Freeswitch jobs
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...
...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...
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...
*** 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...
