Kamailio 5.5 WebRTC Proxy Setup
We’re looking for an experienced Kamailio 5.5 engineer to configure a production-ready WebRTC proxy on Linux. Requirements Kamailio 5.5.x with MySQL, optimized...
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AI Receptionist Call Router
I need a simple, self-hosted AI receptionist that picks up every incoming VoIP call, greets the caller as if it were our front desk, and guides the conversation...
Read moreFix FreePBX One-Way Audio
Incoming calls that arrive through our SIP trunk suffer classic “one-way audio”: the caller hears us, yet we get silence. Outbound calls and all internal extens...
Read moreESP32-S3 AFE I2S Integration
Project Description: I am looking for an experienced embedded audio/DSP developer to design and implement an audio processing firmware based on the ESP32-S3 Aud...
Read moreVOIP app base free calling to Phone with Development Android App & Admin, Backend
To develop a VoIP calling app to calling direct phone/mobile with custom caller ID, Admin portal- User management, wallet system to charge, custom caller ID, ca...
Read moreAdd Features to Linphone App
I need the open-source Linphone client modified please read https://docs.google.com/document/d/15GjPnxmkQudnO30yjq3IJigO14ptkA4FRHSWEkSoY9A/edit?usp=sharing Mai...
Read morePress-1 Autodialer - Custom Solution No Vicidial
I need a press-1 autodialer that places outbound survey calls through my existing PBX over SIP. The goal is simple: play a short recorded prompt, let the callee...
Read moreTriển Khai FreePBX Debian 12
Tôi đã cài sẵn Debian 12 trên máy chủ của mình và từng làm việc với FreePBX nên quen cách vận hành. Giờ tôi cần một chuyên gia hoàn tất việc cài đặt và cấu hình...
Read moreFix One-Way Audio Kamailio/Asterisk
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 acti...
Read moreFreePBX Extension Troubleshooting
My FreePBX install is up and running, yet several extensions refuse to register or place calls. I need someone who knows FreePBX and the underlying Asterisk CLI...
Read moreCustom ROM for Redmi 4A
create a custom ROM for Redmi 4a with 1. special audio routing 2.special restrictions
Read moreAI-Integrated Telephony System Engineer
We are looking for an Engineer to design, implement, and support Asterisk-based telephony systems integrated with AI conversational voice solutions. The role fo...
Read moreESP32-S3 SIP Firmware
I want custom firmware written for an ESP32-S3 SIP module ## Project Overview We are building a **hardware–software audio gateway** that converts a **live cellu...
Read moreBasic Twilio Voice Setup
We just signed on with Twilio, and are looking for some assistance, setting up a basic voice system for our small business.
Read moreLooking for an Experienced Team to Develop a FreeSWITCH-Based Inbound Call Tracking Software (Only Prior Experience Required)
Project Description We are looking for a highly experienced development team or company to build a complete Inbound Call Tracking Software based on FreeSWITCH....
Read moreAI-Based VoIP Call System Development
We are looking for an experienced backend / VoIP developer to build an AI-based call system with a fully integrated admin/management dashboard. This project is...
Read more3CX Autodialer & SIP Logic
I need a fully-automated outbound dialer built inside 3CX. The flow is simple: • The system grabs the next phone number in our CRM list and places the call thro...
Read moreFrance & Germany SIP Trunks Needed
I’m looking for a partner who can supply and configure reliable SIP trunks for both France and Germany on our self-hosted 3CX system. The trunks must support fu...
Read moreFix 3CX-Twilio Outbound Calls
My 3CX PBX is correctly receiving incoming calls through our Twilio SIP trunk, yet every outbound attempt fails. due to outbound rules. i have setup this but i...
Read moreUrgent Cisco 8851 Paging Fix
I’m looking for immediate remote assistance—today—to straighten out two issues on our Cisco 8851 desk phones. First, paging works everywhere except a handful of...
Read moreFreeSWITCH VoIP Programmer Needed
I'm looking for an experienced FreeSWITCH developer to create a robust VoIP solution. Key Features: - Call Routing - SIP Trunking - Call Recording Ideal Skills:...
Read moreInstalación de cloud PBX VoIP con Issabel
Necesito que me instalen desde cero una centralita virtual PBX VoIP basada en Issabel con el objetivo de gestionar el soporte al cliente. Yo facilitaré los dato...
Read moreVoIP Audio Reliability Fix
Our production VoIP app runs on Kamilio/Asterisk is suffering from three clear symptoms: call dropping, audio quality issues. The trouble appears only on a frac...
Read moreTelnyx SIP Routing -Toggle Integration
I already have a Telnyx account, a working SIP trunk, and all of the Supabase and ElevenLabs credentials on hand. What I need now is the control logic that lets...
Read moreMulti-Channel CPaaS Platform Development
I need a complete Communication-Platform-as-a-Service that lets businesses and resellers deliver messages and calls over SMS, Voice, WhatsApp, Email (Gmail-comp...
Read moreReact Native VoIP Server Build
I need a solid back-end that lets a future React Native client place and receive calls with the reliability of a carrier-grade PBX. The server must handle: • In...
Read moreVoIP/OpenSIPS/Cisco CUCM Engineer for FCM Push Notifications
Summary We’re looking for an experienced VoIP/OpenSIPS/Cisco CUCM engineer to design and implement FCM-based push notifications for our mobile and web softphone...
Read moreFlash Cisco 9951 & FreePBX
I have a Cisco CP-9951 that is still running the factory SCCP load. My FreePBX instance is already up, stable, and handling other endpoints, but this particular...
Read moreAsterisk 18 Upgrade & Stereo record
Je fais tourner une plateforme de téléphonie VoIP pour centre d’appels basée sur Asterisk 11.5. Je souhaite la passer sous Asterisk 18 tout en conservant l’ense...
Read moreJambonz Mini Installation and Configuration
I’m looking for an experienced engineer to install and configure **Jambonz Mini** on our own server (not AWS). This will be an all-in-one setup for a small prod...
Read moreASAP USA DID & SIP Trunk
I need a ready-to-use U.S. DID number and a SIP trunk activated today. As soon as you deliver the credentials, I’ll register them on Zoiper and the native diale...
Read moreLondon Cisco CLCOR Trainer Needed
I need a London-based professional who can deliver an intensive Cisco CLCOR (Implementing and Operating Cisco Collaboration Core Technologies & Applications) co...
Read moreAsterisk PBX Initial Setup Assistance
I have a clean server ready and need Asterisk installed and brought to a working “out-of-the-box” state. That covers the core packages, basic SIP/Chan_PJSIP con...
Read moreDesarrollo VoIP+Backend para FreeSWITCH Multi-Cliente
Busco desarrollador VoIP + backend para montar una arquitectura con FreeSWITCH multi-cliente integrada a mi backend en NestJS y frontend ReactJS. Objetivo gener...
Read moreGenesys IVR Integration with ASR, LLM, TTS
Overview I need a freelancer to build and implement a working integration between Genesys (on-prem) IVR and external ASR, LLM, and TTS services. All API endpoin...
Read moreDjango-Based SIP Web Phone
I’m putting together a lightweight voice platform in Python / Django that can register my existing SIP account and let me place and receive calls right from a b...
Read moreConfigure Kamailio SBC With RTPengine -- 2 - 13/11/2025 14:56 EST
We are looking for an experienced VoIP engineer to configure Kamailio as an SBC together with RTPengine. The goal is to fully rewrite all SIP and RTP traffic so...
Read moreSIP Telephony & Voice AI Integration
Implement integration of SIP telephony with voice AI agent for handling incoming/outgoing calls through Asterisk or alternative Initial Data Telecommunication o...
Read moreSIP Telephony & Voice AI Integration
Implement integration of SIP telephony with voice AI agent for handling incoming/outgoing calls through Asterisk or alternative Initial Data Telecommunication o...
Read moreAI-Powered Live Call Translator
I want to bring an AI-driven live-call translation system to life, delivered as both a mobile app and a companion web interface. The mobile side must run smooth...
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