Asterisk pbx jobs
My 3CX PBX is correctly receiving incoming calls through our Twilio SIP trunk, yet every outbound attempt fails. due to outbound rules. i have setup this but i think i am missing one of the check mark or something it is not working
...Affordance Affordance relates to the immediate recognition of what an interactive element does. Confusion can occur if, for example, items are greyed out but are still clickable or reachable. 7. Help your customers A website should minimize user errors and provide assistance, for example, by having a dedicated "Help" section divided into subsections. Highlighting unfilled boxes in red or using an asterisk (*) to indicate importance helps ensure visitors do not make errors. 9. Fast and responsive Website speed and responsiveness are crucial. The design should adjust well to different screen sizes and orientations. Consistency between mobile and desktop experiences, including images correctly resizing and fonts displaying correctly, is required. Slow loading speeds, such ...
My FreePBX install has suddenly stopped accepting any inbound calls. The configuration has not been touched for a while, yet every number on the system now rings out with no sign of life inside the PBX and no visible error messages. Outbound calls still work, trunks show as registered, but nothing is landing on the extensions. I need a seasoned FreePBX / Asterisk troubleshooter to jump in, trace what is blocking the inbound route, correct whatever trunks, inbound routes, firewall, or SIP settings are at fault, and get calls flowing again. Once it is working, I would like a brief note of what you changed so I can keep the system stable going forward. You will have full web GUI access on request; testing can be done live as this is a production box. I am in Australia, so I wi...
...enough for the first release. Deliverable checklist 1. FreeSWITCH SBC build files and working configuration with Opus transcoding enabled. 2. MariaDB schema covering clients, PBXs, numbers, and routing priorities. 3. Web portal to add / update / delete routes and instantly reload FreeSWITCH. 4. README with deployment steps, firewall ports, and a quick test plan (two SIP softphones to emulate PBX and carriers). The solution must install on a fresh Debian VM and pass a live demonstration: a US call failing over to Carrier 2 on simulated 503, and an EU call doing the opposite....
...experienced Machine Learning Engineer specialized in audio processing and deep learning. The goal is to design, train, and deploy a high-performance AMD (Answering Machine Detection) model for telephony, using an existing dataset of approximately 67,000 labeled audio samples. The model must operate in real-time with low latency, and integrate into our existing calling infrastructure (Drachtio / Asterisk / FreeSWITCH / Vicidial). Mission Responsibilities: Analyze and preprocess the existing dataset (cleaning, balancing, train/val/test split) Extract audio features such as Mel-spectrograms, MFCC, STFT, normalization Design and train a CNN/CRNN model for AMD classification (Human / Voicemail / Silence / Fax / Other if needed) Optimize the model for real-time inference (target &...
Our production VoIP app runs on Kamilio/Asterisk is suffering from three clear symptoms: call dropping, audio quality issues. The trouble appears only on a fraction of calls rather than every connection, so I need someone who can track down the underlying triggers—whether they sit in the signalling layer, jitter buffer, codec choice or network traversal logic—and then eliminate them. You’ll start by reviewing the current SIP/WebRTC stack, server logs, packet captures and any client-side metrics I provide. From there, pinpoint the root cause and apply code, configuration or infrastructure tweaks to stabilise audio flow. Deliverables • Detailed diagnosis outlining where packets or media streams fail • Fixed client or server code / configs ready for ...
I need a solid back-end that lets a future React Native client place and receive calls with the reliability of a carrier-grade PBX. The server must handle: • Incoming and outgoing calls • Call forwarding and call transfer • Voicemail storage/retrieval • A flexible auto-attendant (IVR) My preference is to stay in the React Native ecosystem for the client side, but I’m open to your guidance on the most appropriate SIP/WebRTC stack, media server (Asterisk, FreeSWITCH, Kamailio, etc.), and signalling approach. Please outline the architecture you propose, the main tech you’d employ, and an estimated timeline for delivering a first working build that can: 1. Register soft-phones via SIP or a comparable protocol 2. Complete internal an...
...Affordance Affordance relates to the immediate recognition of what an interactive element does. Confusion can occur if, for example, items are greyed out but are still clickable or reachable. 7. Help your customers A website should minimize user errors and provide assistance, for example, by having a dedicated "Help" section divided into subsections. Highlighting unfilled boxes in red or using an asterisk (*) to indicate importance helps ensure visitors do not make errors. 9. Fast and responsive Website speed and responsiveness are crucial. The design should adjust well to different screen sizes and orientations. Consistency between mobile and desktop experiences, including images correctly resizing and fonts displaying correctly, is required. Slow loading speeds, such ...
I need a fully-functional outbound voice agent that plugs straight into my VICIdial / GoAutoDial stack (Asterisk) and dials our CANADIAN leads through the SIP trunk we already use. Once connected, the bot must carry the entire conversation in a friendly, relaxed tone—answering common questions, keeping to our compliance wording, then moving naturally into a soft-close or handing the call over to a live agent when the prospect is ready. Key things I will be testing for: extremely low latency between speech-to-text, LLM response and text-to-speech; barge-in so the prospect can interrupt without awkward pauses; sentiment and intent tracking so the call flows feel human; and a clean handoff of every outcome, recording and transcript back into VICIdial and our CRM. Deliverables I...
I have a Cisco CP-9951 that is still running the factory SCCP load. My FreePBX instance is already up, stable, and ...the correct XML config so it boots straight into SIP and talks cleanly to FreePBX. When the flashing is done, the phone should: • Register to my existing FreePBX extension and pass audio both directions • Retain its settings after reboot • Display the correct time/date from the PBX or NTP server I also want a concise, copy-paste-ready set of instructions so I can repeat the process for any additional CP-9951 units in the future. We can use SSH, AnyDesk, or another secure remote tool for access to the PBX and TFTP host, whichever you prefer. If you have deep experience with Cisco enterprise phones, TFTP provisioning, and FreePBX, let&rs...
...current PBX. • Deployment instructions (Docker/Kubernetes or similar) so we can run everything on our own infrastructure, not as a public SaaS product. • Source code, installation scripts, and a brief hand-off session once the prototype meets the test cases below. Acceptance criteria 1. A test call to the support number is routed, answered by the agent, logged in the CRM, and transcribed in real time. 2. An outbound lead-qualification scenario dials a provided number, carries a short scripted conversation, and posts the outcome back to the CRM. 3. Audio quality and speech latency remain below 300 ms round-trip on our internal network. 4. All components run behind our firewall with environment-specific configuration files. If you already have experience wit...
Replicate on another server the current (working) asterisk + vapi configuration, then create a panel for the APIs and ensure it can be used by multiple servers (each client one server) Max 300 euros.
I’m running Asterisk 20 on an Ubuntu 22.04 (Jammy Jellyfish) server and need it fully configured for secure WebRTC calling with a jsSIP-based browser client. Scope • Enable TLS for SIP traffic and WSS for WebSocket signalling, using self-signed certificates that you generate and install. • Update/adjust all relevant Asterisk, HTTP, and PJSIP settings so that a browser can register and place calls without certificate warnings once the cert is imported. • Deliver a working jsSIP sample page that demonstrates: – two-way audio calls – two-way video calls – basic text messaging (MESSAGE method) • Provide clear, reproducible steps or a shell script that installs dependencies, copies configuration files (, ...
I have a clean server ready and need Asterisk installed and brought to a working “out-of-the-box” state. That covers the core packages, basic SIP/Chan_PJSIP configuration, a starter dial-plan, and all the little touches—service auto-start, log rotation, and sensible security tweaks—so I can begin adding extensions immediately. If you’ve done fresh Asterisk deployments before, especially on Debian or CentOS, tell me where and how many; your hands-on experience will weigh far more than theory. Once you’re finished I want to be able to register two soft-phones, place a test call, and verify audio both ways—nothing fancy yet, just proof the platform is solid. Deliverables (all on my single VPS): • Install the latest stable Aste...
...(Commercial & Residential) – KNX programming (ETS6), dimming, lighting control, shutter control, smart-home panels, scenes • Security Systems – CCTV, Alarm Systems, Access Control, Video Intercom, Fire Alarm Systems • Networking & Infrastructure – Structured Cabling (Cat6/Cat7), Fiber Optic installation & splicing, WiFi systems, router/firewall/switch setup, rack/cabinet installation • Telecom – PBX & VoIP Systems • Technical installation & support for hotels, villas, offices, and commercial buildings ⸻ What I need the freelancer to deliver: 1. Setup • Create search filters for Cyprus business sectors (developers, architects, construction, hotels, facility managers, offices) • Build reusable presets &b...
NEC PBX License Exceeeded Support , Still had licnese 5104
I’m putting together a lightweight voice platform in Python / Django that can register my existing SIP account and let me place and receive calls right from a browser. Think of it as a small PBX-style module but stripped down to the essentials: secure SIP registration, two-way audio over WebRTC, and a clean web phone interface. Core expectations • Django project that handles user auth plus SIP credential storage (encrypted) • Sign-in to the SIP registrar, keep the registration alive, and expose call events through WebSockets • Browser client (JavaScript, preferably using JsSIP or ) that dials out, answers, puts on hold, hangs up, and shows call status • Inbound call pop-up in the UI with basic accept/decline controls • Outbound calls trigger...
...Cisco, softphones, etc.). • Integrate the generator with my PBX/SBC/network. • Optional: configure or verify QoS (DSCP, prioritization). • Perform final tests and provide a summary/report. • Provide documentation and basic knowledge transfer so I can run tests myself later. Required Experience: • Strong knowledge of VoIP: SIP, SDP, RTP, codecs (G.711, G.729, Opus). • Practical experience with traffic generators (SIPp, StarTrinity, or similar). • Hands-on experience with load testing (hundreds of concurrent calls). • Good understanding of networking (UDP, NAT, VLANs, QoS, jitter, packet loss). • Ability to troubleshoot SIP/RTP using Wireshark or similar tools. Bonus Skills (not required but preferred): • Experience with...
I need fully functional voice bots connected to my existing Asterisk + ASTPP setup. The scope covers three distinct use-cases—customer service, sales inquiries, and technical support—so each bot should be able to route calls intelligently, answer routine FAQs, and capture key caller details for later reporting. My current preference is to build the dialogue logic in Flowise because of its visual workflow designer and native webhook support, but if you can demonstrate a more efficient approach with a comparable platform I’m open to hearing it. What matters most is that the solution plugs cleanly into Asterisk, respects ASTPP billing logic, and keeps latency low. Please include: • Configuration of SIP endpoints, dial-plan changes, and any AGI/ARI scri...
I want to bring an AI-driven live-call translation system to life, which works web to mobile without downloading anything software or app to mobile as normal calls. delivered as companion web interface. There will 2 web dashboard, 1 for admin and 2 for user. Core functionality is clear: during any phone conversation—whether dialed through Asterisk, VICIDIAL, SIP carrier—the System should capture each side’s speech, convert it into real time in both native languages, run an AI engine such as Deep Gram, Google Voice AI, etc., and speak the translated audio back to the listener with minimal latency. Two-way, real-time translation is non-negotiable; speech-to-text, text-to-speech, and airtight VOIP integration all need to feel seamless to the user. On the technical sid...
I need a SIP professional to link my existing number with my on-premise Grandstream PBX and leave me with a fully registered trunk that can place and receive calls without hiccups. Both platforms are already active, so the job focuses purely on configuration rather than account creation. Here’s the brief: • Implement two trunks—one that registers via static IP and another that authenticates with username / password—so I can switch methods on demand. • Complete the setup remotely on the Grandstream interface and in the console, then run live test calls to confirm two-way audio. • Provide a short note of any parameters you touch (SIP server, NAT, codecs, authentication strings, firewall rules) so I can replicate or edit the settings later. The...
...calls, reply to emails, and manage the live-chat widget. • Converse naturally in Italian at every touchpoint. • Answer both general queries and product/service-specific questions with accurate, up-to-date information. • Propose and lock in an appointment on my calendar, confirming date and time with the customer before ending the interaction. I’m open to the stack you prefer—Twilio, Vonage, Asterisk, Dialogflow, GPT-4, Rasa, or comparable solutions—as long as it can be integrated with my existing CRM and calendar. I’ll provide API keys, brand guidelines, and a small knowledge base; you’ll craft the conversation flows, configure the telephony and chat channels, set up email parsing, and train/tune the language model for Italian. ...
I have a fresh server waiting for a clean Asterisk installation and I want it to come online already loaded with both VC Dial and a working Voice-AI module. Your mission is to take me from zero to a fully functioning system that can place and receive calls through VC Dial while handing certain interactions off to the AI engine. Core expectations • Install the appropriate Asterisk version on my Linux VPS (CentOS/Ubuntu, your preference as long as it is stable). • Deploy VC Dial, confirm the web interface, database and dialer services all start automatically, and create a sample inbound + outbound campaign so I can see calls flowing. • Integrate the Voice-AI component (Google, AWS, or another open-source alternative—let me know what you prefer) so that...
I need Asterisk 22.6.0 set up on my Windows laptop. Please enable the following modules: - Voicemail - Conference calling - Call recording Ideal skills and experience: - Expertise in Asterisk, especially version 22.6.0 - Proficient in Windows OS - Experience with VoIP and PBX systems - Knowledge of configuring Asterisk modules
I am rolling out a brand-new Odoo instance together with a fresh Grandstream PBX and want them to work as one seamless system from day one. The core goals are clear: every call handled by Grandstream should be logged inside Odoo, contacts must stay perfectly synchronized between both platforms, and voicemails need to appear in Odoo with playback and status indicators. On top of that foundation, I’d like to weave in three AI capabilities: • an automated response layer that can greet callers and route them intelligently, • live voice-to-text recognition so conversations and voicemails are transcribed directly into Odoo records, and • predictive analytics that can flag high-value leads or at-risk customers based on call patterns. Because nothing is inst...
...configure Kamailio as an SBC together with RTPengine. The goal is to fully rewrite all SIP and RTP traffic so that the upstream provider sees only the public IP of the Kamailio server — no trace of the original Asterisk system. Connection from Asterisk to Kamailio will be over direct IP (SIP trunk), no DNS. Environment (already prepared): Clean Debian 12 (on a dedicated server, not VPS) Kamailio 5.6.3, MariaDB, RTPengine installed Proper DNS and Debian repositories configured SSH access is ready Task Requirements: Configure Kamailio as an SBC for SIP trunk traffic coming from Asterisk Integrate and configure RTPengine for full media relay and SDP rewriting Ensure all SIP headers (From, Contact, Via, etc.) and SDP contain only Kamailio’s public ...
...Entire system must use free and open-source software (no per-minute or API-based billing). Deployable on Linux (self-hosted) environment. Deliverables: Fully working AI voice server integrated with VICIdial. Web GUI for survey management and transcript viewing. Auto disposition and full transcription workflow. Installation + setup documentation for self-hosting. Preferred Skills: VICIdial / Asterisk integration. LLaMA / local LLM deployment (for dynamic survey conversations). Open-source ASR/TTS: Whisper, Coqui, Kaldi, etc. Linux server administration, SIP routing, and AI voice automation. 100% open-source (no API or per-minute fees) Self-hosted solution for complete control Ready for real-world UK outbound survey campaigns...
...variations (e.g. crust type, toppings, drink size, etc.). Order Confirmation & Printing: After completing the order, the system will generate a receipt/order slip and print it automatically in the restaurant (through a connected printer or POS integration). Optionally send a confirmation text or summary to the client. Technical Requirements (flexible): ChatGPT or equivalent LLM integration. Twilio, Asterisk, or similar API for voice/call handling. Database for storing customer profiles and order history. Printer or POS connection (API, local, or cloud print). Dashboard (optional) for restaurant admins to view/edit orders and menus. Objective: A fully automated phone ordering system that lets restaurants take orders 24/7, save labor costs, and reduce order errors — all...
...Entire system must use free and open-source software (no per-minute or API-based billing). Deployable on Linux (self-hosted) environment. Deliverables: Fully working AI voice server integrated with VICIdial. Web GUI for survey management and transcript viewing. Auto disposition and full transcription workflow. Installation + setup documentation for self-hosting. Preferred Skills: VICIdial / Asterisk integration. LLaMA / local LLM deployment (for dynamic survey conversations). Open-source ASR/TTS: Whisper, Coqui, Kaldi, etc. Linux server administration, SIP routing, and AI voice automation. 100% open-source (no API or per-minute fees) Self-hosted solution for complete control Ready for real-world UK outbound survey campaigns...
...trunk must support both inbound and outbound traffic. I’m starting from scratch with no existing VoIP provider, so you are free to source and provision the most reliable carrier that meets these requirements: • Local Polish and UK DIDs with proper caller-ID presentation • G.711 and, ideally, G.729 codecs • TLS/SRTP or similar secure transport options • Smooth integration with common PBX platforms such as FreePBX or 3CX Scope of work 1. Recommend and acquire the best-fit carrier(s). 2. Provision the trunks and obtain registration credentials. 3. Conduct test calls in both directions to verify audio, DTMF, and CLI. 4. Deliver concise setup notes so I can replicate the configuration in production. Highlight your experience pr...
יש לי שרת Asterisk פעיל שכבר משתמש בקריאות הישנות של OpenAI. כעת אני צריך לשדרג אותו למימוש Realtime API כדי לאפשר צ'אט בזמן אמת. המטרה ברורה: לקבל תגובות זורמות (streaming) באופן מיידי, בלי לפגוע בשירותי הטלפון שכבר עובדים. מה העבודה כוללת • החלפת או הרחבת הקריאות הקיימות כך שיתמכו ב-streaming דרך WebSocket/HTTP Streaming. • שמירת הלוגיקה העסקית וה-endpoints הנוכחיים של המערכת. • אופטימיזציה לחביון נמוך ויציבות קו. • בדיקה על סביבת staging ולאחר מכן פריסה ל-production. • מסירה של קוד מעודכן + הוראות התקנה ותיעוד קצר. קריטריוני קבלה 1. הודעת טקסט שמגיעה לשרת מחזירה תשובת מודל בזמן אמת ללא דיליי מורגש. 2. שום שירות קיים לא נשבר. 3. כל הקוד מגובה בגיט ומדגים בבירור איך לחבר, לבנות ולהפעיל. יש לי גישה מלא...
Implement integration of SIP telephony with voice AI agent for handling incoming/outgoing calls through Asterisk or alternative Initial Data Telecommunication operator: TBD Telephony server: Asterisk (Linux VPS/server, dedicated IP) Task: automatic call handling through AI agent in real time. Main Tasks 1. Connecting SIP Trunk Configure Asterisk or similar Check routing of outgoing calls. Self-assembly (STT + LLM + TTS) Use a fast engine for STT (for example, Deepgram, Google Speech-to-Text, AssemblyAI). Dialogue logic on GPT (OpenAI) Speech synthesis via Deepgram TTS, Google TTS, Microsoft Azure TTS. Ensure audio streaming to minimize latency (practically in real time). 3. Functional Requirements Automatic response to incoming calls by the voice agent. Sc...
Implement integration of SIP telephony with voice AI agent for handling incoming/outgoing calls through Asterisk or alternative Initial Data Telecommunication operator: TBD Telephony server: Asterisk (Linux VPS/server, dedicated IP) Task: automatic call handling through AI agent in real time. Main Tasks 1. Connecting SIP Trunk Configure Asterisk or similar Check routing of outgoing calls. Self-assembly (STT + LLM + TTS) Use a fast engine for STT (for example, Deepgram, Google Speech-to-Text, AssemblyAI). Dialogue logic on GPT (OpenAI) Speech synthesis via Deepgram TTS, Google TTS, Microsoft Azure TTS. Ensure audio streaming to minimize latency (practically in real time). 3. Functional Requirements Automatic response to incoming calls by the voice agent. Sc...
I want to bring an AI-driven live-call translation system to life, delivered as both a mobile app and a companion web interface. The mobile side must run smoothly on iOS and Android, while the web app acts as an admin and monitoring console. Core functionality is clear: during any phone conversation—whether dialed through Asterisk, VICIDIAL, Twilio, or another SIP/VOIP carrier—the app should capture each side’s speech, convert it to text in real time, run the text through an AI engine such as DeepGram or Google Voice AI, and speak the translated audio back to the listener with minimal latency. Two-way, real-time translation is non-negotiable; speech-to-text, text-to-speech, and airtight VOIP integration all need to feel seamless to the user. On the technical side...
I’m looking for a specialist to put together a simple, always-on customer contact center that can handle day-to-day support without me needing to sit in front of a screen. The core...appointment scheduling, with smooth handoff to a live agent when needed. • Clear guidance on how to keep the system running around the clock, including monitoring and simple fallback steps if something goes down. • A short walkthrough or documentation so my small team can add new FAQ answers or tweak business hours without calling you every time. I’m open to the specific tools you recommend—Asterisk or FreePBX for voice, Amazon Connect, Twilio, or other reliable platforms are all fine as long as they stay within a lean server footprint. Keep the setup straightforward, se...
...of OpenSIPS secured with TLS and fully integrated with Asterisk. The goal is to deliver a working voice platform that lets me: • Route calls through OpenSIPS • Handle voicemail • Support user-to-user calls and call forwarding On the web layer I want clean, well-documented REST endpoints so external services can create, monitor, and terminate calls or fetch voicemail status. You’re free to choose the most stable 3.x OpenSIPS release (3.1 LTS, 3.2 Latest, or 3.0 LTS) as long as you justify the decision and ensure future maintainability. Scope I expect 1. Install and harden OpenSIPS on my VPS, enabling TLS for SIP traffic (certificate provisioning, cipher suite tuning, SRTP hand-off to Asterisk). 2. Install or reuse an Asterisk build and...
We are a security company based in the Republic of Congo using local Airtel SIM cards with a GFU plan that allows free internal calls between our lines under a monthly subscription. Our operations team currently calls security guards on-site manually every 30 minutes to ve...center platform that automates and manages these check-in calls, logs call history, monitors status, and can be used by multiple operators simultaneously. The system must function locally, integrate with GSM gateways or multi-SIM hardware, and not require ongoing monthly fees beyond our existing Airtel plan. The ideal candidate should have prior experience setting up GSM gateway-based call centers, PBX systems, or similar local telephony integrations, and be able to deliver a turnkey setup ready for daily operat...
I need a clear, professional-sounding male voice to record ...notes for company names and technical terms. What I expect back: • Two high-quality WAV files per prompt—one English, one French—recorded at 48 kHz / 24-bit, ready for direct upload to the VoIP IVR. • Clean, noise-free audio with breaths and clicks removed. • A short pause trimmed to 0.5 sec at the start and end of each file for seamless looping. After delivery I’ll test the prompts on the live PBX and confirm volume matching; if any file needs a minor tweak in timing or pronunciation, please include one round of revisions. If you have a broadcast-quality home studio and are comfortable switching fluidly between English and French while maintaining the same tone, I look forward to heari...
I am looking for a skilled developer to create a web application for automating Airtel LAPU recharges using an API. The application should streamline the recharge process and provide a comprehensive management system. Key Requirements: - Develop a web-based application for Airtel LAPU recharge automation. - Implement features for recharge management, transaction history, user authentication, and balance addition. - Ensure secure and efficient handling of user data and transactions. Ideal Skills and Experience: - Proficiency in web application development. - Experience with API integration, particularly for telecom services. - Strong understanding of user authentication and data security. - Ability to create intuitive and user-friendly interfaces. I am eager to collaborate with a develop...
I need a voice-driven virtual receptionist that picks up every inbound call, greets the caller, and intelligently routes the call to the right extension or mobile number. If the caller asks to continue in another language, the system should switch languages on the fly and keep the conversation natural. Core expectations • Cloud or on-premise voice gateway (Twilio, SIP, Asterisk, or similar) that ties into my existing phone line • Speech recognition, NLU, and text-to-speech working together so callers can speak freely, not just press keys • Reliable call transfer with hold music or custom messages while the destination rings • At minimum two languages at launch; adding more later should be configuration, not code re-writes • Simple admin dashboard ...
...Google) Enable/disable features per plan or per user Manage call storage and auto-deletion after 6 months View, download, or delete call recordings Manage KYC verification of businesses Generate system-level reports 5. Backend & Telephony Integrations Entire backend will operate via secure APIs. Integrations with any reliable telephony or VoIP service provider (Twilio, Plivo, SignalWire, or Asterisk). APIs to handle: Incoming/outgoing calls IVR flow setup Call recording and playback Call forwarding and routing Call analytics and distribution logic Real-time webhooks for call status and duration Storage for recordings (AWS S3 or similar) APIs must return all data in JSON format for both web and mobile use. 6. Payment Gateway Integration Integration with Indi...
ONLY - PREBUILD - DONT WASTE TIME TO MESSAGE ME IF YOU NEVER DONT BEFORE AND YOU HAVE NO IDEEA PLANING TO GET IT IN 5 DAY MAXIMUM I need a complete Asterisk-based outbound platform fine-tuned for high-volume marketing calls. The core flow is simple: the system auto-dials a list, detects whether a human picks up, skips voicemail instantly, and routes every live answer straight to a waiting agent. Essential features • Voicemail logic – detect voicemail reliably, bypass Google Voice screening by having the dialer state my company name when prompted, and, if no human answers, automatically redial the same number up to 4 times during the campaign window. • Live transfer – answered calls must land in a real-time agent queue (SIP soft-phones or WebRTC, whichever you...
...cloud-based SIP PBX. The VPN will serve purely as a private communication network, not just remote access, so every endpoint—office phones, softphones, and any future branch sites—must pass voice traffic through it. On the PBX side I only need reliable call routing and forwarding at this stage; features such as voicemail, recording, or conferencing can remain optional add-ons if the platform you choose already supports them. Feel free to propose Asterisk, FreePBX, 3CX, FusionPBX, or another cloud offering—as long as it can live behind the VPN, handle SIP trunks cleanly, and scale without headaches. Deliverables (kept short and specific): • Deploy and configure the VPN with strong encryption, user authentication, and clear subnet planning...
I need a working call-automation flow th...Google Calendar / Outlook once the prospect shows interest A functional prototype is enough for the first milestone—I want to hear the bot live on a test number, handling the two tasks above smoothly and handing off qualified leads to a human. Please show me past work that proves you have already built voice-AI or n8n call workflows. Mention which telephony stack you prefer (Twilio, Plivo, Asterisk, etc.) and any speech-to-text / LLM services you plan to integrate. Success for this project means: • Natural-sounding conversation with <2 sec latency • Appointment details correctly logged with contact info • Call recordings and transcripts saved in my CRM If you can deliver within those parameters, I’m r...
...protocols Configure and manage media servers (FreeSWITCH, Asterisk, RTP Proxy) Work with SIP proxy servers (Kamailio, OpenSIPS) for call routing and signaling Build and maintain call recording solutions using SIPREC Handle WebRTC, RTP streams, and VoIP media processing Develop automation scripts and services using Node.js, Lua, or Python Integrate with relational databases (Postgres, MySQL) Deploy and manage solutions in cloud-native environments (GCP preferred; AWS, Azure) Ensure high availability and scalability using HAProxy or load balancers Collaborate with cross-functional teams for deployment, monitoring, and troubleshooting Required Skills: Hands-on experience with SIPREC for VoIP call recording Expertise in FreeSWITCH / Asterisk / RTP Proxy (media serv...
...looking for an engineer who can stand up a production-ready Kazoo cluster across eight fresh CentOS 7 servers. The platform has to carry roughly 50,000 concurrent calls while staying lean on bandwidth, so efficiency and proper resource tuning are essential. My top priority is a rock-solid call center component with dependable call-termination routing. Around that core, I also need: • Multi-tenant PBX functionality so each customer lives in its own space. • A clean, web-based GUI for both users and admins. • Self-service user registration plus zero-touch auto-provisioning for common SIP endpoints. • End-to-end subscription handling that mirrors the advanced flows found in Zelle and CashApp. • A single “one-click” script that genera...
I need a complete Asterisk-based outbound platform fine-tuned for high-volume marketing calls. The core flow is simple: the system auto-dials a list, detects whether a human picks up, skips voicemail instantly, and routes every live answer straight to a waiting agent. Essential features • Voicemail logic – detect voicemail reliably, bypass Google Voice screening by having the dialer state my company name when prompted, and, if no human answers, automatically redial the same number up to 4 times during the campaign window. • Live transfer – answered calls must land in a real-time agent queue (SIP soft-phones or WebRTC, whichever you prefer to configure). • API hooks – integrate both an Email marketing service and the Doxer API so that contact...
I need a purpose-built module for Issabel that turns the PBX into a fully featured public-address engine. The add-on must let me trigger live or pre-recorded broadcasts to any SIP endpoint registered as a PA speaker, schedule automated messages, and issue one-click emergency alerts that can override all ongoing audio. Beyond simple paging, I also want two-way talkback so selected devices can return audio to the control room. Audio quality matters; the module has to negotiate and play high-quality codecs such as Opus or G.722 whenever endpoints support them, while falling back gracefully to standard G.711. A browser-based dashboard should give me at-a-glance status of every PA device—online/offline indication, current stream, volume level—and provide controls to group e...
I have a Grandstream UCM6300A that is ...inbound call never reaches the phones—there’s no ring at all. Here’s what I already know • The SIP trunk shows as “registered.” • Inbound routes are set up and look correct. Despite that, calls still fail to ring, so I’m clearly missing a detail somewhere (DID pattern, NAT, firewall or similar). I need someone who understands Grandstream UCM-series quirks to: 1. Remotely inspect the PBX configuration (trunk, inbound routes, extensions, firewall / NAT). 2. Pinpoint and resolve whatever is blocking the ring. 3. Provide a short summary of the fix so I can repeat it if needed. Screen-sharing or a quick SSH/HTTP session is fine; I’ll supply access. Once your tweaks let the pho...
