Asterisk iproiecte
Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!
Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,
Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de maxim 1000$
Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de la RDS
Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la RDS
Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.
Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................
Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................
Need an experienced s specialist to configure our SIP and PRI trunks I will provide remote access to the server and trunk credentials once we agree on the approach. Looking forward to working with someone who can get this running quickly and cleanly. Asterisk PBX Linux SIP Software Architecture,Engineering VoIP
I need an experienced telephony engineer to bring up a new IVR on our Asterisk-based server that will answer calls coming in on both a SIP trunk and a PRI. The core requirement is a clean, dependable call-routing tree—callers should reach the right destination every time—with call recording and basic, built-in reporting turned on from day one. Environment • Linux box already running Asterisk (remote SSH available) • One SIP trunk + one PRI (credentials and circuit details ready) Scope of work 1. Configure the IVR in Asterisk, activate call routing options, and confirm that both trunks follow the same logic. 2. Enable call recording for all menu paths, store files locally in an organised directory structure, and verify playback. ...
...vollständig auf eine selbst gehostete VoIP-Lösung umstellen. Als Herzstück soll ein Asterisk-Server laufen, den wir in unserer eigenen Infrastruktur betreiben. Rund 11 – 50 Mitarbeitende werden die neue Anlage täglich nutzen. Folgende Kernfunktionen sind zwingend erforderlich: • Anrufweiterleitung • Voicemail • Konferenzschaltung Alle bestehenden Rufnummern sollen nahtlos portiert und auf der neuen Plattform weitergeführt werden. Neben der eigentlichen Installation erwarte ich eine kurze Dokumentation der Systemarchitektur, eine verständliche Anleitung für zukünftige Erweiterungen sowie einen Testlauf, bei dem alle Funktionen live verifiziert werden. Wenn du Erfahrung mit selbst gehosteten Asterisk-Setup...
I need an experienced PHP developer to step into an existing site and refine the current script. All the change-requests are already captured in a detailed document and I can hand it over as soon as we begin, so you will have clear guidance from day one. The tasks range from polishing user-facing elements to tightening back-end logic and edge-case handling. You’ll be working directly in the live codebase (standard LAMP stack), so clean, well-commented commits are essential. Once updates are complete, I’ll run through a quick round of acceptance testing; if everything matches the spec, we can close the task immediately. Below is an AI summary of the task to be accomplished Project Overview This project involves improving and fixing various aspects of a car renta...
I need a working proof-of-concept Voice Agent CRM whose entire logic runs on an n8n instance that I host. The system’s core goal is customer support, so the voice agent must effortlessly answer FAQs, open or update tickets, and, when necessary, transfer a live call to a human agent. Key expectations • n8n will orchestrate every step—speech-to-text, intent detection, CRM updates, and call routing—so the workflows you build have to be easily readable and fully documented inside the editor. • Scheduling flows should connect to Google Calendar; if you can keep the design flexible enough to plug in Outlook or Apple Calendar later, even better. • Tickets can live in a lightweight built-in board, a fresh Postgres table, or a customary help-d...
I am looking for someone that can write some software either for the Raspberry Pi or an ESP32 that is a SIP client that will register with a remote Asterisk server and wait for telephone calls. When a call is received the software will answer the call then hang up. If the callers number is in a list of valid numbers the software then activates a GPIO line, otherwise it ignores the call. It then goes back to waiting for a call. If you choose to write for the Pi it will be running with the SD card in readonly mode so it will have to store any variables in RAM. The list of valid phone numbers will be downloaded from a remote API - just a simple RESTful JSON client with a token for authentication over a secure (https) link. The software should refresh the list once every hour b...
I need a straightforward yet reliable auto-dialer built so I can power through large lists of cold contacts and focus on closing sales, not manually punching numbers. The core requirement is simple: once I upload or paste a list of phone numbers, the system should start calling each lead automatically, one after another, until the list is exhausted (or a preset daily cap is reached). Because these are cold leads, I want the workflow to be as efficient and compliant as possible. That means: • Quick CSV/XLS upload (or API pull) of new leads • A live screen that shows who is being dialed, call status, and basic contact details • Single-click disposition codes so I can tag each outcome on the spot (e.g., interested, call back, not a fit) ...
I’m looking for a VoIP specialist to give us a working, no-frills auto-dialer and press-1 IVR so we can run a short international customer survey. I’m flexible on the platform—Vicidial, GoAutodial, Asterisk, or another open-source dialer is fine as long as it is stable and cost-effective. Here’s what I need delivered: • Install and configure a predictive dialer on our VPS (or advise on the most economical hosting option). • Load our CSV contact list and enable reliable international dialing. • Upload the survey audio (we’ll supply the file in the target language) and build a simple press-1 IVR. • Show the caller our dedicated VoIP number as the outbound ID. • Transfer every “press 1” response in rea...
I need a self-contained voice-based AI that can place and receive calls, answer common customer-service questions, and run simple outbound sales or follow-up campaigns without routing through an existing CRM or VoIP platform. The whole stack—telephony, speech-to-text, language model, text-to-speech, and call logic—must run as a standalone service I can deploy on my own cloud account. Core scope • Two conversation tracks: inbound support (order status, basic troubleshooting, FAQs) and outbound promotional or warm-lead calls. • Natural language understanding good enough to handle unscripted replies, stay on topic, and gracefully hand off to a live agent when confidence drops. • Simple dashboard or logs so I can review transcripts, call o...
An Ubuntu 22.04 VPS is ready and waiting; all network, firewall, and base OS tweaks are already in place. Your job is to log in, install the latest stable release of Asterisk, layer FreePBX on top, and hand back a clean, well-documented system that is ready for production use. Core expectations • Compile or package-install the current stable Asterisk build, then bring up FreePBX with its standard dependencies. • Create a pair of test SIP extensions, register them from any soft-phone, and prove audio flows both ways (RTP checked). • Confirm the server can originate and receive a call path so we know outbound and inbound are structurally sound (a test trunk or echo test is fine). • Leave every command, config path, and tweak recorded in a simple m...
I have a fresh Google Cloud Platform VM ready and need a complete ViciDial installation and configuration focused strictly on inbound call handling. No CRM or help-desk integration is required; this will run as a standalone system. Here’s what I expect from you: • Clean install of the latest stable ViciDial/Asterisk stack on my GCP instance. • Secure network and SIP settings, including firewall rules and SSL where appropriate. • Create and test one inbound campaign, queue, DID, and agent login to prove calls flow end-to-end. • Provide concise notes or a screen-share walkthrough showing how I can add agents, numbers, and recordings in the future. I’m more comfortable communicating in Telugu, so a Telugu-speaking engin...
I want to replace our scattered tools at PM Solar NGO with one streamlined, largely open-source workflow that looks after every public touch-point. Calls – Inbound and outbound VoIP managed through a virtual PBX (Asterisk / FreePBX / FusionPBX or any comparable free solution). – Rules for smart call routing, automated responses when the line is busy, and voicemail recordings that arrive instantly in a shared email inbox as audio plus transcription. Email integration The same inbox should capture website forms and direct messages so we can categorise and reply from one place. The driving objective is to manage inquiries efficiently, so labels, tags or a simple ticket view are essential. Social media automation Using free-tier or self-hosted tools such ...
I run a small Asterisk lab and I need to run controlled test calls where the caller ID can be set to any value I choose. The goal is strictly testing and development, so everything must remain inside a legal, closed-loop environment. Here is what I need from you: • Configure or show me how to configure my existing Asterisk 18 instance so I can present arbitrary caller IDs on outbound calls. • Supply any dial-plan snippets, AGI scripts, or CLI commands required, along with clear commentary so I understand why each line is there. • Recommend a SIP trunk or gateway configuration that reliably passes the spoofed CLI without rewriting it (I am open to using my current trunk or switching to one you suggest). &b...
My organisation needs a dependable VoIP solution focused on crystal-clear international calling delivered entirely over the internet. I want a provider who can activate service quickly, supply SIP credentials, and guide me through basic configuration on soft-phones and standard IP desk phones. Reliability and audio quality are top priorities, so please outline the codecs you use, typical latency figures, and any uptime guarantees you can share. While my immediate requirement is outbound and inbound international voice, the platform should leave room to enable options such as video conferencing or voicemail later without a major migration. Expected deliverables: • Active account with international calling enabled • Step-by-step setup documentation (including screens...
...manage simple tasks (Change extension names, reset voicemail passwords, view/listen to their own call recordings). * Clients must NOT have access to trunk settings, routing logic, or other clients' data. * Security Hardening: * Fail2Ban configuration. * SIP TLS / SRTP encryption setup. * Firewall rules to prevent toll fraud. Required Skills * Core: FreeSWITCH, FusionPBX, or advanced Asterisk (VitalPBX Carrier / Thirdlane). * Networking: SIP, RTP, NAT traversal, TCP/IP. * Provisioning: Experience with endpoint managers and RPS (Redirect Provisioning Service) for Yealink/Poly. * Database: PostgreSQL / MariaDB management for CDRs. Screening Questions (Please answer in your proposal) * Which specific software stack do you propose using for this project (FusionP...
I need an experienced VOIP and SIP Engineer. I have developed a custom AI Voice Calling Bot and it is currently connected with Twilio. Whereas my plan is to connect this AI Bot with almost any SIP provider. When i call from Twilio using my AI Bot, then it works correctly. Here i am needing you 1. I have installed Asterisk on my AWS. It is fully configured and making outbound calls. 2. My AI Calling Bot works with web hooks, and I have created another instance where I have created an outbound webhook in Node. And the AI Calling bot's webhook is placed in this code 3. When I call from Asterisk, the outbound call works, but there is a sharp noise in the call, and nothing else. My AI Bot should be listening in the ...
I already run a fully-functional Forex trading platform; now I want to layer an AI-driven sales agent on top of it so every prospect or new client gets real-time, round-the-clock help. The agent must pick up incoming phone calls or web chat requests, walk the user through account signup, secure their first deposit, and stay with them through their very first trade. Core flow • Voice & text: the same agent should answer a telephone line (Twilio, Asterisk, or a solution you recommend) and the web chat widget on our site or mobile app. • Signup assistance: collect KYC data, push it through our existing REST API, confirm the account is opened, and report success back to the user. • Payment handling: guide the user through card, wire, or crypto dep...
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...
We have set up an Asterisk server integrated with OpenAI, and while it functions reasonably well, we are not satisfied with the voice quality. At the moment, it sounds more like an old 1980s cassette recording than a modern 2026-quality voice. We are therefore looking for a partner who can help improve the voice quality. We have already tested several solutions currently available on the market. Before selecting a partner, it is a strict requirement for us that you can provide access to a live server setup that we can test by calling a real phone number. We are not interested in solutions that cannot be demonstrated through a previously deployed and functioning server. Please only submit an offer if you can meet this requirement.
My Vicidial survey campaign already asks contacts to “press 1” if they want to speak with an agent. I now need that single key-press to do two things at once: • instantly push the call into the next available agent queue, and • fire a promotional SMS to the same phone number through a bulk-SMS provider’s API. I will be purchasing the SMS package myself, so you can work with whichever HTTP/JSON or HTTPS POST gateway the provider offers. The core work is to add or adjust the AGI / server-level scripting inside Vicidial (Asterisk) so the DTMF event triggers the API call, logs the result, and keeps normal call flow intact. I’m open to advice on whether the SMS text should stay static or be templated dynamically—feel free t...
I have a fresh Google Cloud Platform VM ready and need a complete ViciDial installation and configuration focused strictly on inbound call handling. No CRM or help-desk integration is required; this will run as a standalone system. Here’s what I expect from you: • Clean install of the latest stable ViciDial/Asterisk stack on my GCP instance. • Secure network and SIP settings, including firewall rules and SSL where appropriate. • Create and test one inbound campaign, queue, DID, and agent login to prove calls flow end-to-end. • Provide concise notes or a screen-share walkthrough showing how I can add agents, numbers, and recordings in the future. I’m more comfortable communicating in Telugu, so a Telugu-speaking engin...
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 ...
...and permissions. • Build out role-based dashboards, KPIs and reports that give sales, support and management the insights they need at a glance. • Design custom workflows and automation covering lead-nurture sequences, pipeline stage movements, ticket escalation and any approval loops that reduce manual touchpoints. • Integrate all daily-use channels: IMAP/SMTP email, WhatsApp, telephony (Asterisk/3CX), website lead forms and our analytics stack (GA4, Hotjar, custom SQL). • Migrate legacy data (contacts, deals, historic tickets and custom fields) from spreadsheets and our current SaaS CRM, cleaning duplicates on the way in. • Deliver short, role-specific training sessions plus concise SOP documents to drive internal adoption. • Provide tw...
...traffic are perfectly clear, so the problem is isolated to inbound SIP traffic only. I need someone comfortable with Asterisk, FreePBX, RTP/NAT behaviour, and typical firewall or trunk mis-configurations to jump in, trace what is happening, and implement a permanent remedy without interrupting live service. Remote SSH or VPN access to the PBX and edge firewall can be provided immediately, along with console access to the SIP provider’s portal for any trunk changes. Acceptance means: • Two-way audio is restored on every new incoming call test. • No regression on outbound or extension-to-extension calls. • A brief changelog explaining what you adjusted (trunk settings, NAT rules, port forwards, codecs, etc.) so I can document and roll back i...
...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 need an experienced Asterisk programmer to setup and maintain a custom IVR for an Automated Third-Party Verification Service. Key tasks include: - Implementing a detailed IVR menu flow that I will provide - Troubleshooting existing issues - Regular updates and maintenance - Adding new features as needed Ideal skills and experience: - Strong proficiency in Asterisk - Experience with IVR systems - Problem-solving skills - Ability to interpret and implement detailed plans
... First, I need a sleek, modern-and-minimalist logo that scales from a truck door to a favicon without losing clarity. Once the branding is locked in, the same look and feel should carry through the entire site so visitors instantly recognise us. The website’s single, overarching purpose is to turn visitors into booked jobs. To do that I’ll need: • a friction-free online booking form that lets clients choose a date, leave project details and instantly receive a confirmation email, • an AI-powered chatbot that can guide prospects, answer common questions and, where possible, schedule quotes automatically, • clean on-page SEO for our local concrete keywords so we rank in maps and organic results. If you can connect the chatbot to a voice g...
We are looking for a freelance talent acquisition specialist / technical recruiter to help us identify, screen, and introduce qualified PBX / VoIP engineers, preferably with Asterisk experience, who are available to work in Oman. This is not an engineering role. Your task is to find the right engineer, verify technical capability, and connect them with us. Responsibilities Search for and source PBX / VoIP engineers (Asterisk preferred) Target candidates willing to relocate to Oman Screen candidates for: Hands-on Asterisk / VoIP experience SIP, trunks, gateways, call routing, codecs, NAT, firewalls Real deployment experience (not just theoretical knowledge) Conduct initial interviews and technical vetting Share shortlisted candidates with: CV / profile Availability Expe...
I need a freelancer to install and configure a Vicidial/Asterisk dialer on my cloud server running Ubuntu. The tasks include configuring SIP trunks, setting up basic campaigns and agents, and securing the server with firewall and Fail2Ban. The project must be completed within the same day (post-lunch) in Hyderabad timezone.
I need an end-to-end AI voice agent that can pick up PSTN calls, greet prospective buyers in natural language, and instantly answer questions about my real-estate listings by pulling live data from the website. When callers press a designated keypad key, the system must hand the call off to a human agent without any audible delay. After each interaction, a concise call summary should be generated automatically and emailed to the assigned real-estate agent. Core requirements • Speech recognition and voice synthesis for fluid, human-sounding English and polish conversations • Real-time query to the listings database or website API to fetch price, location, availability, and other details while on the call • Key-press detection to trigger a warm transfer to a live...
...Phase 1.2, which focuses on public-facing views, Joomla integration, and deployment. This is not a greenfield project. Current System (Already Built) Python + Flask application Admin UI for: Uploading Playbill PDFs Parsing cast & crew Human review / edit / approve Structured MySQL schema Categories handled: Cast, Ensemble, Swings, Dance Captain, Understudies Equity (union) detection via asterisk (*) Clean, working proof already demonstrated Phase 1.2 Scope (What You Will Build) 1. Public Read-Only Views Actor profile pages Show pages Theater pages IMDb-style navigation between them 2. Actor Profile Enhancements Credits grouped by discipline Equity indicator per show (not global) Simple USA map showing theaters an actor has worked at Read-only No advance...
...Phase 1.2, which focuses on public-facing views, Joomla integration, and deployment. This is not a greenfield project. Current System (Already Built) Python + Flask application Admin UI for: Uploading Playbill PDFs Parsing cast & crew Human review / edit / approve Structured MySQL schema Categories handled: Cast, Ensemble, Swings, Dance Captain, Understudies Equity (union) detection via asterisk (*) Clean, working proof already demonstrated Phase 1.2 Scope (What You Will Build) 1. Public Read-Only Views Actor profile pages Show pages Theater pages IMDb-style navigation between them 2. Actor Profile Enhancements Credits grouped by discipline Equity indicator per show (not global) Simple USA map showing theaters an actor has worked at Read-only No advance...
I need an Android GSM app that stays always running and connects to my web app to stream real-time audio both ways. Flow: - Website connects to Android phone over Wi-Fi - Android phone makes / receives real GSM calls (SIM) - Call audio is streamed to the website - Website audio is injected into the live GSM call - Full-duplex, low-latency audio. Requirements: - Android app in Kotlin - Foreground service, auto-start - GSM call control + audio routing - WebRTC for real-time audio - Node.js / signaling integration - Asterisk knowledge is a big plus Stack Web: , Node.js, TypeScript (my side) Android: Kotlin, WebRTC VoIP: WebRTC, possibly Asterisk Notes: This is not a VoIP app It’s a GSM ↔ Web audio gateway You’ll work alongside me (I...
...voicemail và ghi âm cuộc gọi – nói cách khác, tôi muốn toàn bộ bộ tính năng tiêu chuẩn được kích hoạt. Bạn có thể làm việc qua SSH; quyền root đã sẵn có. Sau khi hoàn tất, giao diện quản trị web của FreePBX phải truy cập được, các extension mẫu gọi thử thành công và file ghi âm lưu đúng thư mục chỉ định. Deliverables cụ thể • Cài đặt FreePBX phiên bản ổn định mới nhất trên Debian 12 hiện hữu, kèm Asterisk. • Kích hoạt, cấu hình cơ bản ba dịch vụ: Call Management, Voicemail, Call Recording. • Kiểm thử: tạo ít nhất một extension SIP, thực hiện cuộc gọi nội bộ, kiểm tra...
...standares del sistrema. Entrego acceso de solo lectura a la BD y a la documentación del esquema. Considero terminado el proyecto cuando: 1. La consulta se actualiza sin errores con los datos correctos de cualquier rango de fechas que indique. 2. Se incluyen instrucciones breves para que el equipo de TI pueda cambiar credenciales o rutas si fuese necesario. Si ya has trabajado con Issbel, Asterisk o un stack similar de call center y dominas SQL, tu propuesta será prioritaria....
Incoming calls reach the phones, Intermittent issues such as one-way audio where caller can hear us but we cannot be heard. The signalling path is Kamailio acting purely as the control-plane SIP proxy, then the media is anchored by a FreeSWITCH + Asterisk B2BUA cluster. Only inbound legs show the problem; outbound audio is clean. I need a seasoned VoIP troubleshooter to: • trace SIP and RTP on all hops (Kamailio, FreeSWITCH, Asterisk, edge SBC) • pinpoint why RTP from the caller side never makes it to the far end (NAT, codec negotiation, rtpengine mis-pinning, firewall, wrong c= line, etc.) • supply the minimal configuration changes or firewall rules to restore full two-way audio without disrupting live traffic Acceptance will be: 1. SIP pa...
My FreePBX install is up and running, yet several extensions refuse to register or place calls. I need someone who knows FreePBX and the underlying Asterisk CLI inside out to jump in, trace what is blocking these extensions, and get them working again. You will have full access to the web GUI and, if required, SSH so you can dig into SIP settings, PJSIP/Chan_SIP conflicts, firewall rules, or any module mis-configuration that might be behind the problem. Once the extensions come back online I’d like a brief rundown of what you found and the steps you took, so I can avoid the issue in future. Deliverables • Diagnose the exact cause of the extension failure • Restore normal registration and two-way audio for each affected extension • Confirm s...
...Phase 1.2, which focuses on public-facing views, Joomla integration, and deployment. This is not a greenfield project. Current System (Already Built) Python + Flask application Admin UI for: Uploading Playbill PDFs Parsing cast & crew Human review / edit / approve Structured MySQL schema Categories handled: Cast, Ensemble, Swings, Dance Captain, Understudies Equity (union) detection via asterisk (*) Clean, working proof already demonstrated Phase 1.2 Scope (What You Will Build) 1. Public Read-Only Views Actor profile pages Show pages Theater pages IMDb-style navigation between them 2. Actor Profile Enhancements Credits grouped by discipline Equity indicator per show (not global) Simple USA map showing theaters an actor has worked at Read-only No advance...
