIP Trunk & Telephony Configuration Configure SIP Trunk (provider details will be shared) SIP over TLS preferred Handle: Incoming & outgoing calls NAT traversal DTMF Call start / end events Codec setup: Opus (primary) G.711 (fallback) Telephony stack: FreeSWITCH / Kamailio / OpenSIPS (freelancer must justify choice) Media & Call Flow Handling Handle RTP audio streams in real time Call flow: Receive caller audio Stream audio to STT Receive AI response Convert to TTS Play back audio to caller Support: Voice Activity Detection (VAD) Silence detection Caller barge-in (interrupt AI speech) STT (Speech-to-Text)...

...an experienced VoIP / FreeSWITCH Engineer to work on FreeSWITCH open-source v1.10.2 implementation and configuration. The ideal candidate should have strong hands-on experience with SIP signaling, video calling (H.264), SIP gateways, and media handling. You will be responsible for configuring FreeSWITCH to support video, resolving connectivity and codec issues, and implementing SIP to RTMP recording/transcoding. Responsibilities Configure and troubleshoot FreeSWITCH v1.10.2 (open source) Enable and optimize video calling using H.264 codec Configure and manage SIP gateways and SIP interoperability Implement SIP to RTMP recording and video transcoding Debug SIP, RTP, media, and codec-related issues Ensure stable audio/video performance Required Skills ...

I run a small Asterisk lab and I need to run controlled test calls where the caller ID can be set to any value I choose. The goal is strictly testing and development, so e...live demo—screen-share or recorded session is fine—so I can see a test call leave the PBX and arrive with the desired caller ID. • Hand over a concise checklist I can reuse when I spin up new instances of the lab. I’m already comfortable inside the Asterisk CLI and with basic SIP debugging, so please focus on the caller-ID manipulation specifics rather than generic PBX setup. If you prefer FreeSWITCH or a hosted solution, mention why it would make the job easier, but the delivered instructions must work on raw Asterisk. Looking forward to a clean, reproducible solution I can drop into ...

Expert VoIP Engineer Needed: Multi-Tenant PBX Deployment (FusionPBX / FreeSWITCH Preferred) Project Description We are an IT Managed Service Provider (MSP) looking to build a robust, scalable, white-label VoIP platform to host phone systems for multiple distinct clients. We are looking for a senior VoIP engineer to deploy, configure, and secure a True Multi-Tenant PBX System. Important Architectural Requirement: We are NOT interested in a single-instance FreePBX installation hacked with custom contexts. We require a system designed for multi-tenancy from the ground up to ensure strict data isolation and security between clients. FusionPBX (FreeSWITCH) is our preferred platform, though we are open to VitalPBX (Carrier Edition) or Kazoo. Key Deliverables * Multi-Tenant Archite...

I need an experienced VOIP and SIP Engineer. I have developed a custom AI Voice Calling Bot and it is currently connected with Twilio. Whereas my plan is to connect this AI Bot with almost any SIP provider. When i call from Twilio using my AI Bot, then it works correctly. Here i am needing you 1. I have installed Asterisk on my AWS. It is fully configured and making outbound calls. 2. My AI Calling Bot works with web hooks, and I have created another instance where I have created an outbound webhook in Node. And the AI Calling bot's webhook is placed in this code 3. When I call from Asterisk, the outbound call works, but there is a sharp noise in the call, and nothing else. My AI Bot should be listening in the call, but i only hear a sharp noise 4. I know this is due to mis-samp...

I need a fully-functional auto dialer built for my company and I want it up and running fast. The core requirement is seamless VoIP ser...with the VoIP provider I’ll share once we start, including proper authentication and fail-over handling. • A clean, web-based interface where agents can log in, see their queue, and record basic call outcomes. • All source code plus clear deployment and user documentation so my in-house tech team can maintain it afterward. If you’ve built dialers before—especially with Twilio, Asterisk, FreeSWITCH, or similar stacks—let me know what framework you recommend, the timeline you can commit to, and any additional features you can add (call recording, automated messages, analytics, etc.). I’m ready to move q...

I have a fresh Linux server ready and need the latest stable release of FreeSWITCH installed, secured, and connected to my SignalWire space. Beyond a plain install, I also want the common advanced pieces in place—proper codec support (Opus, G.729, etc.), a clean dialplan template I can extend, and any SIP profiles necessary for SignalWire’s endpoints. Once the service is up, please register it to SignalWire, verify inbound and outbound calling, and leave me with clear notes on anything customised (modules enabled, directory changes, CLI commands). Deliverables • FreeSWITCH latest stable compiled or packaged and running on my Linux server • Advanced configuration applied: codecs loaded, base dialplan, SIP/TLS where applicable • SignalWire c...

I have a brand-new Debian box that is still factory-fresh, and I want it running a clean, production-ready FreeSWITCH instance that’s already talking smoothly to SignalWire. My immediate priorities are: • Compile or package-install the latest stable FreeSWITCH build on Debian • Enable and test the Conference calling module (no voicemail or faxing required) • Perform all network and SIP/WebSocket configurations so the server registers with my SignalWire space, routes calls correctly, and survives reboots I’ll need you to handle the firewall rules, TLS certs, and any NAT or port-forward tweaks along the way, then document what you’ve changed so I can keep it maintained. A short test plan proving inbound and outbound conference calls thro...

...compact, standalone system where all components run on a single Virtual Machine for R&D and commercial testing purposes. The project also includes the configuration of WebRTC and the preparation of a White-Label Mobile Softphone (based on Linphone) fully integrated with this system. Scope of Work: 1. Single Node Kazoo Infrastructure (Server-Side): All-in-One Architecture: Installation of Kazoo, FreeSWITCH, Kamailio, RabbitMQ, BigCouch/CouchDB, and RTPEngine on a single VM within Proxmox. Network & NAT Traversal: Proper configuration of IP, Ports, and ACLs to ensure seamless RTP/Signaling flow behind Proxmox NAT. Multi-Tenant Capability: Although it is a single node, it must be configured to support multiple resellers/companies (Multi-Tenant). SIP Trunking: Setup and...

...compact, standalone system where all components run on a single Virtual Machine for R&D and commercial testing purposes. The project also includes the configuration of WebRTC and the preparation of a White-Label Mobile Softphone (based on Linphone) fully integrated with this system. Scope of Work: 1. Single Node Kazoo Infrastructure (Server-Side): All-in-One Architecture: Installation of Kazoo, FreeSWITCH, Kamailio, RabbitMQ, BigCouch/CouchDB, and RTPEngine on a single VM within Proxmox. Network & NAT Traversal: Proper configuration of IP, Ports, and ACLs to ensure seamless RTP/Signaling flow behind Proxmox NAT. Multi-Tenant Capability: Although it is a single node, it must be configured to support multiple resellers/companies (Multi-Tenant). SIP Trunking: Setup and...

...(SEE ATTACHED IMAGE FOR MORE) By using the Hybrid Approach, you are essentially using FreeSWITCH as your logic and media engine (to handle WebRTC and call control) and Twilio merely as the carrier (to bridge calls to the PSTN). This approach drastically reduces costs (Twilio Elastic SIP Trunking is significantly cheaper than the Twilio Client SDK) and gives you granular control over the call audio. The Architecture Blueprint 1. React Frontend: Uses (or similar) to connect to FreeSWITCH via WebRTC (WSS). 2. FreeSWITCH (Middle Layer): Acts as the PBX. It bridges the WebRTC stream (from the browser) to a standard SIP stream. 3. Twilio (Carrier Layer): Connected to FreeSWITCH via Elastic SIP Trunking. It takes the SIP stream and term...

Incoming calls reach the phones, Intermittent issues such as one-way audio where caller can hear us but we cannot be heard. The signalling path is Kamailio acting purely as the control-plane SIP proxy, then the media is anchored by a FreeSWITCH + Asterisk B2BUA cluster. Only inbound legs show the problem; outbound audio is clean. I need a seasoned VoIP troubleshooter to: • trace SIP and RTP on all hops (Kamailio, FreeSWITCH, Asterisk, edge SBC) • pinpoint why RTP from the caller side never makes it to the far end (NAT, codec negotiation, rtpengine mis-pinning, firewall, wrong c= line, etc.) • supply the minimal configuration changes or firewall rules to restore full two-way audio without disrupting live traffic Acceptance will be: 1. SIP packet ex...

Hi, I'm looking for an experienced VoIP developer to build a complete multi-tenant hosted PBX / UCaaS platform with real-time billing and a modern client portal. The system should be scalable, secure, and production-ready for a hosted PBX reseller business. Tech Stack & Requirements: Core PBX: FreeSWITCH + FusionPBX (latest version) True multi-tenant setup with domain isolation Superadmin + tenant admin access levels Enterprise features configured: ACD/Call Queues (strategies, agent states, callbacks, wallboards) IVR, Ring Groups, Time Conditions, Call forwarding with CC/CAP Call Recording, Conferencing, Voicemail-to-email Secure WebRTC (wss) for browser-based calling Call Center modules and reporting High availability & security (Fail2Ban, iptables, SSL) Real-Time B...