主要指导elastix asterisk 的安装配置工作，熟悉Asterisk的模块（包括体系结构，配置文件，Log日志,History）,能重新打包ISO文件
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...
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.
...system. This role is solely for testing and validation – no implementation or reconfiguration. The freelancer must sign a QA NDA before access is granted. **Role Definition** - Independent QA/verification only - No redesign or configuration changes without approval - Temporary access for testing purposes only - System configuration is frozen during the review **System Overview** - **PBX:** 3CX - **VoIP Provider:** Callcentric - **Inbound Numbers:** - Toll-Free: 1-877-359-8098 - Local: 1-914-662-0749 - **Use Case:** Educational SaaS communication **Scope of Review** Verify the following without modification: - Callcentric Configuration (DID routing, inbound logs) - 3CX SIP Trunk (registration, inbound SIP logs) - Inbound Call Flow (busi...
...every change updated in Hostaway without delay. We operate across key locations including Yas Island, Saadiyat, Reem, and Al Raha, so accuracy and speed are critical. This role focuses on two core areas: • Guest relations & customer service – clear, friendly, professional English communication via phone, email, and live chat, following our established tone of voice. Calls come through our cloud PBX, messages are handled in Help Scout, and live chat is managed via Intercom. You will be dealing with tourists, corporate guests, and families, so professionalism and patience are non-negotiable. • Bookings & reservations management – creating, modifying, and cancelling reservations directly in Hostaway, then double-checking that all dates, rates, and gu...
...every change updated in Hostaway without delay. We operate across key locations including Yas Island, Saadiyat, Reem, and Al Raha, so accuracy and speed are critical. This role focuses on two core areas: • Guest relations & customer service – clear, friendly, professional English communication via phone, email, and live chat, following our established tone of voice. Calls come through our cloud PBX, messages are handled in Help Scout, and live chat is managed via Intercom. You will be dealing with tourists, corporate guests, and families, so professionalism and patience are non-negotiable. • Bookings & reservations management – creating, modifying, and cancelling reservations directly in Hostaway, then double-checking that all dates, rates, and gu...
...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 to suggest a clean way to handle both options. Acceptance will be based on: 1. Call transfer happens with no audible delay to the caller. 2. SMS lands on my test handset within three seconds of DTMF “1”. 3. Delivery status is written back to the Vicidial database or a simple log file for later audits. If you’ve done Vicidial AGI work or Asterisk dial-plan integrations before, this should be straightforward. Please...
...inside our existing call-center system. Here’s what I’m after: • A lightweight SDK or API that can be embedded in our iOS and Android builds, listens for the customer’s voice, runs accurate speech-to-text, and hands the text off to our backend. • A routing mechanism that takes that transcript, checks caller intent or keyword triggers, and opens the correct queue/extension in our on-premise PBX (we use SIP). • A simple dashboard so supervisors can see transcripts, routing decisions, and call metrics in real time. • Clear documentation and a short demo app that proves the flow end-to-end. Acceptance criteria 1. Speech recognition latency under two seconds on a standard 4G connection. 2. Accuracy ≥ 95 % on everyday support vocab...
This banner will sit on a pull-up stand at events and retail spaces, working purely for brand awareness. The design should grab attention from a distance, introduce my logo and tagline clearly, and convey the personality of the brand in a clean, modern way—no heavy copy, just strong visuals and hierarchy. I have a rough idea of colours ...printer without extra prep. My ideal timeline is to approve final art within one week of awarding the project, with reasonable revisions along the way. Banner Trxt: STOP OVERPAYING FOR BACKDROPS. GET FAIR PRICES FROM A WOMAN-OWNED BRAND. Fair Pricing. No Tariffs. Woman-Owned. Community-Driven. We’re here to support your photo booth business, not just take your money. VISIT US FOR EXCLUSIVE PBX DEALS & COMMUNITY SUPPORT ⬇️
...off once: 1. A Teams call is captured end-to-end in both automatic and manual modes. 2. The resulting file opens only after the included decryption routine is applied. 3. A legacy pabx system call is captured end-to-end in both automatic and manual modes 4. Recording pause, resume and their related reason codes If you’ve handled audio capture for Teams, VoIP stacks or integrated with older PBX systems before, that experience will be a big plus....
I am looking for an experienced Frappe Framework / ERPNext developer to build a PBX integration module that connects Yeastar Cloud PBX with ERPNext. The goal is to replicate the functionality of a typical call‑center integration (similar to UpeoTelephony for FreePBX), but using Yeastar’s official APIs and webhook system. The module should include: • Real‑time webhook reception from Yeastar • Full call logging inside ERPNext • Click‑to‑call functionality • Incoming call popups • Automatic linking of calls to CRM records • Basic KPI dashboards and reporting Required Features 1. Yeastar → ERPNext Webhook Integration Create an ERPNext endpoint to receive real‑time events such as: • Incoming call • Outgoing call • Ans...
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 engineer would be ideal, t...
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...
...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...
Incoming calls that arrive through our SIP trunk suffer classic “one-way audio”: the caller hears us, yet we get silence. Outbound calls and all internal extension 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-...
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 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
...Setup and testing of Inbound and Outbound routing with a sample SIP provider. 2. Interface & Advanced Features: Crossbar API (Crucial): Full configuration of the Kazoo REST API (Crossbar). It must be accessible externally via HTTPS. You must verify that I can authenticate and create users via API (Postman/Curl). Monster UI: Installation and configuration of the Monster UI (Admin & User Portals). PBX Features: Enable and test Voicemail, IVR, Call Forwarding, and Call Recording. WebRTC: Configuration of Secure WebSocket (WSS) and TLS for browser-based calling support. 3. Mobile Application (Linphone White-Label): Customization: Compilation of the open-source Linphone app (iOS & Android) featuring my custom branding (Logo, Colors, App Name) and pre-provisioned SIP...
...Setup and testing of Inbound and Outbound routing with a sample SIP provider. 2. Interface & Advanced Features: Crossbar API (Crucial): Full configuration of the Kazoo REST API (Crossbar). It must be accessible externally via HTTPS. You must verify that I can authenticate and create users via API (Postman/Curl). Monster UI: Installation and configuration of the Monster UI (Admin & User Portals). PBX Features: Enable and test Voicemail, IVR, Call Forwarding, and Call Recording. WebRTC: Configuration of Secure WebSocket (WSS) and TLS for browser-based calling support. 3. Mobile Application (Linphone White-Label): Customization: Compilation of the open-source Linphone app (iOS & Android) featuring my custom branding (Logo, Colors, App Name) and pre-provisioned SIP...
...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 gateway (Twilio, asterisk, etc.) so it can pick up calls and handle simple quote bookings, even better—speed and time-saving is my goal. Deliverables I’d like to see: 1. Logo package (SVG, PNG, PDF, font info, colour codes). 2. Responsive website with my branding applied, fully tested across modern browsers. 3. Booking form and chatbot integration working end-to-end, includi...
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 Ava...
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 a press-1 autodialer that places outbound survey calls through my existing PBX over SIP. The goal is simple: play a short recorded prompt, let the callee answer a Yes/No question with DTMF, log that response, and then hang up or transfer if they press 1. It should have a good UI not very complex. Im looking for partial / ready made solutions. No 30 day waiting times. Budget depends on how good solution is.
...captures caller details, questions asked, information provided, and transfer status, then dispatches the summary via email What to include in your reply 1. A short CV or portfolio highlighting relevant conversational AI or telephony projects 2. Links or brief descriptions of at least one similar implementation (call center bots, voice IVR, etc.) 3. Your preferred tech stack—e.g., Twilio Voice, Asterisk, Dialogflow, Rasa, GPT/Llama, AWS or GCP speech services—and why it fits this use case 4. A high-level cost and timeline estimate broken down by milestones (prototype, integration, testing, deployment) I will provide API access to the listings site, SMTP credentials for email delivery, and the phone numbers to be used. Looking forward to seeing how you would...
Hi, I built a dialer using aistudio (React) and want to get the dialing functionality integrated. (Dialer - as the carrier (to bridge calls to the PSTN). This approach drastically reduces costs (Twilio Elastic SIP Trunking is significantly cheaper than the Twilio Client SDK) and gives you granular control over the call audio. The Architecture Blueprint 1. React Frontend: Uses (or similar) to connect to FreeSWITCH via WebRTC (WSS). 2. FreeSWITCH (Middle Layer): Acts as the PBX. It bridges the WebRTC stream (from the browser) to a standard SIP stream. 3. Twilio (Carrier Layer): Connected to FreeSWITCH via Elastic SIP Trunking. It takes the SIP stream and terminates it to the PSTN (real phones).
...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 connec...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 handle web/backend) Repo (prototype):
...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 packet ex...
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 stability with ...
...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...
Hi, I'm looking for an experienced VoIP developer to build a complete multi-tenant hosted PBX / UCaaS platform with real-time billing and a modern client portal. The system should be scalable, secure, and production-ready for a hosted PBX reseller business. Tech Stack & Requirements: Core PBX: FreeSWITCH + FusionPBX (latest version) True multi-tenant setup with domain isolation Superadmin + tenant admin access levels Enterprise features configured: ACD/Call Queues (strategies, agent states, callbacks, wallboards) IVR, Ring Groups, Time Conditions, Call forwarding with CC/CAP Call Recording, Conferencing, Voicemail-to-email Secure WebRTC (wss) for browser-based calling Call Center modules and reporting High availability & security (Fail2Ban, iptables, SSL)...
I’m building an on-premises, Asterisk-based IVR that does more than play menu prompts: it must pull and push data to our internal database, record every call, route callers intelligently, and pass key interaction details straight into our CRM. Scope • Design and code a custom Asterisk dial-plan (and any AGI/ARI scripts) that drives a multi-level IVR. • Implement database integration for real-time look-ups and updates—open to MySQL, PostgreSQL, or SQLite, so recommend what best fits solid performance and maintenance. • Enable call recording with configurable retention rules. • Build smart call-routing logic (skill, schedule, or data-driven). • Deliver a lightweight bridge or API hook so our CRM instantly receives caller data and ...
Network, Cloud & DevOps Engineer Experienced IT professional with strong expertise in Networking, Cloud Infrastructure, DevOps, Automation, and Security. Capable of designing, deploying, securing, and maintaining scalable on‑prem, hybrid, and cloud environments aligned with modern industry demands (2025+). CORE TECHNICAL SKILLS Networking & Security Enterprise Network Design (LAN / WAN / WLAN) MikroTik (Routing, Firewall, NAT, VLAN, QoS) VLAN Configuration, Trunking, Inter‑VLAN Routing VPNs: L2TP/IPSec, PPTP, OpenVPN, WireGuard, Site‑to‑Site & Remote Access Firewall Policies & Network Hardening IP Address Management (IPAM) Network Monitoring & Performance Optimization Cloud Computing & Infrastructure Amazon Web Services (AWS): EC2, VPC, IAM, S3, Security ...
...looking for an Engineer to design, implement, and support Asterisk-based telephony systems integrated with AI conversational voice solutions. The role focuses on hands-on engineering, system integration, and technical problem-solving. Key Responsibilities Design, deploy, and maintain Asterisk PBX environments Integrate Asterisk with AI conversational voice platforms Configure and manage SIP trunks, call routing, IVR, and call flows Troubleshoot call quality, signaling, and system performance issues Support testing, deployment, and production environments Work closely with AI, backend, and product teams to deliver voice solutions Document system architecture and configurations Required Skills & Experience Strong experience with Asterisk / ...
...peers, and relay SIP MESSAGE for text. Core requirements • Kamailio 5.5.x using mysql compiled and tuned for high-concurrency WebRTC • DMQ module enabled so this node can exchange dialog state with our existing Kamailio proxies • Full ws / wss support with correct TLS setup • SDP mangling and ICE/TURN handling so JsSIP, SIPml5 and clients connect without manual tweaks (our cloud PBX front-end uses those libraries today) • Proven interop with Odoo VoIP’s JsSIP webphone Nice-to-have If you already have a lightweight HTML/JS WebRTC softphone that exposes simple embed hooks (iframe, JS API or similar), I’d like to see it; the closer it drops into our current CloudPBX UI the better. Deliverables 1. Annotated , TLS materials an...
...usage - Final verification and acceptance by the client Contract & Payment Structure - Base Payment: USD $1,200 (implementation) - Success Bonus: USD $800 (Paid only after full completion, stabilization, and final acceptance) - Total Possible Payment: USD $2,000 - Fixed-price, milestone-based contract - No additional charges within the agreed scope Ideal Candidate - Strong experience with Asterisk / SIP / RTP - Comfortable with real-time audio processing - Experience integrating AI voice APIs (STT / TTS) - Able to deliver end-to-end working systems, not partial demos - Clear communication and responsibility for completion When Applying, Please Include - Your experience with SIP / RTP / VoIP systems - Whether you work individually or as a team - A brief descri...
...SIP integration between FreePBX and ElevenLabs AI, with Twilio used for external PSTN calls. Please do not apply unless you have deep, hands-on experience with Asterisk 20, FreePBX 17, and PJSIP. Objectives (Must Be Achieved) Scenario 1 – Internal Test • Extension 1001 calls ElevenLabs AI via SIP • ElevenLabs processes the call • ElevenLabs transfers the call back to extension 1002 Scenario 2 – External Call • Call arrives from a Twilio number • Routed to FreePBX • Forwarded to ElevenLabs AI • AI transfers the call to a live PBX extension Both scenarios must be fully working and tested Current Environment • PBX: FreePBX 17 • Asterisk: 20.x • OS: Ubuntu 22.04 • Hosting: AWS EC2 • Signali...
My 3CX PBX is correctly receiving incoming calls through our Twilio SIP trunk, yet every outbound attempt fails. due to outbound rules. i have setup this but i think i am missing one of the check mark or something it is not working
Necesito que me instalen desde cero una centralita virtual PBX VoIP basada en Issabel con el objetivo de gestionar el soporte al cliente. Yo facilitaré los datos de acceso al servidor (o la instancia cloud) y las credenciales del proveedor SIP; lo que busco es que la plataforma quede operativa y lista para recibir y distribuir llamadas internas y externas dedicadas para usuarios finales El trabajo debe incluir: • Instalación limpia de la versión estable de Issabel, con Asterisk ya integrado. • Creación de las extensiones iniciales, cola o ring-group para soporte, buzón de voz y, si es necesario, un IVR simple para enrutar llamadas entrantes. • Pruebas de llamada entrante y saliente para confirmar que todo funcione sin eco ni ...
My FreePBX install has suddenly stopped accepting any inbound calls. The configuration has not been touched for a while, yet every number on the system now rings out with no sign of life inside the PBX and no visible error messages. Outbound calls still work, trunks show as registered, but nothing is landing on the extensions. I need a seasoned FreePBX / Asterisk troubleshooter to jump in, trace what is blocking the inbound route, correct whatever trunks, inbound routes, firewall, or SIP settings are at fault, and get calls flowing again. Once it is working, I would like a brief note of what you changed so I can keep the system stable going forward. You will have full web GUI access on request; testing can be done live as this is a production box. I am in Australia, so I wi...
...enough for the first release. Deliverable checklist 1. FreeSWITCH SBC build files and working configuration with Opus transcoding enabled. 2. MariaDB schema covering clients, PBXs, numbers, and routing priorities. 3. Web portal to add / update / delete routes and instantly reload FreeSWITCH. 4. README with deployment steps, firewall ports, and a quick test plan (two SIP softphones to emulate PBX and carriers). The solution must install on a fresh Debian VM and pass a live demonstration: a US call failing over to Carrier 2 on simulated 503, and an EU call doing the opposite....
...experienced Machine Learning Engineer specialized in audio processing and deep learning. The goal is to design, train, and deploy a high-performance AMD (Answering Machine Detection) model for telephony, using an existing dataset of approximately 67,000 labeled audio samples. The model must operate in real-time with low latency, and integrate into our existing calling infrastructure (Drachtio / Asterisk / FreeSWITCH / Vicidial). Mission Responsibilities: Analyze and preprocess the existing dataset (cleaning, balancing, train/val/test split) Extract audio features such as Mel-spectrograms, MFCC, STFT, normalization Design and train a CNN/CRNN model for AMD classification (Human / Voicemail / Silence / Fax / Other if needed) Optimize the model for real-time inference (target &...
Our production VoIP app runs on Kamilio/Asterisk is suffering from three clear symptoms: call dropping, audio quality issues. The trouble appears only on a fraction of calls rather than every connection, so I need someone who can track down the underlying triggers—whether they sit in the signalling layer, jitter buffer, codec choice or network traversal logic—and then eliminate them. You’ll start by reviewing the current SIP/WebRTC stack, server logs, packet captures and any client-side metrics I provide. From there, pinpoint the root cause and apply code, configuration or infrastructure tweaks to stabilise audio flow. Deliverables • Detailed diagnosis outlining where packets or media streams fail • Fixed client or server code / configs ready for ...
I need a solid back-end that lets a future React Native client place and receive calls with the reliability of a carrier-grade PBX. The server must handle: • Incoming and outgoing calls • Call forwarding and call transfer • Voicemail storage/retrieval • A flexible auto-attendant (IVR) My preference is to stay in the React Native ecosystem for the client side, but I’m open to your guidance on the most appropriate SIP/WebRTC stack, media server (Asterisk, FreeSWITCH, Kamailio, etc.), and signalling approach. Please outline the architecture you propose, the main tech you’d employ, and an estimated timeline for delivering a first working build that can: 1. Register soft-phones via SIP or a comparable protocol 2. Complete internal an...
I need a fully-functional outbound voice agent that plugs straight into my VICIdial / GoAutoDial stack (Asterisk) and dials our CANADIAN leads through the SIP trunk we already use. Once connected, the bot must carry the entire conversation in a friendly, relaxed tone—answering common questions, keeping to our compliance wording, then moving naturally into a soft-close or handing the call over to a live agent when the prospect is ready. Key things I will be testing for: extremely low latency between speech-to-text, LLM response and text-to-speech; barge-in so the prospect can interrupt without awkward pauses; sentiment and intent tracking so the call flows feel human; and a clean handoff of every outcome, recording and transcript back into VICIdial and our CRM. Deliverables I...
I have a Cisco CP-9951 that is still running the factory SCCP load. My FreePBX instance is already up, stable, and ...the correct XML config so it boots straight into SIP and talks cleanly to FreePBX. When the flashing is done, the phone should: • Register to my existing FreePBX extension and pass audio both directions • Retain its settings after reboot • Display the correct time/date from the PBX or NTP server I also want a concise, copy-paste-ready set of instructions so I can repeat the process for any additional CP-9951 units in the future. We can use SSH, AnyDesk, or another secure remote tool for access to the PBX and TFTP host, whichever you prefer. If you have deep experience with Cisco enterprise phones, TFTP provisioning, and FreePBX, let&rs...
