Freeswitchtrabajos

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...

...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...

...with flash-based storage systems Linux system optimization for embedded hardware VoIP / SIP / Networking Experience implementing or maintaining SIP-based communications systems Knowledge of RTP / RTCP media streaming Familiarity with VoIP codecs such as G.711, G.729, Speex Understanding of NAT traversal, STUN, QoS (DSCP), and SIP registration Experience integrating with PBX systems (Asterisk, FreeSWITCH, etc.) TCP/IP networking, DHCP, DNS, NTP Audio Processing Experience working with real-time audio streaming Knowledge of ALSA or similar Linux audio frameworks Audio mixing, buffering, and jitter control Experience with microphone, speaker, and audio codec hardware Signal processing basics (tone generation, filtering) Embedded Hardware Integration Experience interfacing with G...

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...

Project Description We are looking for a highly experienced development team or company to build a complete Inbound Call Tracking Software based on FreeSWITCH. Only developers or companies who have already worked on similar call tracking or telecom software projects using FreeSWITCH should contact us. This is a full-cycle project, and the selected team must be capable of delivering: Complete backend & frontend FreeSWITCH integration Stable, scalable, and production-ready solution Proper documentation and deployment support What Is Inbound Call Tracking Software? Inbound Call Tracking Software is a system that allows businesses to track, monitor, analyze, and optimize incoming phone calls from multiple marketing sources such as: Google Ads Facebook Ads We...

I'm looking for an experienced FreeSWITCH developer to create a robust VoIP solution. Key Features: - Call Routing - SIP Trunking - Call Recording Ideal Skills: - In-depth FreeSWITCH expertise - VoIP development experience - Strong background in call routing and SIP protocols - Familiarity with call recording technologies Please share relevant experience in your application.

I want a production-ready FreeSWITCH-based Session Border Controller that registers or peers with my clients’ PBXs by username and password, transcodes to Opus, and then hands the calls off to two preset wholesale carriers. The core logic is: • Country-level routing priorities stored in MariaDB.  – Example: US/Canada → Carrier 1 first, fail over to Carrier 2; EU → Carrier 2 first, fall back to Carrier 1 on 404 or no answer. • Each client may present several PBXs and DIDs, all of which must map cleanly to those database rules. • Carrier trunk details stay hard-coded in the FreeSWITCH XML/JSON configs; only the client and route data live in MariaDB. • CDR I also need a small web interface (PHP, Python Flask or a similarly...

...Machine Learning Engineer specialized in audio processing and deep learning. The goal is to design, train, and deploy a high-performance AMD (Answering Machine Detection) model for telephony, using an existing dataset of approximately 67,000 labeled audio samples. The model must operate in real-time with low latency, and integrate into our existing calling infrastructure (Drachtio / Asterisk / FreeSWITCH / Vicidial). Mission Responsibilities: Analyze and preprocess the existing dataset (cleaning, balancing, train/val/test split) Extract audio features such as Mel-spectrograms, MFCC, STFT, normalization Design and train a CNN/CRNN model for AMD classification (Human / Voicemail / Silence / Fax / Other if needed) Optimize the model for real-time inference (target <200 ms de...

...receive calls with the reliability of a carrier-grade PBX. The server must handle: • Incoming and outgoing calls • Call forwarding and call transfer • Voicemail storage/retrieval • A flexible auto-attendant (IVR) My preference is to stay in the React Native ecosystem for the client side, but I’m open to your guidance on the most appropriate SIP/WebRTC stack, media server (Asterisk, FreeSWITCH, Kamailio, etc.), and signalling approach. Please outline the architecture you propose, the main tech you’d employ, and an estimated timeline for delivering a first working build that can: 1. Register soft-phones via SIP or a comparable protocol 2. Complete internal and external calls with the features above 3. Expose a clean REST/GraphQL API...

...real time. 2. An outbound lead-qualification scenario dials a provided number, carries a short scripted conversation, and posts the outcome back to the CRM. 3. Audio quality and speech latency remain below 300 ms round-trip on our internal network. 4. All components run behind our firewall with environment-specific configuration files. If you already have experience with SIP, Asterisk/FreeSWITCH, Node.js or Python micro-services, and either OpenAI or Google PaLM APIs coupled with ElevenLabs, you’ll get up to speed quickly. I can provide access to our CRM endpoints and a test SIP trunk as soon as we agree on the implementation plan....

...steps - Basic configuration - How to verify the system is healthy - How to restart and manage core services --- ## Required Experience You should already be comfortable with the Jambonz ecosystem, specifically: ### Core Jambonz Services - `jambonz-api-server` - `jambonz-webapp` - `sbc-call-router` - `sbc-inbound` - `sbc-outbound` - `sbc-registrar` - `jambonz-fsw` (FreeSWITCH) ### Dependencies & Infrastructure - **Drachtio** (SIP server) - **RTP Engine** (media proxy) - **MySQL** - **Redis** (caching) - **Node.js** (runtime) - Proper configuration of **HTTPS** and **WSS** for WebSocket signaling --- ## Application Instructions When you apply, please briefly describe: - Your previous Jambonz / Jambonz Mini projects - The envi...

...looking for a Go-savvy FreeSWITCH specialist who can dive straight into a stubborn WebRTC issue that’s crippling call stability. The goal is simple: identify the root cause, patch it cleanly, and leave my stack handling WebRTC calls as smoothly as it does SIP. The platform is already live, written largely in Go, and the problem shows up under moderate load—call setup stalls or drops mid-stream whenever WebRTC endpoints join. I’ll grant you SSH access to the FreeSWITCH node, relevant Go modules, and recent logs so you can reproduce the fault. What I expect from you • A concise but detailed proposal outlining your troubleshooting plan, diagnostic tools you lean on (e.g., sngrep, fs_cli, Wireshark), and an estimated timeline. • Clean, well-co...

I am looking for a freeswitch expert who can configure my freeswitch to use XML API. If you know how to configure then bid with your hourly rate

I am looking for a skilled developer to create a web application for automating Airtel LAPU recharges using an API. The application should streamline the recharge process and provide a comprehensive management system. Key Requirements: - Develop a web-based application for Airtel LAPU recharge automation. - Implement features for recharge management, transaction history, user authentication, and balance addition. - Ensure secure and efficient handling of user data and transactions. Ideal Skills and Experience: - Proficiency in web application development. - Experience with API integration, particularly for telecom services. - Strong understanding of user authentication and data security. - Ability to create intuitive and user-friendly interfaces. I am eager to collaborate with a develop...

...SIPREC protocols Configure and manage media servers (FreeSWITCH, Asterisk, RTP Proxy) Work with SIP proxy servers (Kamailio, OpenSIPS) for call routing and signaling Build and maintain call recording solutions using SIPREC Handle WebRTC, RTP streams, and VoIP media processing Develop automation scripts and services using Node.js, Lua, or Python Integrate with relational databases (Postgres, MySQL) Deploy and manage solutions in cloud-native environments (GCP preferred; AWS, Azure) Ensure high availability and scalability using HAProxy or load balancers Collaborate with cross-functional teams for deployment, monitoring, and troubleshooting Required Skills: Hands-on experience with SIPREC for VoIP call recording Expertise in FreeSWITCH / Asterisk / RTP Proxy (med...

...users and admins. • Self-service user registration plus zero-touch auto-provisioning for common SIP endpoints. • End-to-end subscription handling that mirrors the advanced flows found in Zelle and CashApp. • A single “one-click” script that generates every required configuration file, module, and certificate when spinning up new tenants or nodes. Deliverables 1. Kazoo (with Kamailio, FreeSWITCH, RabbitMQ, BigCouch, and HAProxy) installed and clustered across all eight nodes. 2. Call-center queues, agent log-in/out, live wallboard, and termination routes fully operational. 3. HA and fail-over verified; a node loss must not drop active calls. 4. Load test proof of 50 k simultaneous calls meeting agreed PDD and packet-loss targets. 5. Ha...