Asterisk pbx jobs

I need a complete on-hold production for our PBX: a warm, friendly female voice delivering the three short scripts below, separated by ambient background music beds of roughly thirty seconds each. The finished mix should run somewhere between one and two minutes in total and be supplied as a single 16-bit, 8 kHz WAV file ready to load straight into the system. Script to voice: “Thank you for calling Andrew’s Tyre and Mechanical North Lakes. All of our operators are busy assisting other customers and we will be with you shortly.” —30 s ambient music— “Did you know Andrew’s Tyre and Mechanical has been locally owned and operated for the past 30 years? Now that’s service!” —30 s ambient music— “Andrew’s Tyre...

I already have a Contabo server standing by and simply need a clean, production-ready ViciDial stack on it. The job covers: • Installing the latest stable ViciDial (with Asterisk and its dependencies) from scratch. • Optimising the underlying Linux distro you feel is most reliable for call-centre workloads. • Hardening the box with a well-tested firewall configuration—CSF, UFW, iptables, or a similar solution is fine—as long as only the ports Vicidial, SSH and web administration actually require remain open. • Verifying that the web interface, database, and telephony services all start automatically after a reboot and that calls can be placed through a demo campaign. • Supplying a concise hand-off sheet: all commands run, credentials cre...

My organization is looking for a telephony tech in the Saskatoon, SK to connect a Grandstream ATA to an Avaya IP Office system. The primary objective is to move two existing fax numbers—306-934-5787 and 306-955-3059—onto the ATA so they send and receive reliably through the PBX. The job covers: • Physically or remotely provisioning the Grandstream ATA, assigning it an internal extension, and ensuring it communicates correctly with Avaya IP Office. • Mapping both fax numbers to the ATA ports and confirming successful inbound and outbound fax transmission. • Providing a brief record of any IP Office or Grandstream settings you change so I can reference them later. Acceptance is complete when both fax lines pass test sends and receives without errors.

...interaction feels human. • Dynamic query-based routing – once the caller’s need is clear, the AI should transfer the call to the appropriate extension or external number automatically. • Clean hand-off – when the call is routed, the receiving party must get a short, accurate summary of the caller’s request so they can pick up seamlessly. I’m happy to integrate with existing VoIP platforms (Twilio, Asterisk, FreePBX, 3CX or similar) if that speeds development, but I’m also open to a custom SIP-compatible solution. Cloud-hosted, on-prem, or hybrid deployment can be discussed; reliability, low latency, and call quality are non-negotiable. For deliverables, I’ll need: 1. A working prototype handling live calls. 2. A simple...

I run a growing small business that relies on Microsoft 365 and a cloud-based VoIP phone system. I’m looking for a dependable partner who can step in as our day-to-day IT resource, keeping both environments running smoothly whi...keep us aligned. Deliverables • Same-day response to support tickets during business hours • Resolution of Microsoft 365 and VoIP incidents or escalations • User onboarding/offboarding completed within agreed timeframes • Monthly health report outlining work performed, open issues, and improvement ideas A solid grasp of Azure AD/Entra ID, Exchange Online, Teams admin, and common hosted PBX platforms will help you hit the ground running. If this sounds like your wheelhouse, let’s talk about how we can keep my tech ...

I have a WebRTC soft-phone built with JsSIP that needs to register to an Asterisk 18 server over WSS. SIP credentials are confirmed correct, yet the browser console shows an authentication failure. The signalling path is protected with TLS certificates, so the problem is somewhere in the certificate handling or the way Asterisk presents the challenge. Your job is to trace and eliminate the registration failure, then hand back a clean configuration and proof that the client can successfully register and place a test call. Environment details you will touch: – Asterisk 18 (pjsip stack enabled) – JsSIP running in a standard browser (wss://) – TLS with server and client certificates already issued Acceptance criteria: • JsSIP completes REGIS...

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

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

VoIP PBX, SIP trunk, Extensions setup with US, UK numbers

...that the third-party handoff is seamless. Required Qualifications: Proven experience with Alcatel OmniPCX Enterprise (OXE) or OmniPCX Office (OXO). Proficiency in using OmniVista 8770 or mgr/mtcl command-line interface. Deep understanding of Entities, Time Ranges, and Modification Tables within the Alcatel environment. Strong track record of handling root-level Linux/Unix permissions within PBX environments....

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

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

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

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

I am looking for an experienced VoIP/PBX specialist to set up a small FreePBX-based phone system. This is a fixed-price project, and I am requesting your best offer. The system should be hosted on a reliable and cost-efficient VPS provider of your recommendation. Hosting costs and Twilio costs are not included in your project fee and will be paid separately by me. You will guide me through the account setup where necessary. Project Requirements: The PBX system should include two to three dedicated inbound phone numbers (DIDs), connected via Twilio or a comparable cost-efficient SIP provider with good call quality. Proper SIP trunk configuration and basic VoIP security (firewall configuration, fail2ban, protection against toll fraud) are required. For the first phone numbe...

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

Urgent need for an experienced Cybersecurity specialist for a confidential, short-term private project (1-3 months) in my startup. Key skills: VoIP setup and security (PBX, SIP, encryption, threat protection), VPN configuration and testing, spoofing techniques (for ethical testing/research), spoofed numbers handling/detection. Expertise in data collection from various sources (ethical/OSINT methods), advanced virus/malware detection, analysis, and simulation/creation for defensive/red team purposes (ethical security testing only – antivirus evasion, threat emulation). Experience: 2-4 years in cybersecurity (penetration testing, hardening, tools like Wireshark, Metasploit, ethical hacking). Work mode: Remote/office hybrid possible (Jaipur office visits if required, expenses...

Looking for my professor. I do have some basic knowledge of PBX over the years. I need a teacher to help me learn few things about FreePbx. I think I am pretty good at Telnyx side but Pbx side is going ok, but your expertise will make it better.

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

I need a production-ready softphone for both iOS and Android built on both 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 customers, then expose a clean, corporate-style interface that matches the rest of our product line. You must be able to provide examples of apps you've made in the past which utilise both SIP and WebRTC. This might consist of screenshots, code samples or demos of apps. Core scope • Local audio mixing for conferenced/merged calls - this must be done on the device (might require native code) and will likely be the most challenging part of the project as our server does not support mixing of audio. • Ad-hoc conference/merge, BLF,...

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

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

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

We are building a structured AI-powered call routing system in South Africa. The system must: • Integrate with existing PBX systems via call forwarding or SIP • Use a South African virtual number • Route inbound calls through an AI voice receptionist • Identify query type • Provide structured information • Escalate security-related matters • Send SMS notifications when required • Log call analytics This is NOT a chatbot project. This is a voice AI + VoIP routing infrastructure project. Technical Requirements: Developer must have experience with: • SIP / VoIP integration • PBX systems (3CX, Yeastar, Telkom, etc.) • Twilio or similar telephony APIs • AI voice agent implementation • Call forwarding configurat...

... 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 ...