Asterisk wms jobs

Troubleshoot asterisk trunk TLS/SRTP

...(password) for each extension. Enable basic features for each extension as needed (Voicemail, Call Waiting, Recording). 4. Testing & Handover Verify two-way audio for both internal (extension-to-extension) and external calls. Deliver a "System Access Sheet" containing all IPs, admin logins, and extension credentials. Freelancer Requirements Expertise in Asterisk/Issabel/FreePBX configuration. Prior experience setting up SIP Trunks (e.g., Twilio, Telnyx, ). Solid understanding of NAT traversal and firewall management for cloud-hosted VoIP. Deliverables A fully functional Issabel PBX on AlmaLinux 8. Configured and tested SIP Trunk for inbound/outbound calls. Set of active extensions ready for softphone or desk phone registration. Documentation of

...vasoconstriction. Search: “cold induced vasoconstriction skin diagram” SLIDE 6 — FREEZING INJURY MECHANISMS On screen Ice crystal formation Cell membrane disruption Microvascular thrombosis Inflammatory cascade (Sheridan et al., 2022) Visual Cell damage diagram. Search: “ice crystal cell damage frostbite diagram” SLIDE 7 — FREEZE–THAW–REFREEZE On screen Freeze–thaw–refreeze worsens injury Strong WMS recommendation avoid refreezing (McIntosh et al., 2024) Visual 3-step diagram Freeze → Thaw → Refreeze SLIDE 8 — ENVIRONMENTAL RISK FACTORS On screen Wind chill < −27°C frostbite < 30 minutes (Castellani et al., 2021) Wet skin freezes −0.6°C (ACSM Position Stand, 2...

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...

...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...

Please include your work examples in the first. If you don't have proven record, don't bid. ___________________________________________________________ We are building a modular AI reasoning layer that embeds directly into enterprise systems (ERP, CRM, WMS, proprietary platforms). This is not a chatbot or simple LLM wrapper. It is a real-time, context-aware decision runtime with explainable signal generation and adaptive feedback logic. We are looking for an engineer who has already built advanced AI infrastructure, not just integrated APIs. To Apply, You Must Provide Examples Of: AI systems embedded inside enterprise platforms Custom inference orchestration or multi-model routing Real-time signal or decision engines Explainability layers (confidence scoring, reason...

...(12–18 months) Start: As soon as possible Languages: French and English (mandatory) Context Purever Group is executing Dagard360, a major ERP transformation programme for our industrial site in Boussac, France. The project includes: ERP standardisation and process redesign Separation of industrial ERP and Design & Build tooling Elimination of legacy pricing structures (“nine codes”) Logistics and WMS standardisation Alignment with our group data platform (Helix) Preparation for a future single-tenant architecture model The current go-live target is March 2027. We are seeking an experienced freelance ERP Project Manager to reinforce the existing project structure, reduce execution risk, and ensure leadership continuity. This role will also provide covera...

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...

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

...guidance. This is a fast-turnaround project. Scope of Work: Label Updates You will be working from our existing AI file and implementing the following updates: Insert updated Supplement Facts Panel (we will provide content) Change wording: “Vegan Product” → “Vegan Friendly” Add “Equivalent to” before “1500 MG High Potency” Bold the following: Gummies quantity “Dietary Supplement” Add an asterisk (*) to the title: “Longevity & Vitality*” Enlarge the vertical logo for a stronger brand presence Replace subtle grape imagery with a more accurate visual direction (product is sourced from Japanese Knotweed, not grapes). Open to hearing/seeing a couple of suggestions, and I have a quick idea....

...demonstrate a sample conversation in Hindi. 2. Viewers can open a provided URL and hear the call with <3 s delay. 3. When I speak over the bot, it stops, acknowledges, and either answers or routes to the operator logic you deliver. 4. All conversation text and events appear in the dashboard and in a downloadable JSON log. Tech is flexible—Dialogflow CX, Rasa, Kaldi, Vosk, Twilio Voice, Asterisk, WebRTC, FFmpeg, OBS-style RTMP pipelines—use whatever delivers the smoothest Hindi recognition and low-latency stream, but keep licensing clear for commercial use. Tell me how you would architect the speech pipeline, manage interruptions, and keep the audio stream in sync. If you’ve built similar multilingual voice or streaming tools before, a quick demo link...

I am upgrading an Asterisk 20 installation to version 22 I compiled and installed the code ok, but when trying to run alembic to upgrade the database it fails with various errors. I need someone that understand alembic and Asterisk to fix the database so that alembic runs cleanly and future upgrades will work. The important thing is that the existing database does not lose any data as it is a live system.

...need to know up front: • Environment: fully cloud-hosted, no on-prem gear • Media: SRTP from Asterisk, RTP from 3CX • Symptom: recipient can hear, but caller cannot Your task is to trace the SRTP→RTP path, pinpoint why one leg is silent, and adjust the PJSIP, codec, or NAT/STUN settings so we get clean two-way audio. Once it works, I also need a short write-up of the changes so I can replicate them in staging. Acceptance criteria 1. Two-way audio confirmed on at least five consecutive outbound WhatsApp calls placed from 3CX 2. Updated PJSIP/SRTP configs returned to me (or pushed to my repo) 3. Quick walkthrough or screen-share explaining the fix Tools you’ll likely touch: Asterisk 20+, PJSIP, Wireshark, sngrep, 3CX console. SS...

...Agent transfer notification Recording link notification 5. Backend System Developer should build a backend service that will act as middleware between: Custom CRM Voice Bot Platform Ameyo Dialer Server Technical Requirements Strong Linux experience MySQL database experience Backend development experience REST API development Experience with telephony systems preferred Experience with Ameyo, Asterisk, or Vicidial preferred Preferred languages: Node.js Python PHP (Laravel) Important Notes Dialing will remain in Ameyo We are building our own CRM and backend UI developer is already working on frontend SSH access to server will be provided This project requires someone comfortable working with telephony systems and Linux servers. Future work available for full dialer ...

I am looking for an experienced VoIP/PBX specialist to set up a small FreePBX-based phone system. This is a fixed-price project, and I am requesting your best offer. The system ...testing of all functions together with me Delivery of admin access credentials Short documentation of the setup and configuration Your responsibilities include recommending suitable hosting, guiding me during VPS and Twilio account creation, installing and configuring the system, and ensuring everything is fully functional. Your proposal should clearly state: Your experience with FreePBX/Asterisk Similar projects you have completed Estimated delivery time Your fixed total price Only fixed-price offers will be considered. Security and prior experience with DISA configuration are essential for thi...

...handle process optimization, team coordination, and workflow setup. Key responsibilities & skills required: Manage BPO operations: process mapping (as-is & to-be), SOP creation/revision, bottleneck identification & resolution. Team handling: coordinate small-mid size teams, performance monitoring, training & knowledge transfer. Tools & knowledge: CRM/ERP basics, call center software (e.g., Asterisk, Avaya, or similar), reporting & KPI tracking (Excel, Google Sheets, Power BI preferred). Customer support/operations experience: handling inbound/outbound processes, quality assurance, compliance & escalation management. Strong communication, problem-solving, and organizational skills. Experience: 2-4 years in BPO/call center/operations management (IT...

I am looking for a senior Vicidial / Asterisk Expert to perform a clean installation and professional optimization of a Vicidial system on a single server. Project Requirements: • Installation: Clean install of ViciBox v.11 (All-in-One) on a dedicated/VDS server. • Capacity: The system must be optimized to handle at least 100 concurrent calls on a single server without voice degradation or database lag. • Campaign Type: Configuration of a Survey (Press-1) Campaign. • Flow: Dialer calls the list -> Plays a greeting (IVR) -> If the user presses "1", transfer the call to a specific In-Group (Queue) where live agents are waiting. • CallerID Management: Proper configuration of Outbound CallerID to ensure CID is displayed correctly to customers. &b...

...hardware in place, so I’ll rely on your guidance. My preference is to run the system on a VPS; please specify the exact CPU, RAM, storage and bandwidth you consider safe for 10 simultaneous agents. If you feel a dedicated machine would offer clear advantages, outline those too and I’ll weigh the trade-offs. Core tasks • Fresh installation of the latest stable VICIdial release • Server tuning (Asterisk, MySQL, Apache, networking) for smooth outbound volume • Basic security hardening (firewall rules, fail2ban or equivalent) • SIP trunk integration and configuration, including carrier recommendations that support CLI override • End-to-end testing of dialling, recordings and reports Post-install support & timeline Let me know how...

I need a small 'n' next to an asterisk removed from 6-20 pages of a PDF document. The PDF was created in Atticus. They seem to become visible when printed on kdp. The tech has to be able to id these from the pdf doc and remove. Ideal skills and experience: - Proficiency in PDF editing - Familiarity with Atticus or similar software - Attention to detail - Ability to complete the task efficiently

I need a small 'n' next to an asterisk removed from 6-20 pages of a PDF document. The PDF was created in Atticus. They seem to become visible when printed on kdp. The tech has to be able to id these from the pdf doc and remove. Ideal skills and experience: - Proficiency in PDF editing - Familiarity with Atticus or similar software - Attention to detail - Ability to complete the task efficiently

... services, timings, status, due payments, support requests) • Make automated outbound calls (appointment confirmations, reminders, lead follow-ups, collections) • Transfer the call to a human agent when required This is NOT a keypad IVR. The assistant must understand natural spoken language from callers and respond with a natural-sounding voice. Technical Scope: • SIP/VoIP integration using Asterisk or FreeSWITCH • Real-time Speech-to-Text • Text-to-Speech voice responses • LLM/NLP based conversation handling • Call recording and call logs • Basic web admin panel Budget: The genuine project budget is as mentioned in the posting. It may be extended if required based on technical justification and developer recommendations, subject to mana...

...to minimize clicks, avoid duplicated tasks, and ensure consistent data from product creation to fulfillment and reporting. The system will serve as a unified platform for: Product Lifecycle Management (PLM&BOM) Product, Brand & Collection Intelligence (Collection & Drop Management, Style Performance Tracking, Sustainability & Compliance (increasingly important) Inventory & Warehouse Management (WMS) Order Management (OMS) B2b login (Retail management) Demand Forecasting & Replenishment Customer & Sales Management (CRM) Finance & Accounting Integration Omnichannel & Fulfillment Intelligence (Retail, DTC, B2B, Wholesale) Analytics, Dashboards & Reporting Product Catalog Management Digital Asset Management (DAM) Dashboards (Marketing, ecommer...

I need an account Vitelity, Plivo Or Flowroute and ne...or Telnyx. My goal is a clean, fully tested deployment that plugs straight into our existing PBX without surprises. What I need from you • Configure the new Vitelity account from scratch, including trunks, DID routing, outbound caller ID, and failover. • Troubleshoot any signaling, codec, or registration issues that surface during cut-over. • Integrate the trunks with our current system (Asterisk-based) and verify inbound/outbound call flow, e911, and fax-over-IP edge cases. If you’ve ever spun up Telnyx, Flowroute, or Plivo trunks, mention it—it tells me you know the quirks and can move quickly. Let’s get this live without downtime. If you have extensive experience with Telnyx's S...

I am upgrading an Asterisk 20 installation with real-time credit control. The flow is straightforward: • As soon as a user dials, the dialplan (or an AGI/ARI app—your choice) must hit our REST JSON endpoint, passing the number and an API-key header. • The endpoint replies with the exact seconds of credit. • If the reply is 0, the call never leaves the box; instead we immediately play the WAV message I will supply and hang up. • If credit exists, the call proceeds while an internal timer counts down. When the remaining credit drops below 120 s, the caller hears a short beep every 15 s. • When the timer reaches 0, the called party is released and the caller hears the second supplied WAV prompt before the channel is cleared. Everything sits on ...

...are approved and live Required Experience Hands-on experience with the WebTrit Phone Flutter app — forking, customising, or deploying it Strong Flutter/Dart development (iOS + Android) SIP/VoIP experience — configuring SIP trunks, understanding call flows, WebRTC Python/FastAPI for the BSS adapter App store submission (Apple + Google) Nice to have: Telnyx or , WooCommerce REST API, Asterisk/FreePBX Budget & Timeline Open to proposals (fixed-price per phase ) Ongoing maintenance likely after launch HOW TO APPLY — YOU MUST ANSWER THESE QUESTIONS Proposals that don't answer ALL of the following will be deleted without reading. We are specifically looking for WebTrit experience and will verify your answers. Question 1 — Prove you know the We...

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

...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. 3. Turn on the stock call‐detail r...

...and usable at all resolutions and zoom levels. • CHANGE Button: o Fix issues where previous dates are retained after a new search. • Required Fields: o If a required field is missed, notify the user and highlight the field. o Add a message at the top: “Fields with * are required.” • Checkbox Improvements: o Improve spacing and visibility for the “I have read and agree” checkbox and its red asterisk. Task 2: Contact Form • Captcha: Add an easy-to-use captcha to the contact form. • Email Sending: Prefer using a relay email account (details will be provided), but a PHP email script is also acceptable. Task 3: “GET A QUOTE” Form • Vehicle Addition: Add “Mitsubishi Eclipse Cross” as a new vehicle opti...

...System (WMS) designed for USA-based 3PL (Third-Party Logistics) warehouse operations. This is not a basic inventory app. It is a real operational WMS, inspired by platforms like ShipHero, built to manage inventory, orders, warehouse workflows, clients, and marketplace integrations. The system will be used internally and later sold as a SaaS / white-label product to 3PL warehouse owners. Why We Are Building This WMS Existing WMS solutions are: Very expensive Over-complicated Hard to customize Difficult to white-label We want a system that is: Accurate and operationally strong Simple for warehouse staff Scalable for multiple clients Sellable to 3PL businesses Ready for integrations and white-labeling Target Users of...

...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-desk SaaS as long as your approach is clear and API driven. • I’m open to whichever voice stack you prefer (Asterisk, Twilio Voice, Dialogflow CX, Kaldi, etc.) provided it works smoothly with n8n’s HTTP nodes or custom function nodes. Deliverables – Exported n8n workflow JSON with annotations – README explaining required env variables, external services, and how to add new support intents – A short demo video or live walkthrough proving FAQ...

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 by defau...

...codes so I can tag each outcome on the spot (e.g., interested, call back, not a fit) • The option to pause/resume the campaign or skip a number mid-flight If you can layer in call recording, basic analytics (call length, pickup rate), or an automated voicemail drop when a machine answers, that’s a bonus, but the non-negotiable piece is the hands-free sequential dialing. Tech-wise I’m open: Asterisk/FreePBX, Vicidial, GoAutoDial, Twilio, or any other VoIP stack you’re comfortable with—just spell out the licensing or hosting needs. I’ll happily spin up a cloud instance if you give clear specs. Hand-over deliverable 1. A functioning auto-dialer reachable through a browser (or lightweight desktop app) 2. Admin credentials, setup guide, and ...

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 real time ...

...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 outcomes, and adjust prompts or flows without touching code. • Secure API endpoint for future integrations, but nothing external is wired in for now. I’m comfortable if you build on Twilio Voice, Asterisk, or an open-source SIP stack; for the conversational layer, Dialogflow CX, Rasa, or a GPT-powered custom service are all acceptable as long as latency stays low and costs are predictable. Acceptance criteria 1. I can trigger an outbound call from a CLI or webhook, watch the agent converse, and see the transcript stored. 2. Incoming calls reach the agent, which c...

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...

We're moving away from Shopify and need a clean, reliable, minimal backend that fully replaces Shopify's core functions: - Orders - Checkout integration (payments handled separately) - Product/SKU catalogue - Inventory sync - Fulfilment push to Lynk (our warehouse) - Tracking + status updates back to our DB This is NOT a full website build. Experience Level: Intermediate Duration: 3-6 months Hours: More than 30 hrs/week

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 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 as n8n,...

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). • Walk me through a short live demo—scre...

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 conf...major migration. Expected deliverables: • Active account with international calling enabled • Step-by-step setup documentation (including screenshots or short clips if helpful) • Successful test call report confirming line quality to at least two different countries • Clear escalation path for support issues once the service is live If you already integrate with popular tools like 3CX, Asterisk, FreePBX, or similar, let me know. Please also specify any per-minute rates or bundled minute plans so I can compare to...

...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...

...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 call, but i only hear a sharp noise 4. I know this is due to mis-sampling, bit rates, and can be some other issue. I tried a lot, but I am unable to fix the noise. So, your job is to fix the noise in the call and make my AI BOT listen. This is not a job for beginners or intermediates. Only Experts needed. If you know how to fix and can do the job, then ...

...customer data across omnichannel retail, wholesale, and direct-to-consumer sales, b2b sales and supporting operations. 1.2 Scope The system will serve as a unified platform for: Product Lifecycle Management (PLM&BOM) Product, Brand & Collection Intelligence (Collection & Drop Management, Style Performance Tracking, Sustainability & Compliance (increasingly important) Inventory & Warehouse Management (WMS) Order Management (OMS) B2b login (Retail management) Demand Forecasting & Replenishment Customer & Sales Management (CRM) Finance & Accounting Integration Omnichannel & Fulfillment Intelligence (Retail, DTC, B2B, Wholesale) Analytics, Dashboards & Reporting Product Catalog Management Digital Asset Management (DAM) Dashboards (Marketing, ec...

...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 deposits using our payment-gateway API, verify the transactio...

...status milestones, and feeds confirmations, labels, and ASN data back to the 3PL WMS so warehouse teams can stay in their own screens. Essential capabilities we must ship • Real-time quotes from manufacturers (price, capacity, lead time). • Automated order management that tracks every step without manual email chains. • Out-of-the-box integration hooks (REST / webhooks, EDI if needed) so 3PLs can embed the flow during customer onboarding without custom code each time. User roles & access • Buyers see pricing, capacity, lead times, and order history. • Manufacturers manage quotes, capacity updates, and production status. • 3PL providers monitor order progress and receive shipping data directly into their WMS/TMS. Tech expectation...

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.

...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...

... ⸻ 4. Core Features (MVP Scope) 4.1 Consumables Catalog • Create and manage consumable items • SKU, supplier, cost, unit, category • Multiple suppliers per item (future-proofed) • Notes & internal tags 4.2 Inventory Tracking (Lightweight) • Current stock levels • Minimum / reorder thresholds • Usage rate (manual or calculated) • Alerts when thresholds are reached This is not a full WMS — accuracy over complexity. 4.3 Smart Reordering • Reorder suggestions based on: • Min/max levels • Average usage • Lead time • One-click reorder • Manual or automated reorder rules 4.4 Purchase Orders • Generate POs • Status tracking (draft / sent / received) • Supplier associatio...

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...