Sipemplois

...combinés. L’objectif est de faciliter les traitements de qualité et d’analyse vocale que nous appliquons ensuite. Votre mission : – Mettre à jour proprement l’installe actuelle vers Asterisk 18. – Reprendre la configuration réseau à l’identique. – Activer et valider l’enregistrement en stéréo séparé pour tous les appels entrants et sortants. – Vérifier que l’ensemble des fonctions existantes (SIP, files d’attente, IVR, etc.) restent opérationnelles après la migration. – Me fournir un court mémo récapitulant les commandes utilisées, ainsi que la localisation des nouveaux fichiers ou modules é...

...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 de l’infrastructure VoIP. Mettre en place des outils de monitoring et de gestion des alertes. Documenter i...

...expertise in VoIP, Asterisk PBX, and Linux to configure our GoIP8 device. Our goal is to set up GoIP8 as a SIP trunk within FreePBX 14. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in configuring GoIP devices and SIP trunks. - Examples of previous projects where you successfully configured similar VoIP setups. - An outline of your approach to configuring GoIP8 as a SIP trunk within FreePBX 14. - Your proposed timeline for completing the configuration. - Your pricing structure for this configuration project. **Note:** We are looking for a reliable and efficient configuration of GoIP8 as a SIP trunk within our FreePBX system to ensure the smooth operation of ou...

une application Dialer qui peut faire un appel GSM ou SIP. Au moment de l'appel il vous demande si vous voulez passer par le GSM ou le SIP. Si vous choisissez le GSM c'est l'appel normal qui est lancé et si vous choisissez le SIP, l'appel passe par un PBX. Le SIP supporte le codec G729g. Les parties GSM et SIP sont toutes intégrées dans l'appli. Vous devez fournir le code source pour nous permettre de modifier certaines parties de l'application.

Bonjour, Nous recherchons un intégrateur spécialisé VOIP capable d'installer la pile complète de logiciel Goautodial 3.3 et y installer un trunk SIP pour permettre la mise en place de campagne d'appels sortantes (sondages, prise de rendez vous). Nous fournirons un serveur dédié (idéalement OVH) qu'il faudra paramétrer intégralement et installer avec les logiciels nécessaires au bon fonctionnement de la suite GOAUTODIAL 3.3, il faudra également prévoir une migration vers Goautodial 4.0 dans les prochains mois. Possibilité d'infogérance par vos soins du serveur dédié aux campagnes d'appels.

Bonjour à tous Voici mon projet et mon besoin. Je possède : 1 téléphone IP 1 numéro de téléphone SIP fourni par OVH 1 serveur dédié sur lequel est installé KAMAILIO. Mon téléphone IP était directement connecté à ma ligne SIP chez OVH (Je pense que ma ligne SIP chez OVH est installée sur un serveur ASTERISK chez OVH) Mais depuis peu, mon FAI bloque les enregistrements et appels SIP. Mon projet est de configurer un serveur intermédiaire KAMAILIO pour l'utiliser en tant que PROXY SIP pour enregistrer mon téléphone IP sur KAMAILIO sur un port différent du 5060 et que ce dernier relais les appels en...

Bonjour, je chercher un développeur pour coder une application sous android et IOS avec les spécifications suivantes : 1- Appelle sortant via un server(SIP/IAX). 2- Forward les appelles Voip vers GSM avec un limitation de duré max 10 minutes/24 heures( Call voip ---->GSM) 3-l'application doit être exécuter sur le smartphone en arrière plan. d'autres points doivent être discuter merci de bien vouloir me contacter.

La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à - installer un trunk SIP sur un serveur Asterisk - installer minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel - installer un panel de gestion (XIVO ou autre)

La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à terminer la configuration du logiciel VOIP Xivo sur un serveur VPS Linux : installation d'un trunk SIP installation de minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel

Cherche particulier ou centre d'appel qui utilise l'offre "call center" de ou ippi fr. Pour plus de Détail. veuillez me contacter

Bonjour, Je représente une PME spécialisée dans les équipements de sécurité (alarme, vidéosurveillance, contrôle d'accès, etc...) Nous devons remplacer notre bon vieux PBX PSTN par une solution IP. Nous avons opté pour le model Grandstream IP PBX UCM 6104 Nous allons utiliser six internes Grandstream GS-GXP2160 et des 3 lignes SIP. Nous cherchons une personne intéressée et capable de mettre en service notre système. Nous devons pouvoir utiliser ces services. (indicatif) - Répondeur avec horaire d'ouverture par créneau horaire - Intendant automatique avec 4 départements cibles - Mise en attente avec musique d'attente modifiable - Rép...

Nous avons un serveur asterisk configuré avec quelque sip trunk ovh. Nous avons des problemes dans la presentation des alias dans les appelles sortant. Nous avons besoins de quelqu'un qui connait bien se fournisseur Merci

Se connecter sur l'équipement sip goip, lire les statistics d'appels, et déduire ceux qui ont fait au moins x nombre d'appel échoué depuis la derniere verification. Relancer les appels sur les sims d'une durée de y minutes, avec le numero à appeler en input

Dans le but d'optimiser vos revenus 3D MULTIMÉDIA propose a ces free lancers de revendre ces solutions call center tels que ; interface , CRM , numéros surtaxés de tout destination ( compte sip ) , script de prospection , suivi technique , formation coaching aux centres d'appels intéressés par la réception d'appels et la prospection sur différents campagnes tels que LES JEUX CONCOURS , LA PUBLICITÉ , LA VOYANCE AUDIO-TEL .........etc Le but est de se connecter sur les forums de discutions et passé des annonces qui vous emmèneront a négocié en tant commercial avec vos clients centre d'appels de la solution création de trafic , le client disposera du pack gratuitement en ...

Bonjour, Je suis a la recherche d'une personne pour l’installation du CRM GOAUTODIAL qui est en open source bien sur : 1- L'installation se ferra sur un cloud de chez OVH France. 2- Le menu de GOAUTODIAL devra être en francais. 3- Un nom de domaine "" devra pointer dessus (L’accès a GOAUTODIA...pointer dessus (L’accès a GOAUTODIAL se ferra depuis se domaine par login et mot de passe sécurisé. 4- Conseils pour l'achat du serveur OVH. 5- Grande capacité de clients on appels simultané et compte admin et agent . 6- GOAUTODIAL devra être totalement personnalise "crutomisation" par mon logo et celui du client. 7- Payement par PAYPAL uniquement. 8- Assistance au paramétra...

salut Write an Android application appelle et partage fichier avec sip via wifi wifi+ appelle avec sip + partage fichier voir video qui ajouter je suis etudiant jai besoin bon prix

My FreePBX install has suddenly stopped accepti...not been touched for a while, yet every number on the system now rings out with no sign of life inside the PBX and no visible error messages. Outbound calls still work, trunks show as registered, but nothing is landing on the extensions. I need a seasoned FreePBX / Asterisk troubleshooter to jump in, trace what is blocking the inbound route, correct whatever trunks, inbound routes, firewall, or SIP settings are at fault, and get calls flowing again. Once it is working, I would like a brief note of what you changed so I can keep the system stable going forward. You will have full web GUI access on request; testing can be done live as this is a production box. I am in Australia, so I will need to be present to test, so day time (Bris...

... and basic auth are enough for the first release. Deliverable checklist 1. FreeSWITCH SBC build files and working configuration with Opus transcoding enabled. 2. MariaDB schema covering clients, PBXs, numbers, and routing priorities. 3. Web portal to add / update / delete routes and instantly reload FreeSWITCH. 4. README with deployment steps, firewall ports, and a quick test plan (two SIP softphones to emulate PBX and carriers). The solution must install on a fresh Debian VM and pass a live demonstration: a US call failing over to Carrier 2 on simulated 503, and an EU call doing the opposite....

...runs on Kamilio/Asterisk is suffering from three clear symptoms: call dropping, audio quality issues. The trouble appears only on a fraction of calls rather than every connection, so I need someone who can track down the underlying triggers—whether they sit in the signalling layer, jitter buffer, codec choice or network traversal logic—and then eliminate them. You’ll start by reviewing the current SIP/WebRTC stack, server logs, packet captures and any client-side metrics I provide. From there, pinpoint the root cause and apply code, configuration or infrastructure tweaks to stabilise audio flow. Deliverables • Detailed diagnosis outlining where packets or media streams fail • Fixed client or server code / configs ready for deployment • A...

...تحكم Dashboard شاملة يجب أن تكون احترافية وسهلة الاستخدام. العناصر الأساسية المطلوبة: أ. مؤشرات عامة للثروة إجمالي الأصول إجمالي التكلفة إجمالي القيمة الحالية إجمالي الأرباح العائد التراكمي ROI العائد السنوي أو المتوسط ب. كاش غير مستثمر خانة يمكن تعبئتها يدويًا إضافتها إلى إجمالي الثروة في العرض ج. مؤشرات شرعية إجمالي الزكاة المستحقة إجمالي التطهير المطلوب د. هدف مالي طويل المدى (SIP / Goal Tracker) المبلغ المستهدف (Target) يدخل يدويًا المدة الزمنية (سنوات) حساب نسبة تحقيق الهدف تلقائيًا مقارنة الأداء الحالي مع الهدف هـ. جدول العوائد السنوية عرض العوائد لكل سنة إمكانية تمدد الجدول مع السنوات المستقبلية و. توزيع فئات الأصول إجمالي الأصول المتداولة إجمالي الأصول غير المتداولة نسبة كل فئة من المحفظة ز. اقتراحات المستقل أقبل إضافة رسومات بيانية، مؤشرات، أو...

I need a skilled designer to create an insulated bottle lid. The lid should be made of plastic and include the following features: - Leak-proof design - Integrated handle -Tritan (copolyester) or PP for the lid body, silicone for seals/valves, 304/316 SS for metal straw components; BPA-free, phthalate-free, lead-free, food-contact compliant (FDA/LFGB). - sip and chug lid I have a design I like although want to imporve and make our own branding Ideal candidates should provide detailed project proposals and have experience in product design, particularly with plastic materials. Please include your past work and relevant experience in your application.

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 Telnyx decide, on every inbound call, whether to send traffic down the SIP trunk to my ElevenLabs voice agent or simply forward the call to a mobile phone. My WebApp is already writing a Boolean flag (True / False) to a Supabase table. Your task is to connect that flag to Telnyx Call Control so that: • when the value is True, Telnyx bridges the call through the existing SIP trunk to the ElevenLabs voice agent • when the value is False, the call is diverted to a specified mobile number You are free to implement the lookup however you think best—webhook, Function, or other Telnyx flow ele...

Tacodi’s video-conferencing MCU already exposes a C++ API; what’s missing is a clean, browser-based interface that will let me configure and supervise the system running on our NVIDIA Jetson Xavier boards (Ubuntu 18.04, OPAL/VoIP H.323 / SIP stack). The job is to design and implement that web application end-to-end. The UI should query the existing endpoints, display real-time status, and allow authenticated actions such as creating, editing or deleting resources exposed by the MCU (e.g. conferences, codecs, network parameters). All REST calls, response handling, and any required client-side validation need to be wired up and tested against the live device. I expect: • A responsive front-end built with a modern framework (React, Vue, Angular or similar) styled for...

I need Twilio hooked up to a Windows-based desktop softphone so I can handle both voice calls and SMS from the same workstation. The softphone itself is already installed; what’s missing is the full Twilio connection—SIP domain, credentials, messaging service, and all the event handling that goes with it. You’ll handle the Twilio Console setup, generate the proper tokens, and map everything into the client app so inbound and outbound voice calls register cleanly and SMS messages send and receive without delay. I’d like clear documentation on every step you take so the configuration can be replicated on additional machines later. Deliverables • Working Windows desktop softphone registered to Twilio, verified for inbound/outbound voice • SMS...

I am designing a single-story residence on a steep hillside site in a rural, mountainous area. Steel piers are used to accommodate the slope of the lot and support the building. I need a wind-loading assessment that I can use as the design basis for the design of the piers and connections. The residence is composed of 3 major components. 1. The residence is a timber frame and SIP design that will be engineered by the timber frame design firm. 2. A steel superstructure that supports the residence and transfers the loads to the foundation. The steel structure will be engineered by the steel design firm. 3. The helical pier foundation that will be engineered by the helical pier supplier. I am in need of an assessment of the loads placed on the steel superstructure that are the ...

Seeking a specialist to resolve technical problems in our AI phone agent system. Must understand Zadarma SIP, FastAPI backend, audio pipelines, and LLM-based response generation.

...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 so the React Native app can manage accounts, contacts, and recordings If you&rsq...

...tagging, and scheduling. • Admin dashboard: one location where I can track article performance, manage categories/tags, moderate comments, and upload infographic assets. • Multilingual-ready structure (Bahasa and future English version) although only Bahasa will launch initially. • SEO basics (schema, meta tags, XML sitemap) baked in from day one. Nice-to-have (not mandatory): a simple ROI or SIP calculator module that can be plugged in later without major refactor. Deliverables 1. Responsive website built on the agreed tech stack, deployed to my hosting. 2. CMS configured with sample content to show article and infographic workflows. 3. Admin dashboard with analytics widgets and role-based access. 4. Documentation covering set-up, theme customisati...

...(for omnichannel chat), WhatsApp Business API and Helpdesk ticketing systems. Frontend Updates: Update the React/ frontend to support the new LiveKit-based voice features and real-time dashboards. Tech Stack (Current & Required): Backend: Python 3.11+, FastAPI (Asyncio expertise required) Database: PostgreSQL, SQLAlchemy, Alembic Voice/Real-Time: LiveKit Cloud (Agents Framework), WebRTC, SIP/Twilio AI/LLM: OpenAI (GPT-4o), Deepgram (STT), ElevenLabs/Azure (TTS), LangChain/LlamaIndex Frontend: React, 15, TypeScript, Tailwind CSS, Zustand Infrastructure: Docker, Azure Blob Storage, Redis, Celery Who We Are Looking For: We are open to Freelancers and Part-Time Developers who specialize in one or more of the following areas: Voice AI Specialist: Must have proven experie...

...common questions, and either book a meeting with a human closer or close the sale outright—without sounding robotic. What I expect you to deliver: • A production-ready, cloud-deployable voice agent that can dial out, greet the prospect, walk through a scripted yet dynamic product demonstration, handle objections, and log the outcome. • Seamless integration with our existing phone stack and CRM (SIP/Twilio or comparable—happy to discuss what you prefer). • Real-time transcription, intent detection, and simple fallback rules so a human can take over if confidence drops. • A quick configuration panel (or JSON file) where my team can tweak the pitch, FAQs, and qualifying criteria. • Clear instructions for hand-off, including repo access ...

I need a fully-functional outbound voice agent that plugs straight into my VICIdial / GoAutoDial stack (Asterisk) and dials our CANADIAN leads through the SIP trunk we already use. Once connected, the bot must carry the entire conversation in a friendly, relaxed tone—answering common questions, keeping to our compliance wording, then moving naturally into a soft-close or handing the call over to a live agent when the prospect is ready. Key things I will be testing for: extremely low latency between speech-to-text, LLM response and text-to-speech; barge-in so the prospect can interrupt without awkward pauses; sentiment and intent tracking so the call flows feel human; and a clean handoff of every outcome, recording and transcript back into VICIdial and our CRM. Deliverables I...

...Fund application into working software that runs smoothly on iOS, Android, and the web. The core experience centers on three features: • Portfolio tracking that lets users see real-time valuations, gains / losses, and asset allocation. • Side-by-side fund comparison so investors can weigh expense ratios, historical returns, and risk metrics before buying. • Intuitive investment calculators—SIP growth, goal-based planning, and future value projections—to guide decisions. A secure email-and-password login flow is a must, so the build needs solid user-auth, encrypted data at rest and in transit, and basic account-management screens (sign-up, forgot-password, profile). I’ll supply UI wireframes, brand guidelines, and any required market-data ...

Objective : To develop AI Voice bot for our App User, WhatsApp bot user and SIP calling setup Who will use : User of Industrial Raw Material Buyer / Logistic / Trade Finance Purpose of Voice Bot : Introduction, Price quote and other support service Data source : Backend (Own Developed and Ready for use) Language : Hindi, Gujarati and English Data logged in CRM (3rd Party temporary later on own CRM is under development) API integration with WhatsApp for action triggered (Own web hook ready) API Integration with SIP Calling (Inbound / Outbound) Suggestion for Inbound / Outbound Cost Reduction / per minute Suggestion on Platform / Interface use for cost reduction like Bolna, Eleven Lab, Vapi, Exotel or any other for cost reduction purpose Expectation from AI Engineer : S...

The theme for my birthday is Scilla’s Sip & Slumber. I need an image created to be used on t-shirts. The colors are light blue, silver and white.

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 handset needs to be converted to SIP and fully registered. I do not yet have the SIP firmware package, so part of the job is walking me through the exact steps (and URLs, if publicly available) I need to follow to obtain it myself. Once I have the files, I want you to handle everything else remotely—preparing the TFTP environment, pushing the firmware to the phone, and creating the correct XML config so it boots straight into SIP and talks cleanly to FreePBX. When the flashing is done, the phone should: • Register to my existing FreePBX extension and pass audio both directions • Retain its s...

...are building an end-to-end voice-based AI agent that will sit between our in-house CRM and our SIP-trunk telephony infrastructure. The goal is to let the bot pick up every incoming support call, route it correctly, capture voicemail with live transcription, and also place outbound calls that qualify leads before handing them to our sales team. Here is what I need from you: • A complete, documented architecture—front-end, back-end, and middleware—showing how OpenAI or Google’s LLM will drive the dialog, while ElevenLabs handles both speech-to-text and text-to-speech. • Well-structured APIs that let our custom CRM push tickets, receive call logs, and trigger outbound dialing campaigns. • SIP trunk connectivity logic that supports call r...

...and high performance. This is a fast-track 10–15 day engagement. Only apply if you’re confident delivering polished, testable, and deployment-ready code within this timeline. What You’ll Build 1. Main Backend Service A modular backend with: Firebase + JWT authentication User profile module Rewards module KYC flow Integration with a third-party gold provider (similar to Augmont) Buy/sell Goal/SIP management Withdrawals A full relational database architecture using MySQL or Postgres A clean business logic layer with separation of concerns A double-entry ledger system API documentation (OpenAPI/Swagger) 2. SMS Processor Microservice A separate service responsible for: Pulling SMS data from the mobile app Parsing and categorizing messages Storing raw, processed...

I have an NEC SL2100 on-premise and need the voice SIP trunk programmed so calls flow reliably through my Mobily domestic trunk. The system is already on the latest firmware and running fine for internal extensions; what’s missing is the external connectivity. Here is what I’m after: • Load the Mobily credentials into the SL2100 and complete all registration and authentication fields • Map DID/DDI ranges so inbound calls ring the correct extensions and hunt groups • Verify outbound caller-ID, codec selection (G.711 μ-Law preferred), and any NAT or firewall tweaks in the SIP ALG • Run test calls with me—both directions—and leave the trunk stable for day-to-day use • A brief note or screenshot set of the key programm...

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 production environment, including: - Drachtio SIP server - Feature server - Databases - Monitoring - Call recording Only apply if you are already familiar with Jambonz, its architecture, and all required dependencies and services. --- ## Scope of Work ### Key Tasks - Install and configure **Jambonz Mini**, ensuring **Drachtio** is correctly set up for SIP call signaling. - Configure all relevant components for a small production setup (feature server, database, monitoring, call recording, etc.). - Perform end-to-end testing to confirm calls, signaling, media, and recording all function correctly. ...

I’m looking for someone who can quickly set up simple call forwarding from my existing Zadarma numbers to Vapi.ai. The goal is straightforward: every incoming call should be forwarded to a single destination I specify in —no IVR, no conditional logic, just a clean pass-through. My preferred approac...confirming that receives the calls without interruption. What I need from you: • Activate and configure the forwarding rule on Zadarma. • Test the flow end-to-end with me to verify audio quality and stability. • Provide a short hand-off note so I can edit the destination number later if needed. If you’ve connected Zadarma to other platforms before or have direct experience with SIP forwarding, this should be a fast task. Let’s get i...

I’m running Asterisk 20 on an Ubuntu 22.04 (Jammy Jellyfish) server and need it fully configured for secure WebRTC calling with a jsSIP-based browser client. Scope • Enable TLS for SIP traffic and WSS for WebSocket signalling, using self-signed certificates that you generate and install. • Update/adjust all relevant Asterisk, HTTP, and PJSIP settings so that a browser can register and place calls without certificate warnings once the cert is imported. • Deliver a working jsSIP sample page that demonstrates:  – two-way audio calls  – two-way video calls  – basic text messaging (MESSAGE method) • Provide clear, reproducible steps or a shell script that installs dependencies, copies configuration files (, , , etc...

...Technologies & Applications) course entirely face-to-face. The audience already holds CCNA-level knowledge; I want to take them the rest of the way in just one to two weeks, so deep, real-world expertise is essential. You will design and run the sessions onsite in central London, combining theory with hands-on labs that mirror production environments—CUCM, Unity, Expressway, Jabber, Webex Teams, QoS, SIP, and troubleshooting workflows must all feature prominently. I’ll provide the classroom and lab hardware; you bring the curriculum, step-by-step lab guides, slide deck, and any supplemental reference materials. By the end of the programme participants should feel confident tackling the CLCOR exam and applying the material on live systems. To gauge that, I’...

...to a working “out-of-the-box” state. That covers the core packages, basic SIP/Chan_PJSIP configuration, a starter dial-plan, and all the little touches—service auto-start, log rotation, and sensible security tweaks—so I can begin adding extensions immediately. If you’ve done fresh Asterisk deployments before, especially on Debian or CentOS, tell me where and how many; your hands-on experience will weigh far more than theory. Once you’re finished I want to be able to register two soft-phones, place a test call, and verify audio both ways—nothing fancy yet, just proof the platform is solid. Deliverables (all on my single VPS): • Install the latest stable Asterisk with required dependencies • Configure one sample SIP tr...

I have a ready-to-market drink that scores well on flavor up front, yet the aftertaste lingers longer than I’d like. I want that finish to feel smooth and subtle—no bitterness, no cloying sweetness—just a gentle fade that invites the next sip. At the same time, I need the overall expiration window pushed out without compromising the taste profile we already love. If you routinely work with beverage formulation, flavor modulation, preservatives, or alternative processing techniques (HPP, gentle pasteurization, nitrogen dosing, etc.), I’d like your guidance in choosing the most practical route for us. What I’ll need from you • A revised formula or process adjustment that delivers a sweeter and a noticeably smoother, subtler after-taste. &bull...

...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 desde el backend). Registro de usuarios SIP efímeros. Manejo de límites de canales por tenant. Backend en NestJS con arquitectura limpia/hexagonal: API REST y WebSocket/ para comunicación en tiempo real con el frontend. Integraci&...

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 endpoints for ASR/LLM/TTS will be provided. What You Need to Do Configure Genesys IVR and Media Server Set up and enable MRCP integration for ASR and TTS. Ensure proper SIP/RTP audio routing. Implement ASR Integration Send caller audio from Genesys IVR to the ASR endpoint. Receive the transcribed text and return it to the IVR flow. Implement LLM/NLP Integration Send the ASR text to the LLM via REST API (provided). Handle the AI response (intent or reply text). Implement TTS Integration Send the LLM text to the TTS API via MRCP/HTTP (depending on engine). Play the generated audio back to the caller. Build a Simple Multi...

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 browser. Think of it as a small PBX-style module but stripped down to the essentials: secure SIP registration, two-way audio over WebRTC, and a clean web phone interface. Core expectations • Django project that handles user auth plus SIP credential storage (encrypted) • Sign-in to the SIP registrar, keep the registration alive, and expose call events through WebSockets • Browser client (JavaScript, preferably using JsSIP or ) that dials out, answers, puts on hold, hangs up, and shows call status • Inbound call pop-up in the UI with basic accept/decline controls • O...

...Concurrent Calls, 6 Machines) Title: Looking for a VoIP Engineer to Set Up SIP/RTP Traffic Generator (500 Concurrent Calls, 6 Machines) Project Description: I am looking for an experienced VoIP engineer to build and configure a complete test environment capable of generating realistic VoIP traffic (SIP + RTP). The goal is to stress-test my own network and VoIP infrastructure by simulating approximately 500 concurrent calls, distributed across 6 separate machines. Scope of Work: • Install and configure a VoIP traffic generator (StarTrinity SIP Tester or SIPp). • Prepare SIP call scenarios for ~500 simultaneous calls. • Distribute the load across 6 physical/virtual machines. • Configure realistic VoIP traffic: • SIP signa...

...reporting. My current preference is to build the dialogue logic in Flowise because of its visual workflow designer and native webhook support, but if you can demonstrate a more efficient approach with a comparable platform I’m open to hearing it. What matters most is that the solution plugs cleanly into Asterisk, respects ASTPP billing logic, and keeps latency low. Please include: • Configuration of SIP endpoints, dial-plan changes, and any AGI/ARI scripts required for the bots to pick up calls. • Creation or import of intents and entities that let the bots perform the three core functions above. • A short knowledge-base so the FAQ component is testable on delivery. • Brief documentation outlining how to update flows and monitor usage. If you&...

...dashboard, 1 for admin and 2 for user. Core functionality is clear: during any phone conversation—whether dialed through Asterisk, VICIDIAL, SIP carrier—the System should capture each side’s speech, convert it into real time in both native languages, run an AI engine such as Deep Gram, Google Voice AI, etc., and speak the translated audio back to the listener with minimal latency. Two-way, real-time translation is non-negotiable; speech-to-text, text-to-speech, and airtight VOIP integration all need to feel seamless to the user. On the technical side I expect: • Secure, scalable web application to manage language packs, call logs, user accounts, and billing hooks • SIP registration or API-level hooks that let me plug and play the system into any...