SipJobs

ONLY BUILD IF YOU HAVE STRONG EXPERIENCE IN THESE AREAS SIP, NAT Traversal, Asterisk/FS, React and strong backend servers scaling. I’m building a browser-based VoIP platform dedicated to business communication and I need an experienced React developer to take it from architecture to live deployment. The entire feature list—covering everything from secure voice and video calling to messaging and call-related utilities—is spelled out in the requirements document I’ve attached, so you’ll have clear, granular guidance from day one. SIP, NAT Traversal, Asterisk/FS Tech expectations You’ll craft a responsive single-page app in React (TypeScript preferred) that connects to a SIP/WebRTC back-end. I’m open to your preferred server...

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-deployment guid...

3CX security hardening and SIP failover

...USA-based SIP trunk through Amazon Chime. The job includes creating or hardening the AWS environment, provisioning an Amazon Chime Voice Connector, assigning U.S. numbers, and enabling both inbound and outbound calling. The trunk has to register cleanly with our existing on-premise PBX and soft-phone apps, so please allow for SIP signaling tweaks, codec selection, and basic security (TLS, SRTP, firewall rules). Once the link is live, I’ll need brief documentation of the key settings so we can maintain it internally. Deliverables • AWS account prepared with least-privilege IAM roles • Amazon Chime Voice Connector fully configured and licensed for U.S. PSTN access • SIP trunk integrated and test calls completed (in/out) with our current ...

I’m looking to have IP cloning / masking configured on my VoIP servers so we can distribute traffic evenly and keep call quality high. Load balancing is the sole objective here—this isn’t about anonymity or region-hopping, just making sure each incoming SIP session finds the least-loaded endpoint without exposing multiple public addresses. The environment is already up and running; what I need is the networking magic that presents one or more virtual IPs while the real servers sit safely behind them. If you normally work with tools such as iptables, nftables, HAProxy, IPVS, BGP anycast, or cloud firewalls, that’s perfect—use whatever combination you trust so long as it’s stable and keeps latency low. Because the phone queues are busy, I need thi...

...Based SIP. Developer must have experience working with SOC like NanoPI and OrangePI. Developer must meet the following list of requirements so as not to waste our time. Embedded Linux Development Strong experience developing software for embedded Linux systems Experience with custom Linux distributions (Debian/Ubuntu) Kernel configuration, device drivers, and hardware interface support Bootloader and system startup (U-Boot or similar) Experience working with flash-based storage systems Linux system optimization for embedded hardware VoIP / SIP / Networking Experience implementing or maintaining SIP-based communications systems Knowledge of RTP / RTCP media streaming Familiarity with VoIP codecs such as G.711, G.729, Speex Understanding of NAT traversal, STUN, ...

I need an IVR built and configured so callers reach the right help fast. Its sole purpose is customer support, and it must offer three options in every relevant branch: an automated response for common questions, the ability to transfer to a live agent, and—when queues are long—a call-back request feature. You’ll design the call flow, record or synthesize the prompts, connect it to our existing SIP/VoIP trunk, and make sure the routing to our support team works flawlessly. I’m open on the exact number of menu levels; clarity and speed for the caller are what matter most, so feel free to recommend the simplest structure that still covers our FAQs and escalation paths. Final hand-off should include: • Diagram of the call flow with all customer touch-poin...

I’m developing a reusable water bottle that physically reminds its owner to take a sip every hour. The cue I want is simple and visual: a small flag should rise from the lid when it’s time to drink and drop back down once the bottle is lifted or the cap is reset. I need a mechanical concept that can live entirely inside (or be integrated into) the bottle cap without relying on batteries, electronics, or external apps. Think clock-work, torsion springs, cam drives, or any passive timing solution that can reliably cycle every 60 minutes as long as the bottle is upright. Here’s what I’d like from you: • A clear explanation of the working principle, including sketches or diagrams that show how the hourly timing is achieved and how the flag deploys and re...

...placeholder SIP stack. I now want to swap that out and connect directly to Telnyx so users can place and receive real calls from inside the app. Here is what I need from you: • Wire the Telnyx voice-calling SDK or raw SIP/WebRTC endpoints into the existing Android codebase. • Implement secure authentication with my Telnyx account, handle inbound and outbound call flows, and expose simple callbacks so my UI can update on ringing, answered, muted, held, and ended states. • Provide brief build/run documentation plus any configuration files required for future environments. I will share the current repository and my Telnyx credentials once we agree on scope. When you respond, please point me to past work—links, APKs, or repos—where you have d...

...specjalisty, który poprawnie skonfiguruje połączenie z moim dostawcą SIP trunk – Zadarma – tak, aby system bezbłędnie realizował zarówno połączenia wychodzące, jak i odbierał przychodzące. Zakres prac: • dodać i zweryfikować trunk Zadarma, • utworzyć odpowiednie dial-plany i reguły routingu, • przeprowadzić testowe połączenia w celu potwierdzenia jakości oraz stabilności, • wprowadzić ewentualne korekty konfiguracji serwera Asterisk wchodzącego w skład GoAutoDial. Środowisko obejmuje jeden VPS, więc konfiguracja dotyczy pojedynczego serwera. Po zakończeniu proszę o krótką instrukcję krok-po-kroku, abym mógł samodzielnie zarządzać podstawowymi zmianami w przyszłości. Jeżeli masz doświadczenie z GoAutoDial i integ...

...onboard speaker, WiFi) — 2.8" TFT LCD Touch Screen (ILI9341, 320x240, SPI) What "working" means: Connects to WiFi Registers to a SIP server (FreePBX, , Twilio — whatever's easiest) Can make and receive a voice call Two-way audio through the onboard mic and onboard speaker Display shows basic call status: idle, calling, incoming call, in-call That's it. If I can call a phone number or another SIP device and have a conversation while the screen shows what's happening, we're good. Deliverables: Source code I can compile and flash Quick notes on how to set it up (SIP creds, WiFi config, wiring for display) Short video showing it actually making a call Don't need (yet): Echo cancellation, noise suppression, OTA u...

...functional, non-functional, and failover testing, mapped to the features above • Recommended automation approach (e.g., Selenium for UI where applicable, toolkits such as Cyara, Hammer, or home-grown scripts) with pros and cons • Performance test plan outlining load profiles, KPIs, and tooling suggestions (JMeter, LoadRunner, or similar) • Environmental matrix covering telephony components (SIP, PRI), IVR platform (Genesys or Twilio Studio), speech engines (Nuance, Google Dialogflow), and downstream APIs • Risk analysis plus mitigation tactics • A clear traceability table linking requirements, tests, and expected outcomes I’m not asking for actual test execution right now—just the blueprint that will let us execute efficiently in a ...

...installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architecture, including SIP trunks for external connectivity and internal extensions for users. Detailed Scope of Work 1. Server Installation & Hardening Perform a clean installation of Issabel 5 on AlmaLinux 8 using the official net-install script. Configure Fail2ban and firewall rules to block unauthorized SIP and SSH attempts. Set secure passwords for the Linux root, MariaDB, and Issabel web admin. 2. SIP Trunk Configuration Connect the PBX to my chosen VoIP provider using a SIP trunk. Configure PEER details (host, username, secret) and registration strings. Set up Inbound Routes to direct incoming DID numbe...

...credentials access into. I want to be able to present a specific caller ID on every outbound call placed from my softphone. The softphone I will be using is Linphone, so everything you build must be fully tested from that app before we wrap up. What I need you to do • Configure the relevant SIP profile(s) and outbound route so that the number I supply appears consistently, regardless of the device extension that initiates the call. • Make any dial-plan or gateway adjustments required for a typical commercial SIP trunk. I have not fixed on one provider yet, so keep the setup provider-agnostic and document the few fields I will have to tweak once I decide. • Show me exactly where in FusionPBX I can change the caller ID later, without having to edit XML ...

...Text-to-speech: Respond with friendly, human-like audio. • Fallback: whenever the system is unsure, it should politely ask for more details rather than handing the call off or giving a canned line. I need the full stack—STT, intent handling, response generation, and audio playback—wrapped in a module that I can drop into my existing website widget today and expand to a phone line via Twilio (or a similar SIP/VoIP service) later. Keep the integration layer simple: a REST webhook or lightweight SDK is perfect. Acceptance criteria 1. Demo page or endpoint that I can test from Chrome: user speaks, system replies. 2. Correct answers for “What time do you open?”, “When do you close?”, and “Where are you located?”. 3. When...

...pull from a knowledge base I can update myself. • Live call routing: when the AI decides human help is needed, it should hand the caller off to a live agent instantly and log the transfer. • Voice channel only: chat or email may be added later, but right now the system must excel at telephone interactions. Technical expectations – Cloud telephony integration (Twilio, Plivo, or similar) for SIP/VoIP handling. – AI engine (Dialogflow, Amazon Lex, or comparable) for intent recognition and response generation. – Web dashboard built with a modern stack (React or Vue preferred on the front end, Node or Python on the back) showing call logs, live status, and simple training tools for the bot. – Secure login, role-based access, and scalable ...

...Text-to-speech: Respond with friendly, human-like audio. • Fallback: whenever the system is unsure, it should politely ask for more details rather than handing the call off or giving a canned line. I need the full stack—STT, intent handling, response generation, and audio playback—wrapped in a module that I can drop into my existing website widget today and expand to a phone line via Twilio (or a similar SIP/VoIP service) later. Keep the integration layer simple: a REST webhook or lightweight SDK is perfect. Acceptance criteria 1. Demo page or endpoint that I can test from Chrome: user speaks, system replies. 2. Correct answers for “What time do you open?”, “When do you close?”, and “Where are you located?”. 3. When...

...integration and ready to carry live traffic. The core signalling will run over SIP, so every module you build or configure must interoperate cleanly with SIP endpoints and the upstream carrier trunks I already have in place. Billing is the priority: once a call lands on the switch the CDRs must flow straight into our existing rating platform without manual touches. I am open to whether you plug in a ready-made mediation layer or write custom logic—what matters is that usage records appear in real time and reconcile correctly at the end of each day. You will get SSH access to a fresh cloud instance plus the credential set for my billing server. I expect you to: • Deploy or compile the soft-switch software, enable SIP, and confirm two-way audio on test n...

...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 dashboard or API that lets me view logs, update routing rules, and retrain intents when new scenarios appear. 3. ...

...on the business itself. Most of the work is remote and ongoing. Typical tasks include adding or removing users, managing licences, tightening security policies, monitoring storage limits, resolving sync or mail-flow problems, and fine-tuning Teams, OneDrive, SharePoint, and Exchange whenever needed. On the voice side, I’ll call on you for new handset or softphone rollouts, call-flow changes, SIP or trunk tweaks, quality-of-service checks, and the occasional deep dive when call quality drops. I value quick response times, clear communication, and proactive recommendations—if you see a way to streamline costs or prevent downtime, say so. A short weekly status note and light documentation of any changes you make will be enough to keep us aligned. Deliverables &b...

...launch the very first blended (inbound + outbound) campaign. Here is what I need you to walk me through and, where necessary, configure directly on the box: • Register my SIP-trunking provider inside Vicidial, confirm two-way audio, and run a quick live test call. • Create one blended campaign with at least a test list, DID routing for inbound, and the appropriate dial plan entries for outbound. • Show me, via screenshare or concise step-by-step notes, how to add new agents, upload lead lists, record custom greetings, and monitor live calls so I can repeat the process after you leave. Acceptance criteria – SIP trunk shows “REGISTERED” and passes both inbound and outbound audio without clipping. – Campaign status reads &ldqu...

Scope of Work & Deliverables: 1. End-to-End Audio Bridge Fix: Resolving the clock domain anomalies to establish ... 1. End-to-End Audio Bridge Fix: Resolving the clock domain anomalies to establish a fully functional bi-directional audio bridge. 2. Explicit Routing Requirements: When the Android device initiates a SIM call, the audio will successfully route to the ESP32 via USB (UAC1). The ESP32 will then transmit this audio to the specific SIP server extension provided. 3. Bi-Directional Communication: The system will ensure clear, two-way audio, allowing the designated SIP extension to converse with the recipient of the Android SIM call. 4. Diagnostic Toolchain: The execution includes the deployment of the UDP telemetry network to pinpoint the failure before the fir...

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 REGISTER without 40...

Fixing audio quality issue on ESP32S3 audio (UAC) - SIP/RTP.

...should refresh the on-screen report so supervisors can act on the information while the call is still active. I am not looking to store the data long-term or trigger downstream workflows—just fast, on-the-spot visibility of what the caller keyed in. Because everything must sit cleanly inside (or alongside) 3CX, please base your solution on technology that plays well with the 3CX Call Control API, SIP messages, or any proven method you have used before for intercepting DTMF within 3CX. A compact web interface or an add-in for the 3CX Management Console would both work; I am open to whichever path lets us deploy quickly and keep maintenance light. Deliverables • Fully functional DTMF capture module integrated with my existing 3CX installation • Live reporting vi...

...criteria 1. Calls triggered from Neodove automatically reach the AI agent. 2. Lead information is read from and written back to the correct Neodove record. 3. Transfer to a human agent occurs within two seconds and preserves call audio. 4. End-to-end test recordings and API logs demonstrate reliability over 50 consecutive calls. If you’ve already connected CRMs with voice bots or handled CTI, SIP, Twilio, or similar stacks, your experience will be invaluable. I’m ready to start as soon as we agree on the technical path and timeline....

...for: Setting up AI voice calling (Inbound & Outbound) Integrating: LLM (Gemini, Chat Gpt) Speech-to-Text (Tamil & English) (Sarvam) Text-to-Speech (Eleven Labs, Gemini) Telephony (SIP / DID / API-based calling) (Tata & Other Providers) Designing rule-based call flows Handling: Yes / No / Silence Basic objection handling Fallback to human agent Logging call data (basic logs / Google Sheets / webhook) --- Must-Have Skills ✅ Experience with Voice Bots / AI Calling / IVR ✅ API integration (REST – basic level) ✅ Call flow & conversational logic ✅ Familiar with tools like: Twilio / Exotel / Plivo / SIP providers Dialogflow / Voiceflow / custom logic STT / TTS APIs / LLM * Tamil language experience is a BIG plus --- What This Proje...

...Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs before sign-off. Logistics • Deliverable: fully functioning dialer-to-gateway path with successful test calls, documented settings backup. • Payment: ₹3,000 released the same day once the above criteria are met. Bring your laptop and any tools you normally use ...

...escuchar. Necesito que localices la causa, la corrijas y garantices estabilidad tanto en Android como en iOS. Además del fix, quiero avanzar en el roadmap funcional. Una vez las llamadas queden sólidas estudiaremos incorporar nuevas funcionalidades en la aplicación. Trabajo bajo un esquema por horas, por lo que valoro experiencia demostrable en Ionic/Angular, manejo de plugins de llamada (VoIP/SIP, CallKit/ConnectionService) y soltura con PHP-Laravel para ajustes de API. Imprescindible comunicación fluida en español; la coordinación será diaria mediante Git y board de tareas. Entrego acceso al repo, credenciales de staging y casos de prueba. Espero commits pequeños, comentarios claros y un breve informe de cada sesi&oacut...

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

Mutual Fund Distribution Platform (B2B2C Model) 1. BUSINESS OBJECTIVE Build a digital wealth distribution platform (similar to AssetPlus model) enabling: • Investor on boarding • KYC processing • eSign • Lumpsum & SIP transactions • Mandate creation (eNACH / UPI Autopay) • Portfolio tracking • Distributor dashboard • Commission tracking • Admin & compliance monitoring Platform must support: • Web (Investor + Distributor + Admin) • Android App • iOS App • Multi-tenant SaaS capability (future-ready) 2. USER ROLES 2.1 Investor • Register & complete KYC • Invest in mutual funds • Create SIP • View portfolio • Download statements 2.2 Distributor / Advisor • Onboard investors ...

I need a production-ready voice agent that can speak fluent, natural-sounding Hindi, handle two-way telephone calls end-to-end, and broadcast the conversation live. The goal is a single, deployable service that I can point at a SIP number (or Twilio number) and immediately start taking or placing calls while spectators watch the stream in real time. Core requirements • Real-time speech recognition (Hindi) and TTS with configurable personalities and speed • Dialogue engine that lets me script branching call flows, hand-off to fallback intents, or inject an operator mid-call without dropping audio • Live streaming of the ongoing call (audio only is fine) to a major platform or a lightweight custom player; I’m open to your recommendation as long as latency ...

...Machine B will: • Handle SIP trunk calls • Handle website voice widget calls • Share the same AI Brain + knowledgebase as Machine A • Use centralized architecture for status fetching/updating ________________________________________ Core Architectural Goal Centralized Intelligence Layer We want: Chat (Machine A) Central AI Brain + Knowledgebase + CRM/Portal APIs Call Agent (Machine B) Both Chatbot and Calling Agent should: • Use same vector database • Use same CRM/Portal APIs • Use same site_id architecture • Use same personalization logic • Use same knowledgebase • Use same business logic Only response formatting style will differ (chat vs voice). ________________________________________ Re...

...Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs before sign-off. Logistics • Deliverable: fully functioning dialer-to-gateway path with successful test calls, documented settings backup. • Payment: ₹3,000 released the same day once the above criteria are met. Bring your laptop and any tools you normally use ...

Vicidial is already live on our cloud server; what I need now is an on-site engineer in Hyderabad who can bridge it to a 32-port Dinstar GSM gateway sitting in our office. Your task is strictly configuration and testing—no fresh installs. Scope of work • Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs before si...

Vicidial is already live on our cloud server; what I need now is an on-site engineer in Hyderabad who can bridge it to a 32-port Dinstar GSM gateway sitting in our office. Your task is strictly configuration and testing—no fresh installs. Scope of work • Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs before si...

...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 number: Calls should route to a voicemail/mailbox system. From within the mailbox, it must be possible to enter a PIN code and access an administration menu. From this menu, the user should be able to access a D...

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...

I’m creating a pair of insulated flasks—32 oz and 64 oz—that must keep drinks hot or cold for a full 24 hours. The body will be molded from BPA-free plastic rather than stainless steel so overall weight stays low without sacrificing durability. Key functional goals • A changeable sip head built around a Flip-top lid (no straw). If you can engineer the interface so that a future spout or twist-off cap could swap in without redesigning the whole neck, that would be a plus. • Triple anti-leak protection: I’m picturing a combination of high-tolerance threading, food-grade silicone gaskets, and a secondary internal seal, but I’m open to smarter ideas. • True light-weighting: target mass should beat comparable stainless bottles by at l...

... Machine A → Media Processing Unit (GPU server for STT + TTS + SIP + WebRTC) Voice and chat must share the same AI brain. We require a developer who can build a low-latency (<1 second), GPU-optimized, production-ready system. This is NOT an API wrapper project. This requires real-time streaming AI experience. Infrastructure (Already Available) Machine A RTX 5060 Ti 16GB Proxmox 8.4 Docker running directly on host (NO GPU passthrough via VM) NVIDIA Container Toolkit access Machine B Existing chatbot backend Knowledge base (site-wise) CRM integration Order status APIs Existing React frontend (MUST NOT be modified) Project Scope Media Processing Layer (Machine A) You will build: Audio Orchestrator Handle SIP calls Handle WebRTC / WebSocket brows...

I need my new 3CX phone system connected to a reliable business-grade SIP trunk that will support a blended contact-center operation — both inbound customer service and outbound campaigns. The trunk must be fully configured, tested, and ready for live traffic. Core requirements • End-to-end SIP trunk provisioning on 3CX, including carrier registration, codecs, DID mapping, and secure transport (TLS/SRTP if offered by the provider). • IVR setup for our first menu tree so callers can self-route before reaching an agent. • Validation that inbound, outbound, and internal transfers all behave correctly with call quality, caller-ID and forwarding intact. • Written handover notes covering the trunk settings, any firewall/NAT rules applied, and simp...

I am looking for a senior Vicidial / Asterisk Expert to perform a clean installation and professional optimizati...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. • Security: Full security hardening (White-list IPs, Firewall, and protection against SIP attacks). • Optimization: Fine-tuning of MySQL, Asterisk logging, and Crontab for high-load performance. Skills Required: • Expert level Vicidial / ViciBox knowledge. • Advanced Asterisk Dialplan & Queue management. • Linux (OpenSuSE) server optimization. • VoIP / SIP Trunk troubleshooting. Note: Please provide examples of previous Vicidial deployments you have ...

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/merg...

I have a fresh FusionPBX install that i need help to register with my SIP provider. I have not yet filled in any trunk settings because I am unsure which parameters the carrier needs and where each item belongs inside FusionPBX. I am looking for someone who can: • Tell me exactly which credentials and network details to request from the provider (user ID, auth name, password, proxy, outbound proxy, codecs, etc.). • Remotely configure the trunk in FusionPBX once those details are in hand. • Prove the registration is solid and that I can place and receive at least one test call without errors. Please work directly in the FusionPBX web interface (FreeSWITCH‐based) and keep a quick note of every change you make so I can replicate it later if needed. Once regis...

I need my Windows-based calling app wired up to 3CX and Twilio so both inbound and outbound calls flow without glitches. The work is a mix of fresh setup—provisioning the Twilio SIP trunk, routing numbers, configuring 3CX—and targeted troubleshooting on what’s already in place. A few settings have been touched, so you’ll likely spot and correct any mis-configured codecs, authentication details, or firewall rules. If this sounds straightforward for you, let’s schedule a quick session to outline next steps and timelines.

...version of VICIdial, optimise the server environment, integrate reliable SIP trunks that allow CLI override, and verify everything with live-call tests. Server and hosting I do not yet have 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 integrati...

...development
 - Experience with `mod_audio_fork` and `mod_audio_stream`
 - Deep understanding of SIP/RTP/media flows What you will do:
 - Connect our existing Freeswitch server with Elevenlab's WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured. 
 Enable seamless, real-time, bi-directional audio between the caller and the voice agent
. Stream audio to Elevenlab in real-time and handle incoming transcription/command messages. Maintain high availability and low latency across multiple concurrent sessions. Ensure voice agent can:
 - Execute in-call commands like:
 - End the call
 - Transfer the call to a human agent
 - Trigger DTMF or SIP-based routing actions
 - Play custom messages or handle c...

Vicidial is already live on our cloud server; what I need now is an on-site engineer in Hyderabad who can bridge it to a 32-port Dinstar GSM gateway sitting in our office. Your task is strictly configuration and testing—no fresh installs. Scope of work • Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs bef...

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...

...only has to converse in clear, natural English, but I want the architecture kept language-agnostic so we can add Spanish and French later without rebuilding core logic. Key requirements • 24 / 7 availability with no noticeable downtime. • Seamless, secure integration with RoomRaccoon via its API (authentication, error handling, rate limits). • Works over standard phone lines; if you prefer SIP, Twilio, or another VoIP layer, outline that in your approach. • GDPR-compliant handling of personal data and call recordings. • Dashboard or logs that let my staff review conversations, monitor performance and tweak responses without coding. Deliverables 1. A fully configured, cloud-hosted voice agent connected to our RoomRaccoon test account. 2. So...