Asterisk pbx jobs

We use MicroSIP and Linkus most of our PBX Use Yeastar and Granstream. Our Dial Pattern is: if you want to call Egypt, you must remove the country code (+20******) to 70*****, and similarly for short numbers – the country code must be removed, and a '7' added automatically based on the chosen dial pattern. Also, from Egypt to Saudi, dial pattern +966 changes to 6055******. We need templates for these dial patterns with options to add or edit numbers dynamically, without displaying them directly in the dial screen. It should intelligently detect the correct country based on smart input. Additionally: - Desktop versions (Windows/macOS) - Mobile versions (Android/iOS with publishing to Google Play and Apple Store) - Backup and restore functionality included. - 6 month...

We have an active SIP Trunk service from STC Saudi Arabia and a ready 3CX PBX system. The SIP Trunk has already been activated by STC, and we have the required configuration details. We need an experienced VoIP / SIP / 3CX specialist to complete the remote setup, routing, and testing. Scope of work: 1. Configure STC Saudi Arabia SIP Trunk on 3CX PBX 2. Set up inbound call routing 3. Set up outbound call routing 4. Configure and test the available 5 channels 5. Test incoming calls 6. Test outgoing calls 7. Check SIP registration and call logs 8. Troubleshoot NAT, firewall, codec, or routing issues if required 9. Provide a short handover note with the final configuration Important: This is not a full call center setup project. The SIP Trunk service is already active. We o...

...solution. Current Setup: - A GSM/VoIP gateway (Dinstar). - No physical server at the moment. - Planning to purchase a suitable cloud-based Linux server (e.g., VPS) based on your recommendation. - AnyDesk access is available for remote configuration. Scope of Work: 1. Recommend the best cloud server provider and required specifications suitable for our needs. 2. Install and configure VICIdial and Asterisk on the cloud server. 3. Integrate the Dinstar gateway with the cloud server. 4. Provide detailed guidance on connecting the Dinstar gateway with our router/switch, including cable instructions. 5. Configure inbound and outbound calling. 6. Set up agent logins and ensure web access for all users. 7. Test the complete system to ensure full operational capability. 8. Offer basic t...

...GSM/VoIP gateway (modem). - We do not have a physical server. - We plan to purchase a cloud-based Linux server (VPS), such as from "InterServer" or any provider you recommend. - We can provide "AnyDesk" access to our desktop so you can remotely configure the setup. Scope of Work 1. Recommend the best cloud server provider and required server specifications. 2. Install and configure VICIdial and Asterisk on the cloud server. 3. Integrate the Dinstar gateway with the cloud server. 4. Guide us on how to connect the Dinstar gateway to our router/switch and where each cable should be plugged in. 5. Configure inbound and outbound calling. 6. Set up agent logins and web access. 7. Test the complete system and ensure it is working properly. 8. Provide basic training...

You will be creating a new website using the WordPress theme MaxHost, which is already installed. You can create pages from new or use existing sample data pages and change them to suit our requirements. The menu will look like the following, pages with an asterisk beside them require new content. Top Menu Home* Hosting* Domain Names* Online Presence Builder* Support Support* Help Manuals* Contact* My Account Webmail Login Control Panel Login Domain Name Management Domain Name Renewals Billing Control Panel Pay An Invoice Footer Menu Hosting* Domain Names* Online Presence Builder* Support Support* Help Manuals* Contact* My Account Webmail Login Control Panel Login Domain Name Management Domain Name Renewals Billing Control Pane Pay An Invoice Company Links Co...

I need a complete VoIP call-center setup that plugs seamlessly...recording, warm transfers and supervisor whisper/barge-in. • Straightforward agent interface—soft-phone, web-RTC or Bitrix24 dialler—so onboarding takes minutes, not days. • Clear documentation, admin training and a cost-projection that keeps total spend predictable once the system goes live. Deliverables 1. Architecture diagram and tech stack recommendation 2. Configured Bitrix24 PBX or third-party PBX connected to Bitrix24 via REST/web-hooks 3. Fully tested call flows, dashboards and AI modules 4. Agent/supervisor training session (recorded) and hand-off documentation I will provide the Bitrix24 account, branding assets and SIP trunk credentials; you bring the know-how to ma...

I need a working U.S.–based SIP trunk that plugs straight into my existing 3CX PBX, and I need it online immediately. This is for a live business system, so downtime is not an option. Here’s what I am expecting from you: • Provide a fully-provisioned SIP route (inbound and outbound) that registers successfully with 3CX and passes test calls. • Supply clear configuration details—host, authentication, codecs supported, 911 compliance, and any DID options—so I can import them quickly. • Hand over written step-by-step instructions or a short screen-share session for final cut-over. Please respond only if you can deliver right away. Include a detailed proposal outlining the solution you will supply, how you will integrate with 3CX, and y...

I need my Zoom Phone account turned into a polished business phone system. The priority is to build a virtual receptionist that can immediately answer calls, play a clear custom greeting I will supply, offer recorded information, and then transfer callers to the right extension or voi...receptionist live and tested • Custom greeting uploaded, volume-balanced, and assigned to the correct entry point • Call-routing rules in place (answer, transfer, provide recorded info) • Forwarding logic documented and activated once we choose the final schedule • A short hand-off guide or video explaining how to maintain and update the setup If you’ve built similar Zoom Phone or cloud PBX setups, let me know—screenshots or a brief description of previous w...

I need an experienced VoIP engineer to take a self-hosted PBX environment to the next level. You should feel equally at home inside VitalPBX, FusionPBX, Issabel—or be ready to recommend another open-source platform if it serves the brief better—because I want the freedom to switch or combine solutions as requirements evolve. The immediate goals are clear: • Design and fine-tune advanced call-routing logic so every inbound and outbound call reaches the right place with minimal delay. • Build granular, real-time reporting that lets me track performance by queue, agent, and campaign without relying on costly third-party services. • Bridge the PBX with my existing tools (CRM, help-desk, analytics, or any REST/Open API-friendly system) so data flows...

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

...one DID number, delivers rock-solid reliability, and offers self-service access on par with Twilio (API, online dashboard, real-time monitoring, etc.). The trunk will carry both local and international traffic, so consistent call quality and proven carrier routes are non-negotiable. My existing setup runs through a PBX system. Your job is to help me choose or broker the right carrier, secure the numbers, and complete the technical cut-over so calls flow seamlessly through the PBX without service disruption. Key expectations • 100 % uptime or SLA that truly rivals Twilio • Full documentation of trunk credentials, SIP endpoints, and API details • End-to-end testing—originating and terminating local and overseas calls, caller-ID presentation on ...

...envisioning: the bot connects via our dialer, introduces itself naturally, asks the key qualifying questions we already use, handles common objections, and—once the lead matches our criteria—books a follow-up call on the team’s calendar. All relevant call notes, recordings and contact details should feed straight into our CRM so no information is lost. You’re free to choose the back-end stack (Twilio, Asterisk, custom VoIP, etc.) provided it integrates cleanly with ElevenLabs for TTS/STT and can scale to several hundred outbound calls per day. The agent must sound convincingly human, handle interruptions, and gracefully utilize our calendly to schedule another 15 minute google meeting live to walk through our pitch deck or end the call when needed. Deliv...

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

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

My VICIdial cluster is already live, leads flow into the hopper correctly, and outbound calls are reaching the 3CX PBX through the existing SIP bridge. The sole obstacle is that the system fires only one call at a time, creating a severe bottleneck for my agents. I need the campaign and server-level dialer settings re-engineered so that the dialer can launch up to 20 simultaneous calls, keep the hopper continuously stocked, and eliminate any agent idle time. This assignment is a configuration task, not a hardware upgrade—the goal is to increase the dialing ratio and fine-tune any related rules that could limit call throughput. Key areas to address • Adjust the campaign’s dial level / adaptive or ratio settings to hit 20 concurrent dials • Verify hopper and ...

...entirely online. The critical point is that the provider must NOT require any ID or KYC verification; I simply don’t have those documents. Once live, callers placed on hold should hear music automatically. This is strictly for my own personal use, not for business or resale, so capacity can stay modest as long as quality is clear and latency low. I’m comfortable configuring a soft-phone or lightweight PBX myself if you supply the trunk details, but I’m equally happy to let you handle the setup end-to-end. Deliverables I expect: • Working SIP credentials that register from standard soft-phones (e.g., Zoiper, Linphone) without ID/KYC checks • Ability to both make and receive calls to Irish landlines and mobiles • Hold music that plays whenev...

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

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

...procurement. Scope • CCTV & Access Control: exact camera locations, fields of view, cabling paths, power points, and terminations on the security rack. • Network Infrastructure: floor-by-floor rack elevations, switch counts, patch panel layouts, cable tray routes, Wi-Fi access-point mounting points, data and voice outlets. • Remaining services (to be completed once the first two are approved): IP-PBX, public-address speakers, meeting-room A/V, and any additional data/voice points. Deliverable Package 1. Fully annotated floor plans in DWG (AutoCAD) or RVT (Revit) with layer discipline separation. 2. Rack elevation diagrams and single-line schematics in PDF and native format. 3. Device schedule (camera, AP, switch, speaker, etc.) cross-referenced to...

...waits for DTMF 1, then plays the second MP3 and ends the call. • When all numbers have been processed the bot returns a Telegram summary listing each number that was successfully called. Scope specifics you must cover • Fix the current issue: “Bot not making calls.” • Restrict outbound traffic to USA numbers only; no international dialing should be possible. • Any voice gateway (Twilio, Asterisk, FreeSWITCH, etc.) is fine so long as you wire it cleanly to the Telegram Bot API and document credentials/config in an env file. • Concurrency should be configurable so I can tune call throughput without hitting carrier limits. Deliverables 1. Fully working Telegram bot (source code) that follows the flow above. 2. Simple README co...

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

Orbis Futura is an existing research institute with a Wordpress website at orbisfutura.com. We are launching Orbis Futura Review — an editorial publication focused on defence technology and strategic futures — as a dedicated section of the existing site. The Review requ...period (monthly retainer options welcome) 6. Reference publications for the editorial experience When designing the reading experience and interaction patterns, please reference: Foreign Affairs () — article template, typography The New Atlantis () — restraint, editorial tone in layout Works in Progress () — contemporary editorial publication Asterisk () — issue-based structure Lawfare () — institute-hosted editorial publication

...stage, you place an outbound call (again with a script in hand) to secure approval and update the same card before moving on. That’s it—no selling, no troubleshooting, just clear, warm communication and tidy records. To fit in smoothly you’ll need: • A calm, professional phone manner and fluent, easily understood English • A smartphone or landline that can accept forwarded calls from our cloud PBX • Enough daily availability to answer or return calls promptly during business hours Success for me is simple: customer questions are answered, authorizations are obtained, and every Trello card shows an up-to-date note after each call. If you’re organized, friendly, and can stick to a script, this should be an easy side gig you can weave int...

...stage, you place an outbound call (again with a script in hand) to secure approval and update the same card before moving on. That’s it—no selling, no troubleshooting, just clear, warm communication and tidy records. To fit in smoothly you’ll need: • A calm, professional phone manner and fluent, easily understood English • A smartphone or landline that can accept forwarded calls from our cloud PBX • Enough daily availability to answer or return calls promptly during business hours Success for me is simple: customer questions are answered, authorizations are obtained, and every Trello card shows an up-to-date note after each call. If you’re organized, friendly, and can stick to a script, this should be an easy side gig you can weave int...

...Indian telecom regulations (TRAI/DoT). 2. Objectives • Automate bulk calling campaigns with minimal manual intervention. • Deliver pre recorded or text to speech greeting messages. • Provide a simple dashboard for campaign management, monitoring, and reporting. • Ensure scalability and reliability for large call volumes. 3. Scope of Work • System Setup: o Deploy open source dialer software (e.g., Asterisk, FreeSWITCH, VICIdial, GoAutoDial) or propose alternatives. o Integrate with SIP trunks/VoIP providers. • Features Required: o Upload and manage contact lists (CSV/Excel). o Schedule and batch calls to avoid congestion. o Play greeting messages (audio file or AI TTS). o Real time dashboard for call status (answered, busy, failed). o Reporting and an...

...stage, you place an outbound call (again with a script in hand) to secure approval and update the same card before moving on. That’s it—no selling, no troubleshooting, just clear, warm communication and tidy records. To fit in smoothly you’ll need: • A calm, professional phone manner and fluent, easily understood English • A smartphone or landline that can accept forwarded calls from our cloud PBX • Enough daily availability to answer or return calls promptly during business hours Success for me is simple: customer questions are answered, authorizations are obtained, and every Trello card shows an up-to-date note after each call. If you’re organized, friendly, and can stick to a script, this should be an easy side gig you can weave int...

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

...stage, you place an outbound call (again with a script in hand) to secure approval and update the same card before moving on. That’s it—no selling, no troubleshooting, just clear, warm communication and tidy records. To fit in smoothly you’ll need: • A calm, professional phone manner and fluent, easily understood English • A smartphone or landline that can accept forwarded calls from our cloud PBX • Enough daily availability to answer or return calls promptly during business hours Success for me is simple: customer questions are answered, authorizations are obtained, and every Trello card shows an up-to-date note after each call. If you’re organized, friendly, and can stick to a script, this should be an easy side gig you can weave int...

...stage, you place an outbound call (again with a script in hand) to secure approval and update the same card before moving on. That’s it—no selling, no troubleshooting, just clear, warm communication and tidy records. To fit in smoothly you’ll need: • A calm, professional phone manner and fluent, easily understood English • A smartphone or landline that can accept forwarded calls from our cloud PBX • Enough daily availability to answer or return calls promptly during business hours Success for me is simple: customer questions are answered, authorizations are obtained, and every Trello card shows an up-to-date note after each call. If you’re organized, friendly, and can stick to a script, this should be an easy side gig you can weave int...

I need a seasoned graphic designer to craft a four-page, high-end corporate brochure for Nanban Communications. The piece must feel polished, modern and unmistakably professional—something that immediately attracts new clients while underscoring the quality of our Security Systems, Networking Solutions, Internet Services, Home Automation, Access Control, EPBX/IP-PBX, Structured Cabling and AMC Support offerings. I will supply all copy and service details; your role is to translate that content into a clean, visually striking layout that would resonate with decision-makers at IT companies, corporate offices, educational institutions and industrial sites. Service quality is the hero message, so typography, hierarchy and imagery should consistently reinforce reliability and te...

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

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

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

...Outbound campaigns must dial through the same platform so agents work from a single interface, with those calls recorded in the same repository. Scope of work You’ll deploy, configure, and hand over a production-ready IVR. I expect guidance on menu design, prompt management, real-time monitoring and a clean dashboard where my team can adjust routing rules without touching code. I’m open to Asterisk, FreeSWITCH, Twilio, or another stack—just explain why it’s the right fit for reliability and scale. Acceptance criteria 1. Three-level menu tested end-to-end with live callers. 2. Call recordings saved in .wav or .mp3, downloadable from the dashboard. 3. Outbound dialler integrated and functional. 4. Admin user can add prompts, edit menus, and r...

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

We are looking for an experienced 3CX technician to set up and configure a small business phone system. The system is already partially configured but requires a clean setup and proper routing. Current Setup Platform: 3CX (Cloud, v20) SIP Provider: Intertelecom Existing PBX (Intertelecom) may still be active Mobile usage via 3CX app Requirements 1. SIP Trunk Configuration Proper setup of SIP trunk with Intertelecom Resolve registration conflicts (single registration issue) Ensure stable inbound & outbound calling 2. Users / Extensions Create 4 extensions/users Configure mobile apps (3CX client) Set up voicemail for each user (or central voicemail) 3. Call Routing Outbound rules (basic + prefix handling if needed) Inbound routing: Ring Group (all users) Optional DID-based routi...

I need a fully-functional auto dialer built for my company and I want it up and running fast. The core requirement is seaml...integration with the VoIP provider I’ll share once we start, including proper authentication and fail-over handling. • A clean, web-based interface where agents can log in, see their queue, and record basic call outcomes. • All source code plus clear deployment and user documentation so my in-house tech team can maintain it afterward. If you’ve built dialers before—especially with Twilio, Asterisk, FreeSWITCH, or similar stacks—let me know what framework you recommend, the timeline you can commit to, and any additional features you can add (call recording, automated messages, analytics, etc.). I’m ready to move q...

I need a full-stack voice AI that can automatically call a list of prospects I upload, speak with them in a natural-sounding voice, and switch fluidly between Hindi, English, and everyday Hinglish. The goal is to make the conversation feel as human as possible—no robotic intonation—whi...low-latency ASR supporting Hindi, English, and hybrid sentences • On-the-fly accent recognition, noise suppression, and live transcript display • Conversation logic builder so I can tweak responses myself without coding • Exportable call logs (audio, transcript, call outcome) in CSV/JSON I’m open to whichever speech engines, LLMs, or telephony APIs you prefer—Google, Azure, ElevenLabs, Asterisk, Twilio—so long as the final product delivers the sm...

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

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

...NLP / Intent Recognition Rasa, Dialogflow ES/CX, or custom transformers (e.g., BERT fine-tuned) WhatsApp Integration WhatsApp Business API (Meta) – via Twilio, WATI, or official API Voice Bot Google Dialogflow CX Phone Gateway / Amazon Lex + Polly / Asterisk AGI Backend Microservice Python (FastAPI/Flask) or Node.js Deployment AWS Lambda / EC2 / DigitalOcean (or any cloud) CRM Integration REST API calls to Zoho CRM (OAuth 2.0) Must-have experience: Built at least 2 production WhatsApp bots with negotiation logic. Integrated voice bots with Asterisk/VICIdial. Strong understanding of state machines for conversation flow. Experience with dynamic pricing or e-commerce bots preferred. We have developer working for vicidial and zoho with whom you can collaborte and wor...

My React frontend for a soft-phone style VoIP app is already designed; the missing piece is a clean, reliable bridge into FusionPBX. I need you to wire the two together so that every key action in the UI is backed by live PBX logic. Scope of the integration • Call management – place, receive, transfer and hang-up calls directly through FusionPBX. • User registration – authenticate users against FusionPBX and pull their extension data on login. • Config management – let admins tweak extension or trunk settings from the app and push changes back to FusionPBX instantly. • Real-time updates & notifications – UI must refresh on state changes (ringing, answered, missed, registration status, config saves) without manual refresh. Wha...

My React frontend for a soft-phone style VoIP app is already designed; the missing piece is a clean, reliable bridge into FusionPBX. I need you to wire the two together so that every key action in the UI is backed by live PBX logic. Scope of the integration • Call management – place, receive, transfer and hang-up calls directly through FusionPBX. • User registration – authenticate users against FusionPBX and pull their extension data on login. • Config management – let admins tweak extension or trunk settings from the app and push changes back to FusionPBX instantly. • Real-time updates & notifications – UI must refresh on state changes (ringing, answered, missed, registration status, config saves) without manual refresh. Wha...

I need an experienced adviser who feels completely at home on the phone, because the role is 100 % voice-based. You’ll be the first and last touchpoint for our consumer-goods customers, fielding inbound queries and making outbound follow-ups whenever a case needs an extra push toward resolution or a courtesy check-in. Here’s what the day-to-day looks like: you’ll log into our cloud PBX and CRM (we use Zendesk + Aircall), greet callers, capture the issue in real time, walk them through basic troubleshooting or product-use advice, and close the ticket with clear notes so the rest of the team can see exactly what happened. For outbound calls, you’ll reach out on missed contacts, delivery-status updates, or proactive satisfaction surveys. All conversations must ...

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

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

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

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

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

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