Asterisk pbx työt

...configured to issue USA DID numbers and deliver ready-to-use SIP credentials for a blended (inbound + outbound) call-center environment. The system has to sustain 51–200 concurrent calls with consistent audio quality and tight security. Call recording, an IVR designer, and detailed call analytics/reporting must be native features, not bolt-ons. I am happy to work with an open-source stack such as Asterisk or FreeSWITCH, a commercial carrier API, or a hybrid approach—whichever lets us scale quickly while keeping ongoing costs predictable. TLS/SRTP support, automatic fail-over to a secondary trunk, and the ability to order, port, and manage DIDs on demand are essential. Deliverables • Deployed, fully tested SIP trunk platform with at least one live USA DID reac...

... 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...

...Flowroute and need a seasoned SIP specialist who has already worked hands-on with that platform—or with comparable carriers such as Flowroute, Plivo, 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 mo...

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 ...

...delivers stable, high-quality audio that plays easily on web and mobile browsers. Specific deliverables • Cloud-hosted streaming server fully configured, tested, and secured • Web player (HTML5) or embeddable widget that matches simple ministry branding • Broadcast scheduler set for two daily live slots with automated fallback music/sermons if we go offline • Call-in system integrated (SIP, PBX, or dedicated service) with host controls for screening, volume, and recording • Documentation and a brief hand-off session so I can add presenters, update schedules, and run the desk myself Acceptance criteria • Listeners reach the stream via a single URL and hear 128 kbps stereo without buffering for at least 30 minutes under load test •...

My Panasonic Ns500 PBX sits on but cannot “see” the rest of my network. Everything else flows through a FortiGate 60F firewall, a FortiSwitch 424E-Fiber core, and a FortiSwitch 124F-FPOE at the edge. I need someone to shape the network so this Panasonic box can handle VoIP communication smoothly. What I already know • The PBX will run pure SIP. • Dedicated VoIP rules on the FortiGate are required; simple, generic access is not enough. What I need from you • Review the current FortiGate policy set, VLAN layout, and switch port profiles. • Create or adjust firewall rules, NAT, and any SIP ALG or helper settings so that SIP registration, signalling, and RTP streams pass without one-way audio or dropped calls. • Tag or untag the a...

...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...

Our FreePBX (Astersik) System is rejecting inbound calls and unable to make outbound calls. System uses Laravel UI with PBX system. Telnyx says that problem is with our system. So who is expert to fix it in few clicks? Thank you

...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...

...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...

I’m finalising the concept for a new four-star hotel in Dubai and need a complete ELV design package that is ready for authority submission, tender and construction. The scope covers every low-current system the property will rely on: CCTV, access control, public address, structured cabling, LAN/Wi-Fi, IPTV, lighting control, IP PBX/phones, wireless intercom, guard-tour, vehicle and pedestrian barriers, UPS back-up, and all audio-visual LED display points. CCTV coverage must extend to the reception and lobby, all corridors and other common areas, parking entrances and exterior approaches, as well as the guest rooms themselves. All other subsystems have to share the same structured cabling backbone and integrate seamlessly with the hotel PMS, BMS, fire-alarm interface and any ...

...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 ...

My office phone system is moving from a Grandstream UCM6104 to a UCM6304 and I want the transition to be seamless for every user. The task revolves around configuration and customization of the IP PBX itself—specifically, migrating the complete settings set-up from the 6104 to the 6304. Both appliances are on-site and reachable through the same network. I can provide SSH/HTTP(S) access, the latest firmware files, and a recent backup from the 6104.

I need a production-ready softphone for both iOS and Android built on WebRTC and standard SIP. The app will authenticate users with a simple username-and-password flow against our existing PBX or have an onboarding process for new customer, then expose a clean, corporate-style interface that matches the rest of our product line. Core scope • Implement voice calling with transfer, local audio mixing for two-party and ad-hoc conference/merge, BLF, hold/resume and DTMF. • Add visual voicemail with message playback, delete and download. • Enable two-way SMS inside a conversation view. • Web browser view to show our webpage • Contact lists (local & hosted) • Recent call history Technical notes – WebRTC should handle media; SIP (UDP and TLS)...

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 ...

...VPS with full administrative access. The task is to integrate this VPS with the Phone Module in Odoo version 19 so that the server is used as the VoIP/PBX backend for handling real phone calls. The communication flow will be from Egypt to the Kingdom of Saudi Arabia (KSA). VPS Details Server Type: VPS Operating System: Ubuntu 24.04 LTS Access Level: Root access available Approximate Resources: CPU: 2–4 Cores RAM: 4–6 GB Storage: 50 GB SSD Public IP Address: Available Firewall Management: Available Scope of Work Use the existing VPS as the VoIP/PBX backend Integrate the VPS with the Phone Module in Odoo v19 Configure a VoIP/PBX service compatible with Odoo Set up a VoIP/SIP Trunk that supports international calls Enable and configure call...

...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...

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 ...

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...

...Multi-Tenant PBX Deployment (FusionPBX / FreeSWITCH Preferred) Project Description We are an IT Managed Service Provider (MSP) looking to build a robust, scalable, white-label VoIP platform to host phone systems for multiple distinct clients. We are looking for a senior VoIP engineer to deploy, configure, and secure a True Multi-Tenant PBX System. Important Architectural Requirement: We are NOT interested in a single-instance FreePBX installation hacked with custom contexts. We require a system designed for multi-tenancy from the ground up to ensure strict data isolation and security between clients. FusionPBX (FreeSWITCH) is our preferred platform, though we are open to VitalPBX (Carrier Edition) or Kazoo. Key Deliverables * Multi-Tenant Architecture: * Setup of the c...

...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 ...

...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...

...(QA) review of the BG Online School Management Software communication system. This role is solely for testing and validation – no implementation or reconfiguration. **Role Definition** - Independent QA/verification only - No redesign or configuration changes without approval - Temporary access for testing purposes only - System configuration is frozen during the review **System Overview** - **PBX:** 3CX - **VoIP Provider:** Callcentric - **Inbound Numbers:** - Toll-Free: 1-877-359-8098 - Local: 1-914-662-0749 - **Use Case:** Educational SaaS communication **Scope of Review** Verify the following without modification: - Callcentric Configuration (DID routing, inbound logs) - 3CX SIP Trunk (registration, inbound SIP logs) - Inbound Call Flow (business hours vs after-hours...

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

We have set up an Asterisk server integrated with OpenAI, and while it functions reasonably well, we are not satisfied with the voice quality. At the moment, it sounds more like an old 1980s cassette recording than a modern 2026-quality voice. We are therefore looking for a partner who can help improve the voice quality. We have already tested several solutions currently available on the market. Before selecting a partner, it is a strict requirement for us that you can provide access to a live server setup that we can test by calling a real phone number. We are not interested in solutions that cannot be demonstrated through a previously deployed and functioning server. Please only submit an offer if you can meet this requirement.

...system. This role is solely for testing and validation – no implementation or reconfiguration. The freelancer must sign a QA NDA before access is granted. **Role Definition** - Independent QA/verification only - No redesign or configuration changes without approval - Temporary access for testing purposes only - System configuration is frozen during the review **System Overview** - **PBX:** 3CX - **VoIP Provider:** Callcentric - **Inbound Numbers:** - Toll-Free: 1-877-359-8098 - Local: 1-914-662-0749 - **Use Case:** Educational SaaS communication **Scope of Review** Verify the following without modification: - Callcentric Configuration (DID routing, inbound logs) - 3CX SIP Trunk (registration, inbound SIP logs) - Inbound Call Flow (busi...

...every change updated in Hostaway without delay. We operate across key locations including Yas Island, Saadiyat, Reem, and Al Raha, so accuracy and speed are critical. This role focuses on two core areas: • Guest relations & customer service – clear, friendly, professional English communication via phone, email, and live chat, following our established tone of voice. Calls come through our cloud PBX, messages are handled in Help Scout, and live chat is managed via Intercom. You will be dealing with tourists, corporate guests, and families, so professionalism and patience are non-negotiable. • Bookings & reservations management – creating, modifying, and cancelling reservations directly in Hostaway, then double-checking that all dates, rates, and gu...

...every change updated in Hostaway without delay. We operate across key locations including Yas Island, Saadiyat, Reem, and Al Raha, so accuracy and speed are critical. This role focuses on two core areas: • Guest relations & customer service – clear, friendly, professional English communication via phone, email, and live chat, following our established tone of voice. Calls come through our cloud PBX, messages are handled in Help Scout, and live chat is managed via Intercom. You will be dealing with tourists, corporate guests, and families, so professionalism and patience are non-negotiable. • Bookings & reservations management – creating, modifying, and cancelling reservations directly in Hostaway, then double-checking that all dates, rates, and gu...

...POST gateway the provider offers. The core work is to add or adjust the AGI / server-level scripting inside Vicidial (Asterisk) so the DTMF event triggers the API call, logs the result, and keeps normal call flow intact. I’m open to advice on whether the SMS text should stay static or be templated dynamically—feel free to suggest a clean way to handle both options. Acceptance will be based on: 1. Call transfer happens with no audible delay to the caller. 2. SMS lands on my test handset within three seconds of DTMF “1”. 3. Delivery status is written back to the Vicidial database or a simple log file for later audits. If you’ve done Vicidial AGI work or Asterisk dial-plan integrations before, this should be straightforward. Please...

...inside our existing call-center system. Here’s what I’m after: • A lightweight SDK or API that can be embedded in our iOS and Android builds, listens for the customer’s voice, runs accurate speech-to-text, and hands the text off to our backend. • A routing mechanism that takes that transcript, checks caller intent or keyword triggers, and opens the correct queue/extension in our on-premise PBX (we use SIP). • A simple dashboard so supervisors can see transcripts, routing decisions, and call metrics in real time. • Clear documentation and a short demo app that proves the flow end-to-end. Acceptance criteria 1. Speech recognition latency under two seconds on a standard 4G connection. 2. Accuracy ≥ 95 % on everyday support vocab...

This banner will sit on a pull-up stand at events and retail spaces, working purely for brand awareness. The design should grab attention from a distance, introduce my logo and tagline clearly, and convey the personality of the brand in a clean, modern way—no heavy copy, just strong visuals and hierarchy. I have a rough idea of colours ...printer without extra prep. My ideal timeline is to approve final art within one week of awarding the project, with reasonable revisions along the way. Banner Trxt: STOP OVERPAYING FOR BACKDROPS. GET FAIR PRICES FROM A WOMAN-OWNED BRAND. Fair Pricing. No Tariffs. Woman-Owned. Community-Driven. We’re here to support your photo booth business, not just take your money. VISIT US FOR EXCLUSIVE PBX DEALS & COMMUNITY SUPPORT ⬇️

...off once: 1. A Teams call is captured end-to-end in both automatic and manual modes. 2. The resulting file opens only after the included decryption routine is applied. 3. A legacy pabx system call is captured end-to-end in both automatic and manual modes 4. Recording pause, resume and their related reason codes If you’ve handled audio capture for Teams, VoIP stacks or integrated with older PBX systems before, that experience will be a big plus....

I am looking for an experienced Frappe Framework / ERPNext developer to build a PBX integration module that connects Yeastar Cloud PBX with ERPNext. The goal is to replicate the functionality of a typical call‑center integration (similar to UpeoTelephony for FreePBX), but using Yeastar’s official APIs and webhook system. The module should include: • Real‑time webhook reception from Yeastar • Full call logging inside ERPNext • Click‑to‑call functionality • Incoming call popups • Automatic linking of calls to CRM records • Basic KPI dashboards and reporting Required Features 1. Yeastar → ERPNext Webhook Integration Create an ERPNext endpoint to receive real‑time events such as: • Incoming call • Outgoing call • Ans...

I have a fresh Google Cloud Platform VM ready and need a complete ViciDial installation and configuration focused strictly on inbound call handling. No CRM or help-desk integration is required; this will run as a standalone system. Here’s what I expect from you: • Clean install of the latest stable ViciDial/Asterisk stack on my GCP instance. • Secure network and SIP settings, including firewall rules and SSL where appropriate. • Create and test one inbound campaign, queue, DID, and agent login to prove calls flow end-to-end. • Provide concise notes or a screen-share walkthrough showing how I can add agents, numbers, and recordings in the future. I’m more comfortable communicating in Telugu, so a Telugu-speaking engineer would be ideal, t...

I already have a Linux server online and reachable; what I need now is a clean installation and full configuration of Asterisk, FreePBX, and A2Billing so I can run a reliable business phone system. Your scope covers: • Installing the latest stable releases of Asterisk, FreePBX, and A2Billing on my existing server • Bringing the stack to a “ready-to-use” state—SIP extensions, VoIP trunks, IVR, call recording, voicemail, and billing profiles must all function out of the box • Hardening the server (firewall rules, Fail2Ban, strong passwords/keys, SSL where applicable) • Running test calls to confirm inbound and outbound traffic, rate decks, CDR logging, and balance deductions in A2Billing • Handing over a concise post-deployme...

...and permissions. • Build out role-based dashboards, KPIs and reports that give sales, support and management the insights they need at a glance. • Design custom workflows and automation covering lead-nurture sequences, pipeline stage movements, ticket escalation and any approval loops that reduce manual touchpoints. • Integrate all daily-use channels: IMAP/SMTP email, WhatsApp, telephony (Asterisk/3CX), website lead forms and our analytics stack (GA4, Hotjar, custom SQL). • Migrate legacy data (contacts, deals, historic tickets and custom fields) from spreadsheets and our current SaaS CRM, cleaning duplicates on the way in. • Deliver short, role-specific training sessions plus concise SOP documents to drive internal adoption. • Provide tw...

Incoming calls that arrive through our SIP trunk suffer classic “one-way audio”: the caller hears us, yet we get silence. Outbound calls and all internal extension traffic are perfectly clear, so the problem is isolated to inbound SIP traffic only. I need someone comfortable with Asterisk, FreePBX, RTP/NAT behaviour, and typical firewall or trunk mis-configurations to jump in, trace what is happening, and implement a permanent remedy without interrupting live service. Remote SSH or VPN access to the PBX and edge firewall can be provided immediately, along with console access to the SIP provider’s portal for any trunk changes. Acceptance means: • Two-way audio is restored on every new incoming call test. • No regression on outbound or extension-...

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File:...incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills: PHP, Software Architecture, Asterisk PBX, MySQL, JavaScript See more: making money with international premium rate numbers, iprn telecom, iprn providers, international premium rate numbers providers, international...

I need an experienced Asterisk programmer to setup and maintain a custom IVR for an Automated Third-Party Verification Service. Key tasks include: - Implementing a detailed IVR menu flow that I will provide - Troubleshooting existing issues - Regular updates and maintenance - Adding new features as needed Ideal skills and experience: - Strong proficiency in Asterisk - Experience with IVR systems - Problem-solving skills - Ability to interpret and implement detailed plans

...Setup and testing of Inbound and Outbound routing with a sample SIP provider. 2. Interface & Advanced Features: Crossbar API (Crucial): Full configuration of the Kazoo REST API (Crossbar). It must be accessible externally via HTTPS. You must verify that I can authenticate and create users via API (Postman/Curl). Monster UI: Installation and configuration of the Monster UI (Admin & User Portals). PBX Features: Enable and test Voicemail, IVR, Call Forwarding, and Call Recording. WebRTC: Configuration of Secure WebSocket (WSS) and TLS for browser-based calling support. 3. Mobile Application (Linphone White-Label): Customization: Compilation of the open-source Linphone app (iOS & Android) featuring my custom branding (Logo, Colors, App Name) and pre-provisioned SIP...

...Setup and testing of Inbound and Outbound routing with a sample SIP provider. 2. Interface & Advanced Features: Crossbar API (Crucial): Full configuration of the Kazoo REST API (Crossbar). It must be accessible externally via HTTPS. You must verify that I can authenticate and create users via API (Postman/Curl). Monster UI: Installation and configuration of the Monster UI (Admin & User Portals). PBX Features: Enable and test Voicemail, IVR, Call Forwarding, and Call Recording. WebRTC: Configuration of Secure WebSocket (WSS) and TLS for browser-based calling support. 3. Mobile Application (Linphone White-Label): Customization: Compilation of the open-source Linphone app (iOS & Android) featuring my custom branding (Logo, Colors, App Name) and pre-provisioned SIP...

...booking form that lets clients choose a date, leave project details and instantly receive a confirmation email, • an AI-powered chatbot that can guide prospects, answer common questions and, where possible, schedule quotes automatically, • clean on-page SEO for our local concrete keywords so we rank in maps and organic results. If you can connect the chatbot to a voice gateway (Twilio, asterisk, etc.) so it can pick up calls and handle simple quote bookings, even better—speed and time-saving is my goal. Deliverables I’d like to see: 1. Logo package (SVG, PNG, PDF, font info, colour codes). 2. Responsive website with my branding applied, fully tested across modern browsers. 3. Booking form and chatbot integration working end-to-end, includi...

We are looking for a freelance talent acquisition specialist / technical recruiter to help us identify, screen, and introduce qualified PBX / VoIP engineers, preferably with Asterisk experience, who are available to work in Oman. This is not an engineering role. Your task is to find the right engineer, verify technical capability, and connect them with us. Responsibilities Search for and source PBX / VoIP engineers (Asterisk preferred) Target candidates willing to relocate to Oman Screen candidates for: Hands-on Asterisk / VoIP experience SIP, trunks, gateways, call routing, codecs, NAT, firewalls Real deployment experience (not just theoretical knowledge) Conduct initial interviews and technical vetting Share shortlisted candidates with: CV / profile Ava...

I need a freelancer to install and configure a Vicidial/Asterisk dialer on my cloud server running Ubuntu. The tasks include configuring SIP trunks, setting up basic campaigns and agents, and securing the server with firewall and Fail2Ban. The project must be completed within the same day (post-lunch) in Hyderabad timezone.