Asterisk pbx일자리
...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...
...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. Key requirements • Channel: Phone support only (no email or live chat for now). • Functionality: Automated responses are mandatory; call routing or recording co...
My NEC SV9500 PBX refuses to come back online after a routine restart. The system hangs during boot; I do not see telephones registering and I’m not sure whether any error codes flashed on the console because I wasn’t watching closely. I need an engineer who knows the SV9500 architecture, maintenance commands, and PCPro (or equivalent) to: • Connect remotely (or guide me onsite) to diagnose why the call-processing software is not loading. • Identify and clear any database, card, or processor faults preventing a full start-up. • Bring the PBX to an operational state where extensions are live and calls flow normally. Acceptance will be when the system completes its boot cycle, endpoints register, and we can place internal and external calls...
...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. 3. Outbound dialler integrated and functional. 4. Admin user can add prompts, edit menus, and r...
...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...
We are looking for an experienced 3CX technician to set up and configure a small business phone system. The system is already partially configured but requires a clean setup and proper routing. Current Setup Platform: 3CX (Cloud, v20) SIP Provider: Intertelecom Existing PBX (Intertelecom) may still be active Mobile usage via 3CX app Requirements 1. SIP Trunk Configuration Proper setup of SIP trunk with Intertelecom Resolve registration conflicts (single registration issue) Ensure stable inbound & outbound calling 2. Users / Extensions Create 4 extensions/users Configure mobile apps (3CX client) Set up voicemail for each user (or central voicemail) 3. Call Routing Outbound rules (basic + prefix handling if needed) Inbound routing: Ring Group (all users) Optional DID-based routi...
I need a fully-functional auto dialer built for my company and I want it up and running fast. The core requirement is seaml...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 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 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—whi...low-latency ASR supporting Hindi, English, and hybrid sentences • On-the-fly accent recognition, noise suppression, and live transcript display • Conversation logic builder so I can tweak responses myself without coding • Exportable call logs (audio, transcript, call outcome) in CSV/JSON I’m open to whichever speech engines, LLMs, or telephony APIs you prefer—Google, Azure, ElevenLabs, Asterisk, Twilio—so long as the final product delivers the sm...
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...
...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 to the chat once the call ends. I already co...
I am ready to move a production-grade, real-time contact-center platform that runs on Asterisk v18/21. The goal is to handle both inbound and outbound calls while keeping conversational latency under 800 ms at the 95th percentile, even when 300+ sessions are live. The heart of the system is an AI layer that must combine speech recognition, natural language understanding, and speech synthesis so callers can interrupt (barge-in) naturally and still maintain contextual continuity. Low-latency, full-duplex audio streaming is essential, and the code you deliver should expose clear hooks for campaign management, operator escalation, and horizontal scaling. Operational visibility is just as critical: metrics, structured logs, tracing, and an admin console all need to be in place so ...
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 alternative you trust, as long as the f...
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 ________________________________...
...NLP / Intent Recognition Rasa, Dialogflow ES/CX, or custom transformers (e.g., BERT fine-tuned) WhatsApp Integration WhatsApp Business API (Meta) – via Twilio, WATI, or official API Voice Bot Google Dialogflow CX Phone Gateway / Amazon Lex + Polly / Asterisk AGI Backend Microservice Python (FastAPI/Flask) or Node.js Deployment AWS Lambda / EC2 / DigitalOcean (or any cloud) CRM Integration REST API calls to Zoho CRM (OAuth 2.0) Must-have experience: Built at least 2 production WhatsApp bots with negotiation logic. Integrated voice bots with Asterisk/VICIdial. Strong understanding of state machines for conversation flow. Experience with dynamic pricing or e-commerce bots preferred. We have developer working for vicidial and zoho with whom you can collaborte and wor...
My React frontend for a soft-phone style VoIP app is already designed; the missing piece is a clean, reliable bridge into FusionPBX. I need you to wire the two together so that every key action in the UI is backed by live PBX logic. Scope of the integration • Call management – place, receive, transfer and hang-up calls directly through FusionPBX. • User registration – authenticate users against FusionPBX and pull their extension data on login. • Config management – let admins tweak extension or trunk settings from the app and push changes back to FusionPBX instantly. • Real-time updates & notifications – UI must refresh on state changes (ringing, answered, missed, registration status, config saves) without manual refresh. Wha...
My React frontend for a soft-phone style VoIP app is already designed; the missing piece is a clean, reliable bridge into FusionPBX. I need you to wire the two together so that every key action in the UI is backed by live PBX logic. Scope of the integration • Call management – place, receive, transfer and hang-up calls directly through FusionPBX. • User registration – authenticate users against FusionPBX and pull their extension data on login. • Config management – let admins tweak extension or trunk settings from the app and push changes back to FusionPBX instantly. • Real-time updates & notifications – UI must refresh on state changes (ringing, answered, missed, registration status, config saves) without manual refresh. Wha...
I need an experienced adviser who feels completely at home on the phone, because the role is 100 % voice-based. You’ll be the first and last touchpoint for our consumer-goods customers, fielding inbound queries and making outbound follow-ups whenever a case needs an extra push toward resolution or a courtesy check-in. Here’s what the day-to-day looks like: you’ll log into our cloud PBX and CRM (we use Zendesk + Aircall), greet callers, capture the issue in real time, walk them through basic troubleshooting or product-use advice, and close the ticket with clear notes so the rest of the team can see exactly what happened. For outbound calls, you’ll reach out on missed contacts, delivery-status updates, or proactive satisfaction surveys. All conversations must ...
...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 refi...
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 Asterisk verbose ...
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/SRTP secur...
My 3CX system runs flawlessly for on-site extensions, yet every time we place or receive external calls the line drops or never connects. I need an experienced 3CX specialist to dive straight into troubleshooting and eliminate these connection problems. You’ll get full remote access to the PBX (version 18 on Debian), SIP trunk details, and firewall console. I’m looking for someone who can: • Analyse logs and packet captures to spot what blocks or misroutes the calls • Adjust firewall rules, SIP trunk parameters, STUN or SBC settings as needed • Test several scenarios until every external call completes with stable audio • Leave me with a brief summary of the root cause and the exact changes you made Internal calls are fine, so the focus is ex...
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.), and the test...
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. • Please include brief instruction...
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 This is the text for 1 of 157 Scripture References: Bereshith, Genesis, chapter 1, ver...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 done put all t...
I need a complete, from-scratch VoIP solution for my Australian call centre and I need it running fast. Nothing is in place yet—no trunks, no PBX, no numbers—so you will be taking the project from zero to a fully operational system that can reliably handle 11-50 concurrent calls. I will release payment only once everything is live, tested, and I have unrestricted admin access. Scope • Select and provision an Australian-compliant SIP trunk or carrier. • Install and configure the PBX (cloud, on-prem, or hosted—recommend what fits best). • Map DID numbers, outbound rules, and fail-over paths. • Run end-to-end tests to confirm audio quality, caller ID, and concurrency limits. • Provide all credentials, written documentation of...
Je cherche un spécialiste VoIP capable de prendre en main, à distance, la configuration et la mise en service d’un PBX Yeastar P550 déjà installé sur site. L’objectif prioritaire est de faciliter le contact avec nos clients ; toute la logique d’appels sera donc centrée sur une gestion fluide des appels entrants. Ce que j’attends : • Paramétrage complet du P550 : création des extensions, files d’attente, règles entrantes, messages vocaux, horaires, etc. • Intégration du Grandstream HT813 pour reprendre une ligne analogique existante. • Les 4 postes Yealink VP59 de l'installation sont déjà enregistrés dans le P550. • Vér...
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/f...instance and 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 will configure DID in a new pbx , i will install all
I’m ready to bring my Grandstream environment online and need a specialist who can jump straight into configuration. Both the PBX and the VoIP handsets are on-site, factory-reset, and reachable for remote access. Your task is to register every phone to the PBX, build the dial plan, and then load my custom call-waiting package—this includes a spoken message I’ve already recorded as well as a separate music-on-hold track. Key deliverables • Register and provision all Grandstream VoIP phones with the PBX • Configure extensions, inbound/outbound routes, voicemail, and caller ID as needed • Upload my provided audio files, assign the custom message to call-waiting, and set the music track for hold scenarios • Verify the announceme...
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 stack ...
...routing communication through the unified system. 3. Technology Stack Layer Technology ERP Odoo 18 Enterprise Hosting Telephony Twilio Voice SMS Twilio SMS WhatsApp Twilio WhatsApp AI OpenAI Middleware NodeJS + Python Database PostgreSQL (Odoo default) 4. System Architecture Client Communication Channels Phone calls WhatsApp messages SMS ↓ Twilio Communications Layer Twilio Voice (PBX + IVR) Twilio SMS Twilio WhatsApp ↓ Middleware Integration Layer ↓ Odoo 18 Discuss (communications hub) CRM Recruitment Service workflows Projects Contact database 5. Twilio Configuration The system will use a Twilio EU phone number as the central communications endpoint. The existing public mobile number will forward incoming calls to this Twilio number. Requ...
Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File:...incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills: PHP, Software Architecture, Asterisk PBX, MySQL, JavaScript See more: making money with international premium rate numbers, iprn telecom, iprn providers, international premium rate numbers providers, international...
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 concise post-deployme...
I want my new AWS account configured to host a USA-based SIP trunk through Amazon Chime. The job includes creating or hardening the AWS environment, provisioning an Amazon Chime Voice Connector, assigning U.S. numbers, and enabling both inbound and outbound calling. The trunk has to register cleanly with our existing on-premise PBX and soft-phone apps, so please allow for SIP signaling tweaks, codec selection, and basic security (TLS, SRTP, firewall rules). Once the link is live, I’ll need brief documentation of the key settings so we can maintain it internally. Deliverables • AWS account prepared with least-privilege IAM roles • Amazon Chime Voice Connector fully configured and licensed for U.S. PSTN access • SIP trunk integrated and test calls comple...
...Experience 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 Exp...
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 ar...of the call flow with all customer touch-points • All audio files (or TTS scripts) in WAV and MP3 • Working configuration files or portal access so my team can maintain the menus • A brief test report confirming that automated responses, live-agent transfers, and call-back requests all succeed under real call conditions If you’ve set up similar customer-service IVRs before—especially on Asterisk, Twilio, FreePBX, or a comparable platform—let me know. I’m ready t...
...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 customised to our brand. Deliverables • Fully functional IVR map with five departmental route...
...skonfiguruje połączenie z moim dostawcą SIP trunk – Zadarma – tak, aby system bezbłędnie realizował zarówno połączenia wychodzące, jak i odbierał przychodzące. Zakres prac: • dodać i zweryfikować trunk Zadarma, • utworzyć odpowiednie dial-plany i reguły routingu, • przeprowadzić testowe połączenia w celu potwierdzenia jakości oraz stabilności, • wprowadzić ewentualne korekty konfiguracji serwera Asterisk wchodzącego w skład GoAutoDial. Środowisko obejmuje jeden VPS, więc konfiguracja dotyczy pojedynczego serwera. Po zakończeniu proszę o krótką instrukcję krok-po-kroku, abym mógł samodzielnie zarządzać podstawowymi zmianami w przyszłości. Jeżeli masz doświadczenie z GoAutoDial i integracjami SIP trunk, daj znać, ile cza...
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 usin...
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 need a complete on-hold production for our PBX: a warm, friendly female voice delivering the three short scripts below, separated by ambient background music beds of roughly thirty seconds each. The finished mix should run somewhere between one and two minutes in total and be supplied as a single 16-bit, 8 kHz WAV file ready to load straight into the system. Script to voice: “Thank you for calling Andrew’s Tyre and Mechanical North Lakes. All of our operators are busy assisting other customers and we will be with you shortly.” —30 s ambient music— “Did you know Andrew’s Tyre and Mechanical has been locally owned and operated for the past 30 years? Now that’s service!” —30 s ambient music— “Andrew’s Tyre...
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, credentials cre...
My organization is looking for a telephony tech in the Saskatoon, SK to connect a Grandstream ATA to an Avaya IP Office system. The primary objective is to move two existing fax numbers—306-934-5787 and 306-955-3059—onto the ATA so they send and receive reliably through the PBX. The job covers: • Physically or remotely provisioning the Grandstream ATA, assigning it an internal extension, and ensuring it communicates correctly with Avaya IP Office. • Mapping both fax numbers to the ATA ports and confirming successful inbound and outbound fax transmission. • Providing a brief record of any IP Office or Grandstream settings you change so I can reference them later. Acceptance is complete when both fax lines pass test sends and receives without errors.
...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 development, but I’m also open to a custom SIP-compatible solution. Cloud-hosted, on-prem, or hybrid deployment can be discussed; reliability, low latency, and call quality are non-negotiable. For deliverables, I’ll need: 1. A working prototype handling live calls. 2. A simple...
I run a growing small business that relies on Microsoft 365 and a cloud-based VoIP phone system. I’m looking for a dependable partner who can step in as our day-to-day IT resource, keeping both environments running smoothly whi...keep us aligned. Deliverables • Same-day response to support tickets during business hours • Resolution of Microsoft 365 and VoIP incidents or escalations • User onboarding/offboarding completed within agreed timeframes • Monthly health report outlining work performed, open issues, and improvement ideas A solid grasp of Azure AD/Entra ID, Exchange Online, Teams admin, and common hosted PBX platforms will help you hit the ground running. If this sounds like your wheelhouse, let’s talk about how we can keep my tech ...
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 completes REGIS...
