Asterisk i punët

Pershendetje Ardit, keni eksperience me Suite crm apo Vtiger crm? kam nevoje pernje konfigurim ne lidhje me crm dhe asterisk

...Filtrado antes de 200 OK Reglas basadas en: Diversion History-Info 181 Call Is Being Forwarded 302 Moved Temporarily patrones en Contact patrones en Request-URI headers específicos del carrier Logs de llamadas bloqueadas Motivo del bloqueo Estadísticas básicas Lista blanca / lista negra de carriers o prefijos Configuración editable por archivo o base de datos simple Fuera de alcance No AMD No Asterisk No FreeSWITCH No RTPengine No análisis de audio No grabación No transcodificación Entregables esperados Instalación de Kamailio en servidor ubuntu Archivo documentado Reglas anti-buzón configurables Pruebas con SIPp o tráfico real Documentación básica de operación Recomendación de serv...

I recently added a new SIP trunk to my setup—FreePBX 15 sitting behind a Kamailio SBC—but the trunk will not register and no outbound calls complete. I need someone who understands FreePBX 15, Asterisk SIP settings, and Kamailio routing to: • Diagnose why the trunk fails to register with the provider • Configure the correct authentication, codecs, and SIP headers so outbound calls connect cleanly • Provide a brief step-by-step summary of the changes you make so I can replicate them in the future You will have to work via Anydesk, you won't be able to directly connect to our server.

I want to deploy an AI-powered voice agent, built with ElevenLabs’ speech technology, that can automatically dial through our real-estate lead lists, hold a human-sounding conversation, qualify prospects and, when appropriate, lock in a phone consultation for one of our agents. Here’s the flow I’m envisioning: the bot connects via our dialer, introduces itself naturally, asks the key qualifying questions we already use, handles common objections, and—once the lead matches our criteria—books a follow-up call on the team’s calendar. All relevant call notes, recordings and contact details should feed straight into our CRM so no information is lost. You’re free to choose the back-end stack (Twilio, Asterisk, custom VoIP, etc.) provide...

Project Title: ​Expert needed to Integrate 4-Port GSM Gateway with Asterisk/FreePBX for Automated IVR & SMS API ​Project Description: ​"I am looking for a VoIP/Telephony expert to set up a complete automated IVR calling system for my fintech app. ​Hardware & Infrastructure: ​I have a 4-Port GSM Gateway (Dinstar/GoIP or equivalent). ​I have my own Cloud Server (VPS). ​Scope of Work: ​Asterisk/FreePBX Setup: Install and configure Asterisk or FreePBX on my cloud server. ​SIP Trunking: Securely connect my 4-Port GSM Gateway (located at my office) to the Cloud Server via SIP Trunk. ​Automated Calling API: Create an API that my mobile app can trigger to make automated outbound calls (Voice IVR). ​SMS Integration: Enable the ability to send O...

AI Call Center Developer – Twilio, Dialogflow, CRM Integration (Outbound Loans & Credit Cards) Responsibilities: Set up VoIP/telephony layer (Twilio or Asterisk). Implement Speech-to-Text (STT) for real-time transcription. Integrate NLP engine (Dialogflow or Rasa) for intent detection. Add Text-to-Speech (TTS) for natural voice responses. Connect system to Zoho/HubSpot CRM for lead logging and callbacks. Build branching conversation flows for multiple campaigns: Personal loans. Business loans. Credit card offers. Provide basic analytics dashboard for call outcomes. Requirements: Proven experience with Twilio, Asterisk, or similar VoIP systems. Strong skills in Python/Node.js for API integration. Familiarity with Dialogflow, Rasa, or NLP frameworks. Experience w...

...time, creating a severe bottleneck for my agents. I need the campaign and server-level dialer settings re-engineered so that the dialer can launch up to 20 simultaneous calls, keep the hopper continuously stocked, and eliminate any agent idle time. This assignment is a configuration task, not a hardware upgrade—the goal is to increase the dialing ratio and fine-tune any related rules that could limit call throughput. Key areas to address • Adjust the campaign’s dial level / adaptive or ratio settings to hit 20 concurrent dials • Verify hopper and list mix behaviour so leads feed without interruption • Confirm SIP and carrier limits are respected while preventing call queues or drop spikes • Document every parameter you change so I ca...

I need a dial plan that post messages to a website Have a IVR to read and update the website and database Log information and route based on situations Call back based on timeing Send calls based on timeing

I need a dial plan that post messages to a website Have a IVR to read and update the website and database Log information and route based on situations Call back based on timeing Send calls based on timeing

I already have a Telegram bot that should dial leads, play an IVR prompt, collect a “Press 1”, then play a closing message and hang up. Right now the Telegram side works, but the voice side does not—the bot simply isn’t making calls. Here is the exact flow I need brought to life (or repaired if you prefer to reuse my existing code): • I drop a .txt file of U.S. mobile numbers into the chat with the bot. • I send two MP3s: one to play on answer, the other after the caller presses 1. • The bot dials every number simultaneously (or as close as carrier limits allow), plays the first MP3, waits for DTMF 1, then plays the second MP3 and ends the call. • When all numbers have been processed the bot returns a Telegra...

My Vicidial instance on a Linode VPS has suddenly dropped off-line. The hostname no longer resolves and I cannot reach the machine even by its IP, so I’m convinced this is a network-level problem rather than a recent configuration change—nothing has been touched for weeks. I need someone who is comfortable diving straight into a production Linode, checking DNS records, firewall rules, networking interfaces, and Vicidial/asterisk services to get everything visible on the internet again. Root SSH will be provided the moment you’re ready to start; the goal is minimal downtime and a clear explanation of what went wrong so it doesn’t repeat. Deliverable: • The server’s web and SIP services must be reachable externally at (and by direct...

...mid-article Clean print stylesheet Section landing pages — three template variants for the Review's internal sections (Research, Essays, Signals). Each is a typographic listing page with a section title, a one-paragraph section description, and a reverse-chronological list of pieces. No hero image. See sample HTML for visual treatment. Issue page — template for displaying a curated issue (e.g., "Issue I"). Includes an editorial introduction from the editor, issue cover illustration, and a curated list of the pieces in the issue, grouped by section. Author / contributor page — template for individual contributors showing biographical information, photo, affiliation, and a list of their published pieces. Review About page — static page templ...

Necesito montar un servicio de atención al cliente que responda, ante todo, a consultas generales y mantenga la misma eficiencia en todos los puntos de contacto. Alcance del trabajo 1. Configurar Chatwoot como centro de chat en línea y conectar los perfiles de redes sociales para que los mensaje...primera respuesta, tasa de resolución y satisfacción). Entregables esperados • Plataforma Chatwoot operativa con los cuatro canales activos. • Integración funcional con el sistema de facturación, CRM y marketing. • Documentación paso a paso (credenciales, flujos y manual rápido de uso). • Capacitación breve por videollamada para mi equipo (máx. 1 h). Valoro experiencia previa en Chatwoot, Aster...

...Indian telecom 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 an...

## 1. Project Scope The service provider will design, install, and configure a production-ready VoIP Call Center system using Asterisk on a Linux-based environment. The scope includes: * Server Environment: Installation and hardening of the latest stable Asterisk version on a cloud-based or local Linux server (Ubuntu/Debian). * Trunking & Connectivity: Integration of [X] number of SIP Trunks for external inbound and outbound calling. * Extension Management: Creation and configuration of [X] agent extensions supporting PJSIP/SIP protocols. * Call Flow Logic (Dialplan): * Development of a multi-level IVR (Auto-Attendant) menu. * Implementation of time-based routing (Business Hours vs. After Hours). * Queue Management (ACD): Setup of call queues with specific distributio...

Click-to-call, Click-to-hang-up using Asterisk Manager api (AMI) In click-to-call (in outbound) api first call hit the agents number (not using softphone or hardphone) after receiving call in agents mobile then customer number will dial. Also handling the Inbound PBX – Call is hitting in a virtual number after that call will forward in agent mobile number. WebRTC – HTTP/WebSocketSecure (WSS), PJSIP with DTLS encryption. Sngrep – to monitor the sip calls. Webhooks – Send real time call data Using Dial plan, AGI, or AMI. Web Sockets – Web RTC based browser phone. STIR/SHAKEN – Identify the source of a call. Current Project – IVR Integration with Chat bot System using asterisk 13 and above version...

...contract, you must deliver: Working call centre system with IVR and routing Integrated WhatsApp, SMS, and chat support Case management system with user tracking and referrals AI triage, sentiment detection, and risk scoring Real-time dashboards with key metrics Secure deployment with backups and monitoring Full documentation and staff training Technical Scope (Condensed) You will handle: Telephony: Asterisk or equivalent Backend: FastAPI / Django or similar Messaging: Chatwoot or equivalent AI: NLP classification, sentiment, risk scoring Data: PostgreSQL + dashboard tool (Metabase or similar) Deployment: Docker-based, secure cloud setup Budget Guidance We are open to serious, well-justified proposals. Expectation: $3,000 – $7,000 fixed price Proposals far below this range w...

...contract, you must deliver: Working call centre system with IVR and routing Integrated WhatsApp, SMS, and chat support Case management system with user tracking and referrals AI triage, sentiment detection, and risk scoring Real-time dashboards with key metrics Secure deployment with backups and monitoring Full documentation and staff training Technical Scope (Condensed) You will handle: Telephony: Asterisk or equivalent Backend: FastAPI / Django or similar Messaging: Chatwoot or equivalent AI: NLP classification, sentiment, risk scoring Data: PostgreSQL + dashboard tool (Metabase or similar) Deployment: Docker-based, secure cloud setup Budget Guidance We are open to serious, well-justified proposals. Expectation: $3,000 – $7,000 fixed price Proposals far below this range w...

I’m launching “Redy 24 jam,” a round-the-clock customer support line that will operate exclusively by phone. I need a freelancer to set up a reliable system where callers reach automated responses that handle common questions before, if necessary, forwarding to a live agent. Scope • Configure an IVR or similar solution that greets callers and delivers clear, automated answers. • Draft concise voice prompts in natural-sounding Indonesian and English. • Integrate the phone system with my existing carrier or a cloud platform (e.g., Twilio, Asterisk, or 3CX). • Provide a simple dashboard or method for me to update messages without coding. • Supply basic training documentation so my team can monitor call logs and tweak scripts. ...

...conversation is automatically recorded and stored with searchable time/date tags. • Outbound 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. ...

...gateway (Ozeki → Kannel migration) Setting up and troubleshooting SMPP connections Creating SMPP users and integrations Managing Asterisk SIP system (users, routing, configs) Troubleshooting delivery and connectivity issues This is a remote contract role (monthly retainer) with flexible hours, but requires availability for incident response. Requirements: Strong experience with SMPP Experience with SMS gateways (Kannel, Ozeki, or similar) Experience with Asterisk Strong Linux + networking knowledge IMPORTANT – Application Requirements To be considered, please include: Your experience with SMPP (real examples) SMS gateway(s) you have worked with Your Asterisk experience (what you configured) Your preferred monthly retainer fee (in USD) Applications w...

Criação de um sip proxy usando kamailio e rtpengine em um servidor debian. Objetivo: Ser o único ponto com IP Público para diversos asterisk (freepbx 17). Estes terão apenas ips privados Os freepbx já existem e já funcionam, porém cada um com IP público, eles apenas devem passar a usar apenas ip privado. Um arquivo (dispatcher) deve ter os hosts separados com dominio, ip e porta (será diferente de 5060) Cada PBX possui seu próprio tronco com seu próprio usuário para uma operadora IP (IDT / net2phone) O sip proxy deve entender ramal@dominio e encaminhar o tráfego para o freepbx correto de acordo com o arquivo dispatcher. e também deve saber diferenciar o tráfego que vir&aa...

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 service integration so my agents can place and receive calls directly through the platform without juggling extra software. Here’s what I expect: • A stable dialer application that can import contact lists, initiate outbound calls, and display real-time call status. • Direct integration 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 af...

I need a full-stack voice AI that can automatically call a list of prospects I upload, speak with them in a natural-sounding voice, and switch fluidly between Hindi, English, and everyday Hinglish. The goal is to make the conversation feel as human as possible—no robotic intonation—while capturing key data from each call. The application must run smoothly on mobile devices, in a browser, and as a desktop client so my sales team can launch and monitor campaigns from anywhere. While the call is in progress the system should detect regional accents accurately, filter typical background noise found in Indian office or field environments, and generate real-time transcriptions that I can download or push to my CRM. I do not need highly philosophical chat...

Asterisk tabanlı sistemimizi uçtan uca yönetebileceğim, Issabel’e benzer ancak güvenliği çok daha ön planda tutan yeni bir web paneli yaptırmak istiyorum. Kod tabanı temiz, modüler ve bakımı kolay olmalı; ek paketlere bağımlılığı minimumda tutmanızı tercih ederim. Panel tamamen Türkçe arayüzlü olacak, modern bir UI kütüphanesi (Vue, React veya benzeri) ile geliştirilebilir; arka planda ise Asterisk AMI/ARI entegrasyonu sağlam, rol-bazlı erişim kontrollü bir mimari isterim. Teslimatta mutlaka bulunması gereken ana modüller • Çağrı yönetimi ve kuyruk yönetimi • Kullanıcı / dahili yönetimi • IVR tasarımı ve yönetimi • Raporlama ve istatistik p...

I have a Telegram bot written in Python and an Asterisk server already exposing the ARI interface. What I now need is a clean bridge between the two so that a simple text command sent to the bot can spin up an outbound call, play an IVR menu, record the conversation, and finally report the result back to the same chat. Here is the flow I have in mind: • A user types a command such as /call <number> in Telegram. • Your code uses ARI (ari-py or any other reliable ARI client) to originate the call. • On answer, the callee is taken through a short IVR tree that I will supply as audio prompts; the bot just needs to play them and collect DTMF. • The entire interaction is recorded and the recording URL or file is posted back ...

My current Python application already takes care of business logic, but I now need it to talk directly with Asterisk through the ARI websocket so it can steer the live call flow itself. The immediate requirements are simple and focused: I want the script to: • Answer an outbound or bridged call as soon as it arrives in the ARI app • Play a short audio prompt (a .wav I will supply) • Hang up cleanly when the dialogue is finished or when I signal it in code Event-wise, the code must listen for any outgoing-call events I raise inside Asterisk, acknowledge them, and then run the flow above. Please keep the implementation in plain Python 3.10 and feel free to rely on either ari-py, Starlette websockets, or a lightweight alte...

Telefonía y VoIP • Experiencia en implementación de PBX / IP-PBX • Manejo de protocolos: o SIP o RTP • Experiencia con plataformas como: o Twilio o Asterisk o 3CX y otras • Configuración de: o Troncales SIP o IVR o Grabación de llamadas ________________________________________ Call Center / Contact Center • Implementación y soporte de: o Colas de llamadas o Routing (ACD) • Experiencia con plataformas como: o Five9 o Genesys • Monitoreo de agentes y KPIs ________________________________________ Redes y sistemas • Conocimiento sólido en: o TCP/IP o VLANs o VPN o QoS • Configuración de: o Routers / switches o Firewalls ________________________________________ Sist...

...dialogues with context retention (e.g., customer asks for price, then changes origin city). Negotiation & Closing Logic: Auto-counter offer when customer says "Price is high" (respect max discount % from dynamic config). Urgency triggers: if customer says "Comparing options", push limited-time discount. Objection handling ("I'll think", "Too costly", "Need to ask partner"). Final closing: "Shall I send a payment link to lock this price?" Cross-sell Engine: Based on primary enquiry, proactively offer relevant services (e.g., Transport enquiry → Transit Insurance). Ask "Do you also need insurance for your goods?" and capture response. Dynamic Pricing Integration: Apply multipliers for urgency...

...Goal: Ensure all website content is compliant and aligned with regulatory and platform guidelines Tasks: ● Review all website copy for: ○ Medical or disease-related claims ○ Non-compliant or overly aggressive language ● Ensure proper use of: ○ Structure/function language (e.g., “supports,” “promotes”) ○ Avoidance of prohibited claims (e.g., “treats,” “cures,” “prevents”) ● Verify: ○ Asterisk (*) usage where applicable ○ FDA disclaimer visibility and placement ● Review: ○ Terms & Conditions ○ Privacy Policy ○ Shipping & Return Policies ○ FAQ content ● Ensure consistency across: ○ Website messaging ○ Product claims ○ Brand tone ● Flag risks and recommend revisions Deliverables ● Full audit re...

Project Overview: We are looking for a developer with strong hands-on experience in FreePBX, Asterisk dialplan development, and AGI scripting to build out a phone dating and live pickup line platform. This is not a basic PBX setup job. We need someone who understands how to create a fully interactive call-flow application inside Asterisk/FreePBX, with custom logic, menu routing, caller handling, agent availability, and database-driven call processing. The platform will function similarly to classic phone dating systems, where callers can interact with prompts, browse options, connect to live users or operators, leave messages, and move through a structured call journey. We want a developer who can translate a process flow into a working telephony system and help refine the ...

I have Asterisk running on a Raspberry Pi that talks to a CASQ gateway, with a small Python service on the same Pi forwarding AMI events to our cloud-based campaign manager. Answered calls report fine but when a call is simply left ringing the “no-answer” status never reaches the central server, so campaign statistics remain wrong. The break seems to be somewhere between Asterisk’s event stream and the Python handler on the Pi, yet I’m open to any finding you uncover. You’ll join me through AnyDesk (or your preferred remote desktop tool), dive into the dial-plan, AMI permissions and Python callbacks, reproduce the issue with a couple of test calls, then patch the logic so missed calls are reported instantly. I can provide full Asteris...

I need a mobile PBX application that lets my team place and receive internal calls entirely over our local Wi-Fi network. The app must run smoothly on both iOS and Android and register 51-200 extensions without relying on cellular voice or external trunks. Core functions I must see on day one: • Call forwarding, including unconditional, busy, and no-answer rules • Visual voicemail that stores messages locally and syncs with the PBX I have no immediate need for call recording, yet I prefer a codebase that leaves room to add it later. Whether you build the PBX side (Asterisk, FreePBX, 3CX, etc.) from scratch or integrate with an existing server, the result has to deliver reliable SIP registration, push notifications for incoming calls, TLS/SRT...

FreePBX (current version) is already up, SIP trunks are active, and everything is stable. I now need an Asterisk / FreePBX specialist to finish the last mile: • Upgrade the firmware on my Sangoma DB20N base station. • Register its wireless handsets and ensure they stay connected. • Configure each handset for reliable inbound and outbound calling and smooth call parking. Once everything is in place I should be able to pick up a handset, make and receive calls, and park or retrieve them without glitches. I will give you SSH / GUI access to the PBX and the DB20N’s web interface so you can work directly. Please outline how long the upgrade and configuration will take, any precautions you want me to handle on-site (reboots, backups, etc.)...

I want to reach my customers by phone without dialing each number myself, so I’m looking for a small-scale automated calling system that will run on just one phone line. Your task is to recommend the most cost-effective hardware or VoIP service, configure the software (Asterisk, FreePBX, Twilio, or a comparable solution—whichever fits best), and walk me through basic operation so I can upload a list of numbers, schedule calls, and play a short pre-recorded message or transfer to me when someone picks up. Key points • Only one active line is required, so the setup should be light and simple—no full call-center stack. • I need the system installed, tested, and documented so I can manage future campaigns myself. • Plea...

Need 157 short audio text cleaned up. Have audio files that AI built that have issues. They need to be cleaned up, timing spaced out, and anything weird fixed. Command Number 1: Be fruitful ...physical offspring, it encompasses raising up a righteous generation devoted to Yahuah, bearing spiritual fruit, and producing good works in every area of life. Plant a family garden, invest in your community, and multiply blessings wherever the Father has placed you. Family Reflection Questions: 1. How am I actively "being fruitful" in my home, work, and community? 2. Am I investing in things that will produce lasting fruit for Yahuah's Kingdom? When the AI says things like Asterisk need that edited out. Need each audio to sound perfected and then when ...

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 before the call proceeds. The only document we will accept is a Driver's License, so the solution should extract and validate all standard licence fields—name, address, date of birth, licence number and expiry—while logging a full audit trail for compliance. I am flexible on the underlying VOIP stack, provided the final workflow runs smoothly on my current hosted instance and is maintainable in-house after hand-over. Deliverables •...

ONLY BUILD IF YOU HAVE STRONG EXPERIENCE IN THESE AREAS SIP, NAT Traversal, Asterisk/FS, React and strong backend servers scaling. I’m building a browser-based VoIP platform dedicated to business communication and I need an experienced React developer to take it from architecture to live deployment. The entire feature list—covering everything from secure voice and video calling to messaging and call-related utilities—is spelled out in the requirements document I’ve attached, so you’ll have clear, granular guidance from day one. SIP, NAT Traversal, Asterisk/FS Tech expectations You’ll craft a responsive single-page app in React (TypeScript preferred) that connects to a SIP/WebRTC back-end. I’m open to your preferred server...

...total client payout, clicking on any line leads to detailed CDR of that result, this should be CSV downloadable, -Stats: Reports: Per date range, Per carrier total amounts. -Stats: Live calls, also with search options -Access lists: you know about this :) i will provide you a sample, and should be downloadable -Numbers which are not allocated to any client, they should play random IVRs from an IVR groups which we create and upload on the platform -Ability to download by the end of the month invoices for each carrier, with the form which i will send you. -Access list download: checking latest calls by date range and export file with range name, call from (country), call from (number calld minus 3-4 digits), last call date and time Client Interface: -CDR, amounts, re...

I already have a Linux server online and reachable; what I need now is a clean installation and full configuration of Asterisk, FreePBX, and A2Billing so I can run a reliable business phone system. Your scope covers: • Installing the latest stable releases of Asterisk, FreePBX, and A2Billing on my existing server • Bringing the stack to a “ready-to-use” state—SIP extensions, VoIP trunks, IVR, call recording, voicemail, and billing profiles must all function out of the box • Hardening the server (firewall rules, Fail2Ban, strong passwords/keys, SSL where applicable) • Running test calls to confirm inbound and outbound traffic, rate decks, CDR logging, and balance deductions in A2Billing • Handing over a ...

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

I need an IVR built and configured so callers reach the right help fast. Its sole purpose is customer support, and it must offer three options in every relevant branch: an automated response for common questions, the ability to transfer to a live agent, and—when queues are long—a call-back request feature. You’ll design the call flow, record or synthesize the prompts, connect it to our existing SIP/VoIP trunk, and make sure the routing to our support team works flawlessly. I’m open on the exact number of menu levels; clarity and speed for the caller are what matter most, so feel free to recommend the simplest structure that still covers our FAQs and escalation paths. Final hand-off should include: • Diagram of the call flow with all customer touch-poin...

I need a complete IVR built and configured to streamline our customer-support line. The system must greet callers, then automatically route each call to one of five departments through a simple, voice-prompt menu. In addition to this core routing, I also want the IVR to trigger bulk messages (SMS or WhatsApp blasts) and hook directly into Meta Business Suite so our phone flows, social inbox, and chat automations all stay in sync. You are free to choose the underlying platform—Asterisk, FreePBX, Twilio, or a comparable cloud service—so long as it supports reliable call routing, bulk-messaging APIs, and Meta’s official integrations. If a ready-made template speeds things up, that is fine; the voice prompts, language, and menu logic must still be custom...

Troubleshoot asterisk trunk TLS/SRTP

Project Title: Full Issabel 5 PBX Deployment: Installation, Trunks & Extensions Project Overview I am seeking a VoIP specialist to perform a complete installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architecture, including SIP trunks for external connectivity and internal extensions for users. Detailed Scope of Work 1. Server Installation & Hardening Perform a clean installation of Issabel 5 on AlmaLinux 8 using the official net-install script. Configure Fail2ban and firewall rules to block unauthorized SIP and SSH attempts. Set secure passwords for the Linux root, MariaDB, and Issabel web admin. 2. SIP Trunk Configuration Connect the PBX to my chosen VoIP provider using a SIP...

Necesito conectar nuestro PMS Cloud con la centralita Asterisk para cubrir funcionalidades de gestión de hotel. El objetivo es que el personal pueda manejar desde el PMS: • Registro de huéspedes: que cada check-in o check-out actualice automáticamente el estado de la extensión telefónica asignada. • Asignación de habitaciones: que al cambiar una habitación en el PMS se reprograme la extensión correspondiente en Asterisk sin pasos manuales. • Facturación y pagos: que las llamadas salientes e internas se registren en la cuenta del huésped y se reflejen en la factura final. Ya contamos con un software específico de gestión hotelera en la nube; requiero que el integrador traba...

I already have a Contabo server standing by and simply need a clean, production-ready ViciDial stack on it. The job covers: • Installing the latest stable ViciDial (with Asterisk and its dependencies) from scratch. • Optimising the underlying Linux distro you feel is most reliable for call-centre workloads. • Hardening the box with a well-tested firewall configuration—CSF, UFW, iptables, or a similar solution is fine—as long as only the ports Vicidial, SSH and web administration actually require remain open. • Verifying that the web interface, database, and telephony services all start automatically after a reboot and that calls can be placed through a demo campaign. • Supplying a concise hand-off sheet: all commands run, credenti...

...call to the right person or department purely on the basis of what the caller actually says. The key outcomes I need are: • Advanced conversational AI – the system must go well beyond simple keyword spotting and understand context, intent, and follow-up questions so the interaction feels human. • Dynamic query-based routing – once the caller’s need is clear, the AI should transfer the call to the appropriate extension or external number automatically. • Clean hand-off – when the call is routed, the receiving party must get a short, accurate summary of the caller’s request so they can pick up seamlessly. I’m happy to integrate with existing VoIP platforms (Twilio, Asterisk, FreePBX, 3CX or similar) if that speeds ...

I have a WebRTC soft-phone built with JsSIP that needs to register to an Asterisk 18 server over WSS. SIP credentials are confirmed correct, yet the browser console shows an authentication failure. The signalling path is protected with TLS certificates, so the problem is somewhere in the certificate handling or the way Asterisk presents the challenge. Your job is to trace and eliminate the registration failure, then hand back a clean configuration and proof that the client can successfully register and place a test call. Environment details you will touch: – Asterisk 18 (pjsip stack enabled) – JsSIP running in a standard browser (wss://) – TLS with server and client certificates already issued Acceptance criteria: • JsSIP complete...

...configured CRM integration required with Zoho CRM The freelancer’s role is to properly configure campaign logic, IVR flow, DTMF capture, reporting, and CRM integration. 2️⃣ Existing Infrastructure Dialer: VICIdial (ViciBox 12, Asterisk 16.x) TTS Engine: Amazon Polly (Aditi voice configured) Audio: Pre-recorded WAV files ready GSM Gateway: Dinstar Server: On-prem Linux (Public IP available) CRM: Zoho CRM SSL: Not configured yet (freelancer may configure if required) 3️⃣ Scope of Work A. Audio & TTS Handling Freelancer must: Verify WAV format compatibility (8kHz, mono PCM for Asterisk) Upload & map audio properly in VICIdial Use Amazon Polly only where dynamic text is needed Implement fallback if TTS fails Optimize playback clarity B. Voice Blaster Campaign Set...