Asterisk pbx ael jobs

...the caller can respond naturally rather than keying in touch-tones. • A lightweight analytics layer that aggregates results and exposes them through a simple web dashboard plus downloadable CSV/JSON exports. I expect the stack to be Python-centric—FastAPI or Django for the service layer, a PostgreSQL or similar store for call and response data, and a telephony gateway such as Twilio, Plivo or Asterisk for call handling. If you have a preferred combination that still delivers those three core features, I’m open to it. Deliverables at the end of this phase: 1. Deployed MVP on my cloud account (AWS, GCP or Azure—your call). 2. Source code in a private Git repo with clean commit history and README-level documentation. 3. Written test plan plus unit test...

Looking for an app developer to create a business application with the following features: - SIP login - Incoming/Outgoing call functionality - Dial pad and call history - Speaker, Mute, and Hold options - SIP server integration with Asterisk/FreePBX support - Background incoming call functionality - Source code delivery - APK + Play Store-ready build

...ebook and now need a simple phone-based listening experience. Here’s what I’m after: • A caller dials a dedicated number and immediately hears the pre-recorded narration. • While listening, the caller can press keys to Play/Pause, jump to the next or previous chapter, or repeat the last section. • All audio files are ready; I just need you to build the telephony application, host it (Twilio, Asterisk, FreePBX or a comparable service is fine), and provide clear instructions so I can update audio or chapters later. Deliverables 1. Working phone number with the call flow live and tested. 2. Upload of my narration files, mapped to the correct chapters. 3. Touch-tone controls: Play/Pause, Chapter navigation, Repeat last section. 4. Brief setup ...

...will be confirmed after reviewing the existing VICIdial/Asterisk environment. B2B AI Outbound Calling Platform — Freelancer.com Project Brief Page 2 1. Scope of Work A. AI Voice Agent • Natural, professional English-speaking AI voice for outbound business calls. • Configurable sales scripts, qualification questions, and conversation flows. • Meeting booking when a prospect expresses interest. • Call outcome classification, summaries, and transcripts. • Recording support only where legally permitted and properly configured. • Handling of interruptions, silence, voicemail, poor audio, and call failures. • Immediate opt-out handling and suppression of future promotional calls. B. VICIdial Integration • Assess the installed VICIdial and ...

...store receives a steady flow of customer questions—especially between 10 AM and 8 PM CST —and I need a reliable agent who can jump straight into handling phone calls, live-chat sessions, and email tickets. The vast majority of contacts are simple order-status checks, but you should also be ready to give basic product information or begin a return when required. You will log in through our cloud PBX for voice calls and use Willdesk (or a similar ticketing tool you’re already fluent with) for chat and email. Clear, friendly English, accurate data entry, and the ability to keep response times under two minutes for chat and under one hour for email are essential. Familiarity with Shopify’s order dashboard is a plus because it speeds up status look-ups. Deliv...

...Successfully places up to four concurrent calls through our SIP trunk using the specified codecs. 2. Processes a 20,000-number text file in a single run without crashing or memory leaks. 3. Generates an output text file with accurate status for every number. 4. Pause/Resume buttons work instantly and the on-screen counters update in real time. Language, platform, and SIP libraries (e.g., PJSIP, Asterisk AMI, JAIN SIP) are up to you as long as they meet the above requirements and compile cleanly on a current Windows or Linux desktop environment....

...e-commerce store receives a steady flow of customer questions—especially between 10 AM and 8 PM—and I need a reliable agent who can jump straight into handling phone calls, live-chat sessions, and email tickets. The vast majority of contacts are simple order-status checks, but you should also be ready to give basic product information or begin a return when required. You will log in through our cloud PBX for voice calls and use Zendesk (or a similar ticketing tool you’re already fluent with) for chat and email. Clear, friendly English, accurate data entry, and the ability to keep response times under two minutes for chat and under one hour for email are essential. Familiarity with Shopify’s order dashboard is a plus because it speeds up status look-ups. ...

I need an Avaya engineer to finish setting up our IP500 V2 system remotely through TeamViewer or AnyDesk. The hardware is already racked and patched; what’s left is the software configuration inside Avaya IP Office Manager. Here’s what I expect when we are done: • External lines – all six existing analogue PSTN trunks must be programmed for both inbound and outbound calling, placed in a hunt group so that if one line is busy the next picks up automatically. • IVR / Auto Attendant – I will supply the written welcome script. You’ll record or convert it to a WAV file, upload it, and build a simple menu (Press 1 Sales, 2 Support, 3 Accounting). • User permissions (ARS) – three clear calling tiers must be enforced: internal-only, local-on...

Senior Asterisk / ARI / Python Developer Needed — Finish Existing Real-Time IVR Autodialer — 48-Hour Priority I have an existing Python/Asterisk outbound autodialer running on a Debian 12 VPS that needs to be finished and stabilized. This application is already built. I am NOT looking for somebody to rebuild it from scratch. I need an experienced developer who can inspect the existing codebase, understand how the components work together, preserve the functionality that already works, and finish the remaining live IVR integration. IMPORTANT — 48-HOUR DEADLINE / AVAILABILITY I need someone who can start immediately and dedicate substantial focused time to this project over the next 48 hours. I am willing to pay a premium for the right developer and for ...

I need an experienced field technician in the Anchorage area who can be my on-site hands at Joint Base Elmendorf–Richardson for a federal voice services contract. The work centers on enterprise-grade PBX systems—specifically Cisco Unified Communications Manager—so you should already feel at home provisioning phones, trunks, dial plans and survivable gateways in that environment. When an outage hits, you will be the first person on the ground, coordinating with our remote Network Operations and engineering teams to trace the fault, swap hardware if required, and restore service quickly. Between incidents, you will keep the plant healthy by installing and configuring new equipment, troubleshooting anything that crops up, and carrying out routine maintenance such a...

I run an Asterisk-based PBX on Linux that handles SIP/RTP traffic through a Synway GSM gateway. During live calls we hear persistent background noise that degrades voice quality. Static, echo, and other artifacts are not the main problem—the line simply has a constant hiss that makes conversations tiring and unprofessional. I need someone who knows Asterisk, Synway hardware, SIP signalling and RTP media flow inside out to: • trace the source of the background noise, whether it sits in the gateway, codec choice, gain levels, jitter-buffer settings, or network path • apply and document configuration tweaks (Asterisk .conf files, Synway parameters, codec packs, echo-cancellation, AGC, etc.) • prove the fix with test calls recorded before-...

Hi All, I am looking for an experienced FreePBX/Asterisk consultant or service provider to assist with the design, deployment, and documentation of a new FreePBX solution. Scope of work: FreePBX installation and configuration SIP trunk, extensions, IVR, queues, voicemail, and call routing setup Security hardening and best-practice implementation High-Level Design (HLD) documentation Solution architecture and network diagrams Configuration and operational documentation Knowledge transfer and handover documentation Additional requirement: I would also appreciate assistance in selecting the most suitable server hardware for the deployment. The recommendation should consider: Number of users/extensions Expected concurrent calls Storage requirements High availability and redundancy o...

I need the VAPI Voice AI cloud to talk fluently with my on-premise Yeastar P550 PBX. The goal is simple yet technical: connect my own local carrier to VAPI as a BYOC trunk, then let the AI place outbound calls—mobile and landline—using phone numbers that live in a Google Sheet. Here is how I picture the job flowing: • Configure my carrier’s SIP trunk inside VAPI so traffic routes through my existing minutes and numbering plan. • Link VAPI to the Yeastar P550, mapping extensions and caller IDs so the PBX sees every AI call as a normal trunk call. • Build or point VAPI to a script that reads the Google Sheet in real time (or at scheduled intervals) and dials each number. • Prove the setup by running test calls to both mobile and loc...

STRICT RULES - READ BEFORE BIDDING 1. NO BEGINNERS & NO EXPERIMENTS: If you are a beginner or looking to "experiment" and figure things out on the go, your bid will be REJECTED. You must have proven, hands-on experience with ALL of these specific devices: Cudy P2 5G Router DrayTek Vigor 2915 Router Yeastar S20 PBX We are a live production business, NOT a testing ground. If you have not actively worked with these exact models inside out, DO NOT BID. 2. THE CORE AUTOMATION GOAL (MANDATORY): We demand a 100% seamless, automated experience for the end-user. Moving in and out of the KSA office must require ZERO manual network adjustments. 3. STRICTLY ANYDESK OR ULTRAVIEWER ONLY: ALL work MUST be done remotely via AnyDesk or UltraViewer on my machine. No d...

STRICT RULES - READ BEFORE BIDDING 1. NO BEGINNERS & NO EXPERIMENTS: If you are a beginner or looking to "experiment" and figure things out on the go, your bid will be REJECTED. You must have proven, hands-on experience with ALL of these specific devices: Cudy P2 5G Router DrayTek Vigor 2915 Router Yeastar S20 PBX We are a live production business, NOT a testing ground. If you have not actively worked with these exact models inside out, DO NOT BID. 2. THE CORE AUTOMATION GOAL (MANDATORY): We demand a 100% seamless, automated experience for the end-user. Moving in and out of the KSA office must require ZERO manual network adjustments. 3. STRICTLY ANYDESK OR ULTRAVIEWER ONLY: ALL work MUST be done remotely via AnyDesk or UltraViewer on my machine. No d...

I’m rolling out a new Asterisk-based PBX that spans both an on-prem rack server and a small cloud instance. I need a specialist who can take the project from a clean slate to a fully functional, production-ready system. Here’s what I’m expecting: • Complete installation of the latest stable Asterisk on each node (one local, one cloud) with SIP trunks registered and ready. • Seamless call routing between sites, extension dialing, voicemail, IVR, and basic queue logic operating flawlessly across the hybrid topology. • Secure configuration—TLS/SRTP, strong authentication, sensible firewall rules, fail-over settings between the two servers. • Real-time monitoring and alerting set up (Nagios, Zabbix, Grafana, or a tool you&...

Dinstar UC2000-VE-4T + FreePBX/Asterisk + Retell AI Voice Setup Project Description We are looking for an experienced individual VoIP/Asterisk freelancer to configure a small business telephone system. Hardware available: Dinstar UC2000-VE-4T LTE/VoLTE Gateway 2 mobile SIM cards initially 4 available gateway channels Required configuration: Install and configure FreePBX/Asterisk Configure the Dinstar UC2000-VE-4T with 2 SIM cards Create two completely independent inbound/outbound call routes Configure two remote SIP softphone users working from PCs/mobile phones Each user must make and receive calls through their assigned SIM/number Incoming caller ID should be passed correctly to the remote user Configure correct outbound routing so each user always cal...

...handwritten notes compiled in a single PDF that I need turned into a clean, typed PDF. The original pages include clear headings and bullet-point lists, and I want that structure kept intact in the final document. Please make sure every heading is formatted consistently and that any bullet lists appear exactly as they do on the page. Where the notes flag an important point—usually marked with an asterisk or underline—use bold text so those items stand out immediately. Other than that, no special styling is required; a simple, readable font such as Calibri or Times New Roman is fine. Your deliverable is: • A print-ready PDF that mirrors the original structure while being fully selectable, searchable text. Accuracy is critical, so proof the copy before sending...

...must sound formal and professional rather than chatty. All I require in the script is our company name followed by a concise welcome message; no menus or next-step instructions. Once the wording is agreed, please deliver a broadcast-quality audio file (WAV or 320 kbps MP3). A clear, neutral background, well-balanced levels, and natural pacing are essential so the file can drop straight into our PBX without further editing. If you can provide two subtle tone variations, even better; I’ll choose the one that fits best. Let me know your turnaround time and any studio specs I should be aware of, and I’ll supply the exact line to record as soon as we start....

...The observed sequence is: agent 101 becomes READY → `outbound_autodial=Y` → VICIdial should select an eligible lead → an auto-call should be created/sent → `` should determine that a call needs to be placed → Asterisk should originate the call. Instead, the debug output from `` shows `READY=1 agents=0 dial_level=0 calls to place=0`, meaning the script is seeing the agent as READY but ultimately determines that there are zero agents/calls available for dialing and therefore does not originate a call. No corresponding outbound Asterisk channel is created. We have also inspected the VICIdial database and found that the AU_SOLAR campaign is active, the campaign has leads available in its list, and the campaign has previously shown a `vicidial_auto_call...

...Router Yeastar S20 PBX We are a live production business, NOT a testing ground or a network engineering academy. If you do not know these exact models inside out, do not waste our time. We will not teach you how our network works. 2. THE CORE AUTOMATION GOAL (MANDATORY): We demand a 100% seamless, automated experience for the end-user. Moving in and out of the office must require ZERO manual network adjustments. Smart VPN (Always-On): It must run 24/7. When a user walks into the KSA office and connects to the local Wi-Fi, the routing must automatically prioritize local traffic without conflicts. When outside on 5G, it must route to the remote sites smoothly. No manual ON/OFF switching. Linkus App: It must auto-switch instantly. It must use the Local PBX IP...

...must sound formal and professional rather than chatty. All I require in the script is our company name followed by a concise welcome message; no menus or next-step instructions. Once the wording is agreed, please deliver a broadcast-quality audio file (WAV or 320 kbps MP3). A clear, neutral background, well-balanced levels, and natural pacing are essential so the file can drop straight into our PBX without further editing. If you can provide two subtle tone variations, even better; I’ll choose the one that fits best. Let me know your turnaround time and any studio specs I should be aware of, and I’ll supply the exact line to record as soon as we start....

*** STRICT RULES - READ BEFORE BIDDING *** 1. DO NOT BID if you do not have direct, hands-on experience with ALL of these specific devices: - Cudy P2 5G Router - DrayTek Vigor 2915 Router - Yeastar S20 PBX I will NOT explain how these devices work or provide any tutorials. 2. ALL work MUST be done remotely via AnyDesk ONLY. No direct access, VPN credentials, or passwords will be handed over. 3. Do NOT message me asking "What is the problem?" or "What is your budget?". The entire problem, infrastructure, and scope of work are detailed precisely below. Your bid is your final price. Any messages asking these questions will result in an immediate BLOCK. --------------------------------------------------------- --- PROJECT DETAILS & SCOPE OF WORK --- We are look...

I need an experienced FreePBX professional who will teach me what I want to learn about fre...what I want to learn about freepbx operation and backup You can share your freepbx installation or use mine. You will look over my current installation if needed and help me fix the issues so that I can run it smoothly. So the focus is strictly on troubleshooting and ongoing support rather than fresh installation. I plan to use your experience instead of AI support. If you routinely work with Asterisk/FreePBX CLI, have a solid feel for SIP tracing tools such as sngrep or Wireshark, and are comfortable working/teaching on production systems where downtime needs to stay minimal, you are exactly who I’m after. I would like to use a help convenient with my work hours so that I don't ...

I'm looking for an experienced Asterisk VoIP engineer to help with PBX configuration, specifically for call routing. Key requirements include: - Expertise in Asterisk PBX - Proficiency in configuring call routing, especially geographic routing - Experience with VoIP systems and network configurations Ideal skills and experience: - Background in telecommunications or related field - Previous projects involving Asterisk and PBX systems - Strong problem-solving skills and attention to detail Please provide examples of similar work done.

...APIs or webhooks—so every interaction is logged automatically without any manual data entry. I only need straightforward rule-based flows: IVR menus for phone calls, and button- or keyword-driven replies on WhatsApp. Natural-language understanding is not required; speed, reliability, and clear routing rules are the priorities. Please propose the tech stack you’ll use (Twilio, 360dialog, Asterisk, Vonage, Dialogflow CX in “no-NLP” mode, or any other you prefer) along with the integration method for my existing CRM and ticketing tools. A brief outline of how you’ll handle testing and hand-over training will help me evaluate fit. Deliverables should include: • Fully configured call and WhatsApp workflows connected to my CRM and ticketing...

I need a working UAE local number that lets me place and receive calls from my Android phone exactly as if I were inside the country. Linkus is the reference point, yet I’m open to any SIP softphone or cloud-PBX solution that delivers equal or better results. Voice quality must be stable on both Wi-Fi and 4G, and the same local caller ID has to appear to the person I’m calling. SMS integration is essential—I must be able to send and receive texts on the very same UAE number through the chosen app or an associated API. Acceptable solutions may involve: • Provisioning a UAE DID, SIP trunk or virtual number. • Full configuration on an Android softphone (e.g., Linkus, Zoiper, Grandstream Wave, etc.). • Demonstrated test call and SMS exchange showin...

I have a brand-new Yeastar cloud instance already spun up and a Twilio account fully authorised for Australian numbers. My existing 3CX PBX is still live, but I now need everything cut across to Yeastar within the next 48 hours. Scope of work • Build the Twilio SIP trunk on Yeastar and confirm registration. • Create six extensions that mirror the current 3CX users (keep the same extension numbers). • Replicate the existing call routing rules—day/night logic, voicemail destinations, and any hunt or ring groups—so inbound calls land exactly where they do today. • Point each Twilio DID away from 3CX toward the new Yeastar trunk and verify inbound/outbound calling. • Run test calls with me on completion to confirm audio quality, caller-ID ...

I have an existing Python/Asterisk outbound autodialer running on a Debian 12 VPS. The application and dashboard are already built. I need an experienced Asterisk/ARI/AMI/Python developer to finish integrating the existing components, NOT rebuild the project from scratch. The system currently places outbound calls through a VMGate SIP/mobile gateway. Calling is working. We have also confirmed working two-way media/audio capture, AMI MixMonitor recording, DTMF transmission, and speech-to-text/transcription. The main remaining task is to connect the existing IVR state-machine () to the working live audio/transcription and DTMF system so the dialer can react to IVR prompts in real time rather than relying on fixed delays or post-call transcription. Important technical details...

...last_update_time updates, ensure agents remain properly active/READY without falsely becoming LAGGED, and verify that autodialing works normally. The solution must be a permanent code/configuration fix, not a workaround such as manually updating MySQL timestamps, disabling LAGGED detection, or masking the symptom. The freelancer should be experienced with VICIdial, PHP, MySQL/MariaDB, JavaScript/AJAX, Asterisk and VICIdial background Perl processes and should document exactly what was changed and why....

I want to wire Vapi’s Voice AI directly into my Yeastar P550 PBX so every outbound call to potential customers is handled automatically. The goal is true hands-free cold calling: the system should pick numbers from my lead list, place the call through the P550, open with an AI-generated script, hold a natural conversation, and either book the meeting or pass the call to a live agent when certain triggers are met. You’ll need to bridge Vapi’s API with the P550 via SIP or an equivalent method, build the call flow inside Yeastar (dial plans, queues, or IVR branches as required), and make sure the audio streams cleanly in both directions with minimal latency. I already have both Vapi and the PBX accounts active; what I don’t have is the middleware that le...

...trunking solution running on Asterisk under Linux. At the moment I’m unsure whether the server already in place can be salvaged or if it will be cleaner to start from scratch, so the first step will be a quick audit of what’s there. From that point we can move straight into installing / re-installing Asterisk, pointing it at my chosen SIP trunk provider, and fine-tuning the dial-plan so inbound and outbound calls flow smoothly. Security hardening (fail2ban, TLS / SRTP where supported), codec optimisation for solid call quality, and basic call-flow features such as voicemail, IVR and call recording should all be factored in. I’ll need everything documented well enough that I can tweak extensions or add trunks later without starting over. Acceptance crite...

...what I still need is a solid Asterisk-based voice layer that simply works every time a user places or receives a call. My priority is configuration and integration: connecting SIP trunks, wiring up DID numbers, mapping inbound and outbound routes, and ensuring that every two-way call is recorded and pushed securely to my chosen cloud storage bucket. The core stack will run on a Linux server with Asterisk at its heart. You’ll handle the dial-plan logic, trunk registration, codec selection, security hardening, and the APIs that let the mobile app trigger voice calls and fetch recording links. I already know the basic architecture I want, so we can move quickly into hands-on setup, iterative testing, and final roll-out. Deliverables: • Fully configured Aster...

...Debian or Ubuntu—explain why, and then handle the entire build. I do have a few custom requirements inside FreePBX (not a plain-vanilla install), so once the basic stack is running I will share the detailed dial-plan tweaks, trunk settings and any extra modules I need. Key deliverables: • Provision and harden the VPS with your recommended Linux distro • Install the latest stable FreePBX and Asterisk • Configure trunks, routes and SIP credentials for so calls in/out work straight away • Apply my specific FreePBX settings once I provide them • Supply root/GUI credentials plus a brief hand-off document outlining what you did and how to update the system in future Please factor in standard security best practices (firewall, fail2ban, S...

We are looking for a freelancer with hands-on experience in Asterisk, telephony, and SIP setup to support us on a short-term project.

I need a Telegram OTP bot integrated with Asterisk that includes a Caller ID feature. Key Requirements: - The bot should send and verify OTPs, as well as log OTP requests. - The Caller ID feature should display incoming caller numbers on just one device. Ideal Skills and Experience: - Experience with Asterisk and Telegram bot development - Familiarity with OTP generation and verification - Knowledge in implementing Caller ID functionalities

I need an expert who knows the Panasonic KX-NS100 inside out to take over ongoing maintenance and, more urgently, to hunt down stubborn routing problems. At the moment we are plagued by ghost calls, fake trunk activity and sporadic audio drop-outs that confuse users and clog the logs. These symptoms point to deeper call-routing faults, so I’m looking for someone who can dive straight into ARS tables, COS settings, SIP trunk parameters and firmware quirks to pin down the real cause, clean it up and stop the phantom traffic for good. You’ll be given remote access via VPN or TeamViewer during our business hours so you can trace packets, watch live call traffic and adjust the configuration in real time. Once the issues are eliminated I’d like concise documentation of every...

Require is a custom Asterisk-based autodialer engineered for high-volume, unregulated outbound traffic. The system must run on a hardened Linux distribution (e.g., Debian or CentOS) with Asterisk 18 or 20 compiled with specific modules: chansip for SIP trunking, appdial for call routing, and app_playback for IVR management. The core architecture includes a predictive dialing engine built in Python or C++ that interacts with Asterisk via AGI (Asterisk Gateway Interface). It must integrate the sofia or chansip channel driver to handle SIP trunks, supporting multiple concurrent legs for rapid call placement. For DTMF handling, the system will utilize the appdial with D option to detect dual-tone multi-frequency signals, routing them to specific IVR branches define...

...MVP) Metodología: Desarrollo Ágil / Entrega basada en hitos (Milestones) 1. Visión General del Proyecto y Arquitectura Objetivo Buscamos construir una aplicación descentralizada (dApp) en arquitectura Web2.5 que requiere un framework de comunicación de voz peer-to-peer (P2P) en tiempo real basado 100% en internet. El sistema debe omitir completamente el modelo de telefonía tradicional (PSTN/SIP/PBX), reemplazando los números telefónicos convencionales por direcciones de billetera cripto / Identificadores Descentralizados (DIDs). El acceso a los canales de audio debe estar estrictamente restringido por tokens / roles (token-gated) mediante el estado de contratos inteligentes o firmas en el backend Web2.5, garantizando der...

M1: Outbound 407 diagnosis + VM fixes Full SIP trace of the failing outbound call, root-cause diagnosis of the 407, written trace-backed ticket for TeleCMI stating the carrier-side change required, and four Asterisk/PJSIP config bugs fixed on silent-sip-edge2. M2: Outbound live + AI pipeline handover Stable outbound SIP and RTP end to end, Silent Voice Kernel connected to the live call path, single pipeline with VOICE_MODE=FREE (Sarvam Bulbul v2) and VOICE_MODE=PREMIUM (ElevenLabs), test calls in both modes, handover notes.

Customer Portal & Admin Portal Requirements Project Overview Develop a modern, secure, and scalable multi-tenant Customer Portal and Admin Portal for managing Business Phone System (PBX/UCaaS) services. The platform should provide customers with a self-service experience while giving administrators complete control over provisioning, billing, support, and system management. Customer Portal The Customer Portal should allow customers to: Manage Business Phone System (PBX/UCaaS) services. View and manage extensions, users, departments, ring groups, queues, IVRs, voicemail, call routing, and other PBX features. Manage Direct Inward Dial (DID) numbers. Configure call forwarding, caller ID, business hours, time conditions, and other telephony features. Auto-provision...

...media company. We connect advertisers with retail media across multiple retailers through one point of access, and we give retailers the tools to sell and manage their own media offering. Tagline: "Retail media, made easy." Our audience is consumer brands, media agencies and retailers in the Nordics, with wider European ambitions. The symbol is already decided We have our logomark: the fine-line asterisk shown in the attached file. It is not up for redesign, and you do not need to propose a new one. Your job is to build the rest of the identity around it. You may adjust its scale, line weight and optical alignment so it sits correctly with the type, but the form itself stays as it is. Important: the name The full company name today is Retail Media Networks, but the weig...

...brands feel seamless together. Scope of work • One square Instagram post (1080 × 1080) • One Facebook feed post (1200 × 1500 or similar recommended size) • One email banner (1200 × 500) • Website hero banners for both desktop (3200× 1200) and mobile (1200 × 1600) Key messages – “Afterpay Day” – “15 % off* site-wide” – Clear call-to-action such as “Shop Now” – Include the small asterisk on “off*” for T&Cs Resources Brand site: Afterpay toolkit: Please keep copy minimal and let the bold type and colour speak. Final files should be supplied as layered PSD or AI plus web-ready PNG/JPG. Acceptance criteria 1. All dimensions correct ...

I need a PBX deployment for a small office with seven new PoE IP phones that still require initial provisioning. We will handle the wiring part. You handle the software remotely. Scope of work • Install and configure the PBX on the platform. • Register all seven IP phones, apply the correct extensions, and confirm they can dial in, dial out, transfer, and park calls. • Enable the advanced features we know we’ll use right away: voicemail, call-routing rules (day/night, hunt groups, etc.), and conference calling. • Upload or record our custom audio prompts (welcome greeting, on-hold music). • Run end-to-end testing, then document the final settings so we can manage minor changes ourselves. Acceptance will be based on: – Every ha...

...including concurrent calls, proper carrier failover, and responsive web interfaces. Essential features • Call recording enabled on every leg of the call, saved with logical file names and dates. • Automated call distribution so incoming queues and outbound callbacks are routed evenly across agents, with simple skill-based routing rules. Your work will include installing or hardening ViciBox/Asterisk, configuring Twilio Elastic SIP Trunking, setting up the dial plan, test campaigns, agent log-ins, and showing me where to adjust lists, dispositions, and recordings. A quick walk-through over Zoom at the end will confirm everything is operational from soft-phones and, if possible, a physical SIP handset. Deliverable is a fully functional ViciDial dashboard that dials...

I am moving a small team (fewer than ten extensions) from our current cloud-based PBX to 3CX and want the changeover to be seamless. The help I need centres on two areas: • Data transfer – bringing across call records, phonebooks, voicemails and any other assets that 3CX can import. • Configuration and setup – installing the latest stable 3CX version on our chosen host, mapping SIP trunks, creating extensions, setting IVRs, ring groups and basic call-flow logic so that everything works exactly as it does today (or better). If you have ideas for user training, feel free to mention them, but it is not part of the core scope. Acceptance criteria • All existing numbers route correctly through 3CX. • Voicemail, call history and contacts are presen...

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all det...pay clients for their 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" includin...

...their side. I suspect the PJSIP trunk settings are to blame, though the outbound route or some subtle SIP-signaling mismatch could also be involved. No recent changes have been made to the trunk configuration, so whatever is wrong has likely been there from the start. What I need from you • Trace and isolate the fault across the outbound route, PJSIP trunk, and SIP signaling (sngrep or tcpdump, Asterisk CLI, etc.). • Apply the required corrections directly in FreePBX and confirm calls reach Twilio and complete successfully. • Provide a concise summary of the adjustments so I can reproduce them on other servers. Acceptance criteria 1. A test call to a public number completes with two-way audio. 2. The call appears in Twilio’s live call log. 3. No “All...

...Introduce or provision a reputable provider that can activate an account in my company’s name with all necessary KYC completed. • Ensure the trunk is enabled for two-way voice, CLI presentation, 911 compliance, and international reach. • Supply full connection details (SIP URI, username/password or IP auth, codec list) plus concise step-by-step setup instructions for popular soft-switches or an Asterisk/FreePBX environment. • Confirm channel limits and any burst options so I can scale call concurrency when required. • Walk me through a quick test call—both inbound and outbound—to prove audio quality and stability. Once everything works as promised, the engagement is complete. If you already have direct carrier agreements in place a...

...cleanly—Apache, MariaDB/MySQL, Asterisk and all VICIdial services report as running and phpMyAdmin shows the database online. Yet every agent now sees “Invalid Username/Password” on the campaign screen and the admin page just reloads when credentials are entered. Here is what I have already done: examined the Apache, MySQL, Asterisk and VICIdial logs for obvious errors, repaired and optimized the vicidial_users table, and confirmed the credentials being typed are correct. No browser-side tricks helped. We also do not have a usable database backup, so whatever fix you apply must leave the current data intact—no wipe-and-reinstall options are acceptable. I need you to: • Identify the exact point in the stack (MySQL privilege tables, HT...