Freeswitchemplois

Déploiement & Configuration Expert FusionPBX / FreeSWITCH en Haute Disponibilité Description détaillée : Nous recherchons un expert en VoIP spécialisé dans FusionPBX / FreeSWITCH pour déployer, configurer et optimiser une infrastructure haute disponibilité. Le projet vise à gérer des appels simultanés avec équilibrage de charge intelligent (Load Balancing). Missions principales : Installer et configurer FusionPBX/FreeSWITCH sur plusieurs serveurs. Mettre en place un Load Balancer SIP intelligent (natif FusionPBX ou externe Kamailio/OpenSIPS). Assurer l’intégration parfaite avec une plateforme Vicidial existante. Garantir la sécurité et la robustesse d...

besoin d'avoir un script php(perl/lua) pour créer un bridge en freeswitch pour initialiser un appel. si on appelle le script avec un numéro de téléphone alors on crée un appel entre une extension (3000) et le numéro appelé

j'ai besoin de installer un serveur opensips avec une proxy rtp (rtpproxy) + serveur Freeswitch le tout avec debian jessie x64 comme sur le plan: Merci

Besoin des competneces pour des scripts lua avec freeSwitch et avec le mode SMS et chat

...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...

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 ...

...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...

*** PLEASE READ IMPORTANT NOTE *** ONLY BID ON THIS PROJECT IF YOU ARE EXPERIENCED IN WEB APP DEVELOPMENT AND FREESWITCH INTEGRATION, PLEASE DO NOT WASTE BOTH OF OUR TIME IF YOU ARE DO NOT HAVE PRIOR EXPERIENCE IN WEB APP DEVELOPMENT AND FREESWITCH. I’m building a browser-based virtual number buying platform and need the voice layer hooked up to FreeSWITCH. The web app itself still has to be put together, so the project covers both the application stack and the telephony integration. Core scope • Build a responsive web interface where users can register, log in, buy numbers, configure calling features, see their contact list, and place or receive calls on mobile app and PC browser. • Perform SOFIA Mod api integration and enable real time data sync an...

I’m putting together a Freeswitch based number buying platform and need a developer who is comfortable wiring FreeSWITCH into a modern web stack. The core use-case is simple: authenticated users must be able to signup buy virtual numbers, configure various basic call settings on these numbers. Freeswitch will be used to control call functions. Here’s how I picture the flow: • A lightweight web interface (React, Vue or vanilla JS—whatever you work fastest in) connects to FreeSWITCH via WebRTC/SIP. -Any backend DB. -Freeswitch integration for telephony • Users signup, log in, buy number, configure settings, start using it. I already have a clean VPS ready for deployment; you are welcome to spin up containers or go bare-meta...

I’m running FreeSWITCH in production and need an experienced engineer to step in, review the current configuration, and ensure everything is running at peak performance. The first milestone is a quick screen-share so you can examine the dialplan, logs, and network environment. After that, I’ll rely on you to propose best-practice adjustments, implement any agreed changes, and document them clearly so I can maintain the system going forward. Solid command of SIP signalling, XML dialplans, ESL, and Lua scripting is essential; please reference similar FreeSWITCH work you’ve completed and let me know your typical turnaround time. If we uncover additional needs—such as integrating with a database, a CRM, or a new SIP trunk—you should be comfortable...

...your own words. I’m a PHP developer building a system to connect FreeSWITCH (Debian 12) with an external PHP/MySQL API using mod_xml_curl. I’ve already: Installed FreeSWITCH on a Debian 12 VPS Created a MySQL table called sip_accounts with username and password fields Built a PHP API that returns FreeSWITCH-compatible XML from that database Right now, SIP registration from my softphone is failing. The FreeSWITCH CLI shows XML-related errors, so there’s likely a configuration issue. I’m looking for someone who has full FreeSWITCH configuration knowledge and strong Linux experience to help me fix this setup. You don’t need to work on PHP or MySQL — I’ll handle that part. Your job is to: Guide me through ...

We are looking for an experienced VoIP developer with deep expertise in Asterisk or FreeSWITCH to build a custom multi-tenant PBX web portal from scratch. The platform should allow multiple tenants (resellers/customers) to manage their own PBX systems independently — including extensions, SIP trunks, DIDs, call routing, CDRs, billing, and reporting — all accessible through a secure, modern web interface with tenant-level branding. Our goal is to create a white-label PBX SaaS solution similar to FusionPBX / Thirdlane, but with our own branding, design, and extended billing features. Key Features: Tenant Management – Create/edit/delete tenants, assign resources, and define roles (Admin, User, Reseller). Domain/Subdomain Based Access – Each tenant should ac...

I need a skilled FreeSWITCH engineer to set up a new FreeSWITCH server (Debian/Ubuntu) where all configurations — users, dialplans, gateways, and DIDs — are stored in MariaDB instead of XML files. What I need: Install and compile the latest stable FreeSWITCH on my VPS Set up MariaDB and create the necessary tables/schemas Configure FreeSWITCH to read and write everything from the database Add Lua dialplan scripts so routing logic runs from Lua and database references Set up two working routes: One SIP trunk DID (inbound/outbound) One IP-auth DID (no registration, direct routing) Provide clear SQL examples or docs so I can easily add new users, DIDs, and routes later Acceptance: I can log in with a softphone Inbound/outbound calls work f...

...up instantly. Skills Required: • Strong experience in real-time audio processing (WebRTC, RTP, SIP audio streams, or equivalent). • Proficiency in speech and signal processing (e.g., VAD, MFCC, spectral analysis). • Machine Learning/Deep Learning for audio classification. • Experience with latency optimization in streaming systems. • Familiarity with telephony protocols (SIP, Asterisk, FreeSWITCH, etc.) is a strong plus. • Python/Node.js/Go/C++ (any language capable of handling low-latency audio). Deliverables: • Real-time streaming AMD system (replace current download method). • Early decision logic with configurable thresholds. • Integration of new detection features (synthetic voice, music, bit verification). • ...

I need a seasoned engineer to build and fine-tune a WebRTC-to-SIP gateway that lets browser clients place and receive calls on my existing SIP infrastructure. The core goals are seamless audio and video negotiation, robust NAT traversal, and production-grade security. Here’s what I’m expecting: • A working gateway service built from proven components (for example Janus, FreeSWITCH, Kamailio, or any combination you recommend) that accepts WebRTC calls over WebSocket and bridges them into standard SIP trunks. • Full support for ICE, STUN/TURN, DTLS-SRTP, and proper codec handling so calls connect reliably across networks and devices. • Clear, well-commented configuration files, deployment scripts (Docker or bare metal), and a concise technical guide t...

...ongoing basis. The role involves scripting and configuration for FreeSWITCH, including PBX features and media server integrations. You’ll work alongside our internal team and assist with troubleshooting and implementing features as needed. Responsibilities: Develop and maintain dialplans using XML for FreeSWITCH. Write and update Lua scripts for custom API integrations. Configure and optimize PBX settings (IVR, call routing, voicemail, call recording, etc.). Troubleshoot SIP, RTP, and codec-related issues on the media server. Collaborate with internal developers to support ongoing projects. Provide ad hoc support (a few hours per week/month) for new features or bug fixes. Requirements: Proven experience with FreeSWITCH configuration and administration...

I’m looking for an experienced VoIP engineer to build a small-scale yet dependable admin portal that lets me offer USA toll-free calling from overseas. SIP is the protocol I want throughout the stack. Core scope • Set up a VoIP server (FreeSWITCH, Asterisk, or another SIP-compatible platform) and obtain or integrate at least one working USA toll-free route. • Configure call management and routing rules so I can quickly add, edit, and remove destinations. • Enable call monitoring and recording with searchable logs and downloadable files. • Provide a clean web-based admin interface where I can view active calls, route status, and basic statistics. • Secure the system (firewall rules, strong auth, TLS/SRTP where practical) and document all cred...

Make a production FusionPBX/FreeSWITCH fully operational for international outbound (via DIDWW and IDT Express) and inbound (via DIDWW), with endpoints using TLS 5061 + SRTP. Keep current hardening (Fail2Ban, firewall, Zero Trust on GUI) intact. Environment FusionPBX/FreeSWITCH on Debian/Ubuntu (public VPS) Primary tenant context is the domain used by endpoints (same hostname as TLS cert) Let’s Encrypt TLS installed; WSS/HTTPS OK Firewall already restricts to 5061/TCP and RTP 16384–32768/UDP (UDP 5060 is blocked) Security: Fail2Ban, Zero Trust (for GUI) Softphones: Linphone / MicroSIP / Zoiper (must register via TLS 5061 and place calls) Current Issues Phones register over TLS but outbound calls from phones fail (likely CLI/From & context/routing) DI...

...engine must run on FreeSWITCH, while the front end can rely on standard WebRTC/JavaScript stacks that play nicely with it. Here is what I need from you: • Set up and configure FreeSWITCH (including any required SIP profiles, TLS certificates, and WebSocket modules) so it can handle secure, low-latency voice traffic. • Develop the web app client: Build a web application framework with calling features • Implement real-time call status updates—ringing, active, ended—through WebSocket events. • Provide concise deployment documentation so I can reproduce the environment on a fresh Linux VPS. I already have the server and DNS ready; you bring the code, configuration, and Freeswitch and Web Appc development expertise to glue it toget...

We are building a real-time voice agent using Deepgram's WebSocket API and need an expert in Freeswitch to help us bridge our current call flow to the Deepgram 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 Deepgram’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 Deepgram in real-time and handle incoming transcription/command messages. Maintain high availability an...

...are essential, so please bring experience with suitable codecs, QoS and trunk selection. I also need complete CDRs: date-time stamp, duration, and the location identifier that handled the call. A browser-based GUI should let me view real-time call-duration statistics, call-volume by department, and update forwarding rules without digging into the command line. You are free to suggest Asterisk, FreeSWITCH, Kamailio or any comparable open-source stack; once I know the specs, I will provision the server on my side and point the toll-free SIP trunks at it. Deliverables • A fully configured VoIP / SIP server (or automation scripts) ready to drop onto my hardware • Web dashboard providing the reporting views and rule-management tools described above • Clear, concis...

I want to prove out an all-IP call-handling flow that lets an AI...and information about how to call without hiding the number. Deliverables • Deployed PoC reachable from a public swedish test number • Source code and any configuration or build scripts • Simple web admin/dashboard plus audit log storage • Step-by-step setup notes so I can reproduce the environment Feel free to leverage Twilio Programmable Voice, Amazon Connect, SignalWire, FreeSWITCH, Asterisk—whatever lets you move quickest while keeping everything IP-based. My main requirement is that the dialog relies on true natural language understanding, not just DTMF menus or keyword spotting. If you have relevant telecom experience and can stand up the demo quickly, let’s ...

I need a clean, source-based installation of the latest stable FreeSWITCH (with the call-center and any other modules required for a contact-centre scenario) on a Linux server. Right beside it, the latest stable OpenSIPS must be compiled and configured, including the OpenSIPS AI Voice Connector. The topology is simple: OpenSIPS will act as the SIP gateway between FreeSWITCH and external AI voice services such as Vapi or ElevenLabs. Once everything is in place I want to: • Place SIP calls between two regular extensions registered on FreeSWITCH. • Route a call from a FreeSWITCH extension through OpenSIPS to the AI service and hear the synthetic voice reply. • Confirm two-way audio and proper SIP signalling in both cases. Feel free to rely o...

I want to move audio from a standard SIP trunk straight into LiveKit without using any third-party gateways. The bridge you build will sit between the SIP endpoint and...End-to-end demo: place a SIP call, hear the audio inside a LiveKit room via WebSocket, and close the call without leaks. Acceptance criteria • Latency under 200 ms round-trip on a local AWS region test. • No packet loss or distortion during a 30-minute call. • Automatic reconnection if either the SIP trunk or WebSocket endpoint briefly drops. If you’ve already worked with LiveKit, FreeSWITCH, Asterisk or similar tooling, that experience will help a lot. Let me know which approach you prefer for the SIP side—Go, C++, or Node are all acceptable—as long as the final bridge i...

...Uganda, and our goal is to: Configure our own SIP server Enable reseller accounts and billing Allow us to sell SIP services to clients/resellers Ensure security, scalability, and profitability This will be a carrier-grade setup suitable for handling high call volumes with proper billing, routing, and reporting. - Scope of Work Server Setup & Configuration Install & configure Kamailio / FreeSWITCH / Asterisk (preferred stack) Secure the server with firewall, SIP security, and anti-fraud measures Integrate with Airtel Uganda SIP trunk (upstream provider) Billing & Reseller Platform Deploy ASTPP / MagnusBilling / A2Billing (open source or licensed, depending on recommendation) Enable multi-level reseller management (resellers can create sub-resellers or clie...

...an AI voice agent built with FreeSWITCH. You'll install FreeSWITCH on one VM and build the AI voice agent server on another. Key Features: - AI voice agent for inbound and outbound calls (supporting Arabic and English) - Comprehensive call analytics: - Live call insights - Call recording and transcription - AI provider usage - Sentiment metrics - Local storage for all recordings, transcripts, and sentiment analysis on the server - Simple web admin dashboard with: - Knowledge base upload (files and URLs) - Access to all call recordings, transcripts, and sentiment analysis - Switch between AI keys (Google AI and VAPI) The project should be built using the MERN stack on an Ubuntu server. Ideal Skills and Experience: - Expertise in FreeSWITCH ...

We are looking for an experienced VPS Manager / DevOps professional to maintain, secure, and further develop our servers. We run on a Ubuntu 20.04 VPS environment hosting several applications. Our Server Environment Our current stack includes: Nextcloud ERPNext FreeSwitch & FusionPBX Matomo Mautic CyberPanel (for website management) N8N Paperless ActivePieces BaseRow Planned additions: Invoice Ninja A Large Language Model (LLM) We also operate a separate server running Mailcow for managing email domains. Immediate Priorities Install and configure Invoice Ninja for operational use. Resolve current issues and security problems with Nextcloud (including SSL). Set up the correct integration between Nextcloud and Paperless (via a Consume map). Fix DNS configurati...

...hang up instantly. Skills Required: • Strong experience in real-time audio processing (WebRTC, RTP, SIP audio streams, or equivalent). • Proficiency in speech and signal processing (e.g., VAD, MFCC, spectral analysis). • Machine Learning/Deep Learning for audio classification. • Experience with latency optimization in streaming systems. • Familiarity with telephony protocols (SIP, Asterisk, FreeSWITCH, etc.) is a strong plus. • Python/Node.js/Go/C++ (any language capable of handling low-latency audio). Deliverables: • Real-time streaming AMD system (replace current download method). • Early decision logic with configurable thresholds. • Integration of new detection features (synthetic voice, music, bit verification). • API or di...