J2me softphone jobs

Project Title: Full Issabel 5 PBX Deployment: Installation, Trunks & Extensions Project Overview I am seeking a VoIP specialist to perform a complete installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architectu...credentials. Freelancer Requirements Expertise in Asterisk/Issabel/FreePBX configuration. Prior experience setting up SIP Trunks (e.g., Twilio, Telnyx, ). Solid understanding of NAT traversal and firewall management for cloud-hosted VoIP. Deliverables A fully functional Issabel PBX on AlmaLinux 8. Configured and tested SIP Trunk for inbound/outbound calls. Set of active extensions ready for softphone or desk phone registration. Documentation of all credentials and security se...

I don't have FusionPBX installed on any server yet. You will install and setup fusionpbx on a server I will provide 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 on...

...environments running smoothly while I focus 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 e...

Recruitment Robin is a values-driven, independent...data accuracy. What we’re offering: • Flexible, remote working arrangement. • Ongoing, consistent workload for the right candidate. • Opportunity to grow with a developing independent consultancy. • Clear expectations and structured processes. • Supportive and collaborative working relationship. • Performance-based commission or retained fee structure (to be agreed). • Access to company VOIP/softphone system and branded email address. To apply: Please include: • A brief summary of your recruitment experience • The sectors you have worked within • Your availability (hours per week) • Your project rate We are looking to appoint promptly and welcome applications from e...

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,...

Our customer-s...queue, captures voicemail after hours, and logs every interaction for reporting. Here’s what I need from you: • Account configuration: numbers, call flows, business hours, holiday rules, call recording and analytics activated. • IVR design & build: concise English prompts (I can provide scripts or you can polish them), multi-level menu if required, zero dead ends. • Agent setup: user roles, softphone/mobile configuration, and quick training so my team can pick up calls from day one. • Testing & hand-over: run test scenarios with me, refine anything that isn’t smooth, then document the final flow. The project is complete when incoming calls reach the right agent without delay, reports show accurate call data, and my sta...

...questions at the bottom of this posting or it will be immediately rejected. We're a UK-based eSIM and mobile network company launching a branded international calling service. We need an experienced developer to fork, rebrand, and customise the WebTrit Phone open-source softphone app and get it published to both app stores. IMPORTANT: Read this entire posting carefully. There are specific questions at the bottom you MUST answer. Generic proposals will be rejected immediately. What is WebTrit? WebTrit Phone is a Flutter/Dart softphone app that uses WebRTC for voice and video calling. It connects to SIP-based VoIP systems via a REST API. The full source code is available on GitHub. What We Need Done Phase 1 — Fork & Rebrand Fork the WebTrit Phone repo into o...

...общаетесь с клиентами и быстро обучаетесь. Мы работаем удалённо и по гибкому графику: можете брать полный восьмичасовой слот или делить его на смены — оплата почасовая либо за смену, по договорённости. Связаться со мной можно здесь в личных сообщениях; расскажите коротко о своём опыте, времени, когда готовы выходить на линию, и о технических средствах, через которые планируете принимать звонки (Softphone, SIP-клиент, мобильный и т. д.). Если вам комфортно вести разговор и на английском, и на русском, готовы проявлять инициативу и соблюдать порядок в заявках, буду рад сотрудничеству!

Looking for someone who can do basic customization of Linphone Windows app. If someone has already done liphone or any other windows softphone app, please message me directly. please don't do if you don't have relevant experience.

...live phone calls (both inbound and outbound) using the ElevenLabs Speech-to-Speech API. What We Need: The app should capture microphone input, stream it to the ElevenLabs voice changer API, and route the transformed audio back through a virtual audio device (like VB-Cable or VoiceMeeter) so it can be used as a microphone input for any calling app — Zoom, Google Voice, Teams, RingCentral, or any softphone. Key Requirements: Real-time voice transformation with minimal latency (target sub-500ms) Works with any phone/calling application via virtual audio device Supports both inbound and outbound calls Simple desktop GUI with voice selection, on/off toggle, and audio level meters Ability to browse and switch between ElevenLabs voices Background noise removal support Hotkey to ...

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)...

...Spanish-speaking customers. The bulk of each conversation centers on service scheduling and bookings, so you’ll spend most of your shift walking callers through available dates, entering jobs into our cloud booking system, and confirming addresses inside our CRM. Clear, upbeat communication on the first ring is critical to maintaining our local reputation. You’ll work remotely, logging in through our VoIP softphone; a stable internet connection and a quiet workspace are therefore must-haves. I supply call scripts, service menus, and step-by-step scheduling guides—you supply the professionalism and fluency to make every customer feel at ease. When you apply, include brief examples of past phone-based customer service work that highlight your bilingual skills an...

I’m running a production web soft-phone that relies on pure JavaScript and JsSIP. Incoming SIP INVITEs reach the browser, but the UI keeps the “Answer” button disabled, leaving me unable to accept the call at all. I’m looking for someone who knows JsSIP, WebRTC and SIP signalling inside out to jump in, trace the handshake, and get the call flow moving from “ringing” to “in-call”. Wireshark, browser dev-tools, JsSIP debug logs—use whatever helps; I simply need the root cause found and fixed fast. Deliverables • Patch or configuration update that re-enables the Answer action and completes a full two-way call in my staging environment. • Short write-up of what was wrong and how you corrected it so I can maintain the change...

I’m looking for a Spanish-speaking professional who can step in as th...escalation guidelines I will provide. • Keep average handle time low while maintaining high-quality, empathetic service. Key expectations for acceptance: 1. Native-level Spanish with confident, friendly phone presence. 2. Proven track record in customer service—experience with retention or sales upsell is a plus. 3. Dependable internet connection and a quiet workspace; you’ll use our VoIP softphone. 4. After each shift, submit a call summary report so I can track resolutions and any follow-up tasks. If you’re ready to manage callers’ issues end-to-end and make every interaction count, send a brief note on your relevant experience and your typical availability. Let&rs...

Single Node (All-in-One) 2600Hz Kazoo infrastructure on a Proxmox virtualization environment. Important Note: I do NOT need a massive, distributed cluster. The goal is to have a compact, standalone system where all components run on a single Virtual Machine for R&D and commercial testing purposes. The project also includes the configuration of WebRTC and the preparation of a White-Label Mobile Softphone (based on Linphone) fully integrated with this system. Scope of Work: 1. Single Node Kazoo Infrastructure (Server-Side): All-in-One Architecture: Installation of Kazoo, FreeSWITCH, Kamailio, RabbitMQ, BigCouch/CouchDB, and RTPEngine on a single VM within Proxmox. Network & NAT Traversal: Proper configuration of IP, Ports, and ACLs to ensure seamless RTP/Signaling flow be...

Single Node (All-in-One) 2600Hz Kazoo infrastructure on a Proxmox virtualization environment. Important Note: I do NOT need a massive, distributed cluster. The goal is to have a compact, standalone system where all components run on a single Virtual Machine for R&D and commercial testing purposes. The project also includes the configuration of WebRTC and the preparation of a White-Label Mobile Softphone (based on Linphone) fully integrated with this system. Scope of Work: 1. Single Node Kazoo Infrastructure (Server-Side): All-in-One Architecture: Installation of Kazoo, FreeSWITCH, Kamailio, RabbitMQ, BigCouch/CouchDB, and RTPEngine on a single VM within Proxmox. Network & NAT Traversal: Proper configuration of IP, Ports, and ACLs to ensure seamless RTP/Signaling flow be...

...handles VoIP calls. ### Requirements * SIP REGISTER / INVITE / ANSWER / BYE * Digest authentication * SDP negotiation * RTP transport * **Opus codec mandatory** * UDP SIP (TCP optional) * Configurable SIP credentials * Clean reconnection logic ### Acceptance Criteria ESP32 registers and stays registered Can place outgoing SIP calls Can receive incoming SIP calls Two-way audio works with a SIP softphone Stable for **10+ minutes** # PART 3 — Audio Transport Bridge ### Objective Real-time bridge between USB PCM audio and SIP RTP audio. ### Required Pipeline USB Speaker PCM → Opus Encode → RTP Send RTP Receive → Opus Decode → USB Mic PCM ```### Architecture Requirements * Separate FreeRTOS tasks: * USB RX * RTP TX * RTP RX * USB TX * Ring ...

...this node can exchange dialog state with our existing Kamailio proxies • Full ws / wss support with correct TLS setup • SDP mangling and ICE/TURN handling so JsSIP, SIPml5 and clients connect without manual tweaks (our cloud PBX front-end uses those libraries today) • Proven interop with Odoo VoIP’s JsSIP webphone Nice-to-have If you already have a lightweight HTML/JS WebRTC softphone that exposes simple embed hooks (iframe, JS API or similar), I’d like to see it; the closer it drops into our current CloudPBX UI the better. Deliverables 1. Annotated , TLS materials and startup scripts 2. DMQ peer configuration tested with another Kamailio instance 3. Sample Nginx/Apache reverse-proxy snippet for wss termination (if used) 4. README t...

...understands RingCentral’s admin console to tune our small phone system so it works the way my staff expects. The goal is simple: create three shared Call Park locations and make sure every user can see at a glance which of those parking spots—and which of our 6–10 extensions—are free or busy. We take both internal and external calls, and our users bounce between desk phones and the RingCentral softphone apps, so presence status must stay perfectly in sync across all devices. Once you’re done, any team member should be able to press a single button to park, retrieve, or monitor calls without hunting through menus. Deliverables • Three functioning shared Call Park locations, visible system-wide • Presence monitoring enabled for all ex...

I need Twilio hooked up to a Windows-based desktop softphone so I can handle both voice calls and SMS from the same workstation. The softphone itself is already installed; what’s missing is the full Twilio connection—SIP domain, credentials, messaging service, and all the event handling that goes with it. You’ll handle the Twilio Console setup, generate the proper tokens, and map everything into the client app so inbound and outbound voice calls register cleanly and SMS messages send and receive without delay. I’d like clear documentation on every step you take so the configuration can be replicated on additional machines later. Deliverables • Working Windows desktop softphone registered to Twilio, verified for inbound/outbound voic...

...opens the correct firewall ports, and restarts the services. • Include concise notes on how to renew or regenerate the certificates and how to add extra users/extensions in the future. Acceptance I will consider the job complete when I can load the demo page in Chrome/Firefox, register a test extension, and successfully place audio, video, and chat sessions between two browser tabs or a softphone and the browser. Please be comfortable troubleshooting Asterisk 20, PJSIP, and WebRTC quirks on Ubuntu before bidding....

I already have an in-house VoIP solution in progress—a SIP server built with ASP.NET /C# and a cross-platform softphone written in .NET MAUI. The core functionality is there, but it is not yet production-ready. I need someone who can step in, review the existing codebase, and see the whole project through to a stable, polished release. Key objectives • Resolve all stability and performance issues so calls remain reliable under load and the server maintains low-latency connections. • Refactor and streamline the code for efficiency and long-term maintainability. • Refresh the mobile UI with improved navigation, a modern design, and effortless responsiveness across iOS and Android form factors. • Add the extra features and workflow enhancements ...

...and reply in your own words. I’m a PHP developer building a system to connect FreeSWITCH (Debian 12) with an external PHP/MySQL API using mod_xml_curl. I’ve already: Installed FreeSWITCH on a Debian 12 VPS Created a MySQL table called sip_accounts with username and password fields Built a PHP API that returns FreeSWITCH-compatible XML from that database Right now, SIP registration from my softphone is failing. The FreeSWITCH CLI shows XML-related errors, so there’s likely a configuration issue. I’m looking for someone who has full FreeSWITCH configuration knowledge and strong Linux experience to help me fix this setup. You don’t need to work on PHP or MySQL — I’ll handle that part. Your job is to: Guide me through the correct Fre...

...prepaid/postpaid accounts, monthly invoices, payment gateway integration (Stripe/PayU). API Access – REST APIs for provisioning, user management, and call data retrieval. Branding Options – Each tenant can upload their logo and customize their portal theme/colors. Security & Monitoring – TLS/SRTP, fail2ban, alerts for trunk status, call volume, and resource usage. Optional Add-ons – WebRTC softphone, CRM integrations (Zoho, HubSpot, Salesforce), or SMS gateway. Technical Requirements: Strong experience in Asterisk or FreeSWITCH (dialplans, SIP, RTP, codecs, NAT, SRTP, WebRTC). Familiar with FusionPBX, FreePBX, or ASTPP is a big plus. Backend: PHP (Laravel) / Node.js / Python (Django or FastAPI). Frontend: React.js or Vue.js (modern, responsive...

...tables/schemas Configure FreeSWITCH to read and write everything from the database Add Lua dialplan scripts so routing logic runs from Lua and database references Set up two working routes: One SIP trunk DID (inbound/outbound) One IP-auth DID (no registration, direct routing) Provide clear SQL examples or docs so I can easily add new users, DIDs, and routes later Acceptance: I can log in with a softphone Inbound/outbound calls work for both DIDs No XML configs used — only MariaDB...

Our company is ready to move to a full VoIP solution. I already have bandwidth in place; what I need is an end-to-end setup that lets every employee work from a desk phone, a mobile app, or a softphone without noticing a difference in call quality. Looking for an expert to recommend what the best route to accomplish this would be. Cloud based vendors or a standalone system that we integrate ourselves. Here’s what I want working smoothly from day one: • Full auto attendant first response system • Full call recording • Routing system to get calls routed to the right personnel • Call forwarding that lets each user hop between devices without dropping the call • Voicemail to email so messages arrive as audio files in Outlook and Gmail • C...

I already hold an active account but it is still sitting on its default settings. I need someone who can walk in, turn the blank account into a working phone line, and leave me with a fully-tested softphone on my Windows PC. Here’s what I expect: • Configure the portal—create the SIP trunk, assign a DID, set up caller-ID, inbound and outbound routes, and the usual codecs/security options. • Recommend a reliable Windows softphone (Zoiper, 3CX, or any better option) and install or remotely guide me through installation. • Register the softphone to the credentials you created, confirm two-way audio, DTMF, and voicemail. • Provide a short hand-off document or screen-share recording so I can replicate the setup or edit it later. If you ...

...(for media translation between WebRTC and RTP) coturn (for STUN/TURN and NAT traversal) Nginx + SSL (for secure WSS/HTTPS access) Develop a simple web softphone (using JsSIP or similar) with the following features: SIP login and registration Make and receive calls in the browser Mute and unmute functionality Music on Hold (MOH) option DTMF tone support during active calls Display of call status and connection state Ensure that all calls are properly routed through MagnusBilling for: Real-time billing updates Call detail records (CDR) Correct balance management Deliverables: Fully working WebRTC ↔ SIP gateway Web softphone interface with Mute / MOH / DTMF support Documentation covering main configuration files and installation steps Live test confirmi...

...Phones register to same hostname as cert; SRTP mandatory; compatible crypto suites Security (keep as-is) Don’t re-open UDP 5060 Keep Fail2Ban jail for FS auth and provider ACLs Zero Trust stays on GUI Handover Short runbook: trunks, dialplan regex, variables (sip_from_user, from-domain, etc.), how to add carriers/DIDs Test commands & how to capture logs Acceptance Outbound from a TLS/SRTP softphone to two international numbers succeeds via DIDWW Outbound via IDT Express succeeds with valid CLI Inbound via DIDWW reaches the target destination Carrier no longer sees trunk username as caller ID Dialplan shows the outbound rule chosen and correct normalization Security posture unchanged Required Experience FusionPBX/FreeSWITCH (sofia profiles, gateways, dial...

1. Platform • Android 9 2. Core Functionality • The app must accept incoming SIP calls from my Asterisk server. • Upon receiving a SIP call: 1. The app does not answer the SIP call immediately. 2. It first initiates a GSM call to a number , this number can be obtained from incoming sip call information . 3. Only after the GSM call is answered, the app answers the SIP call. 4. At that moment, the audio streams are bridged: • SIP RTP ↔ GSM audio path. • If the GSM call is not answered or fails, the SIP call must be rejected or continue ringing (not answered silently). 3. Audio Handling • Full two-way audio bridging between SIP (RTP) and GSM telephony path. • Must work when: • Screen is off. • Phone is locked. • App is in background. &...

We are a logistics company working with German clients and looking for a bilingual freelancer (German + English) to handle inbound calls and provide customer support. You will represent our comp...company image. Requirements: German level B2–C1 (spoken and written). English level B1+ (for internal communication). Previous experience in customer service or logistics is a plus. Availability during CET working hours (09:00–15:00, Monday–Friday). Average workload: 4–5 calls per day. Reliable internet connection and a quiet workspace. Ability to work with tools like: IP telephony, Softphone, Telegram/WhatsApp, Excel/Google Sheets. Conditions: Remote freelance work (part-time). Trial period: 1–2 weeks. Payment: €15/day for availability +...

...mobile handsets to worry about—just a clean, streamlined soft-client environment. The core objective is rock-solid call quality to and from European numbers. I would like you to: • Complete the initial 3CX hosted deployment and apply best-practice security settings. • Connect and configure a SIP trunk (or advise on the most reliable provider) optimized for European routes. • Provision the softphone clients for every user, including extension creation, codec selection, and BLF/shortcut configuration where helpful. • Run end-to-end tests—outbound, inbound, and transfer—confirming clear audio and stable connectivity. • Hand over concise documentation so I can manage day-to-day tasks such as adding new users or updating greetings. ...

I’m looking for a Hindi-speaking caller who can become the friendly voice of my brand. The core of the role is straightforwa...Resolve general customer questions or concerns on the spot whenever possible, then escalate anything outside your scope to me. • Keep short, accurate call notes in the shared CRM so we always know the status of each customer. • Provide a concise end-of-day report summarizing the number of calls handled and any issues that need follow-up. You may use whatever setup is most comfortable—mobile, softphone, or a VoIP platform—as long as the audio is clear for the caller and you have a stable connection. If you’re friendly on the phone, patient with people, and ready to start right away, let’s talk so we can arrange a ...

I need a freelancer to white-label the Linphone open-source softphone into a branded mobile app for my service Unlimited World. Requirements: Replace branding: app name, logo, colors, and splash screen. Deliver Android + iOS builds (ready for Play Store & App Store). Integrate SIP login with our own PBX + GoIP gateways. Each customer will be assigned their own SIM hosted in our GoIP. SIP username = customer’s local phone number. Support inbound & outbound calls through the customer’s own SIM (via GoIP). Show call history (made, received, missed) inside the app. Allow us to assign calls to agents (so agents can answer on behalf of customers) and show missed call alerts. What We Expect: Flat setup cost (no per-user/month fees). Quick delivery — we...

Project Description : I successfully completed a project involving the installation, configuration, and integration of FreePBX with multiple softphone clients (such as MicroSIP and mizudroid) for a hospital communication system. Key Tasks Completed: • Installed and configured FreePBX server. • Created and managed SIP extensions for multiple users. • Integrated softphones (MicroSIP/Zoiper) on Windows PCs and mobile devices for seamless communication. • Configured internal calling between departments (e.g., OPD, Emergency, Pharmacy, Wards). • Implemented call routing rules and voicemail setup. • Tested and optimized call quality & stability. Technologies Used: • FreePBX (Asterisk) • MicroSIP / Zoiper softphones • SIP Protocol • L...

I'm seeking a skilled developer to troubleshoot and resolve issues with our custom ...Familiarity with mobile app backend services Your tasks will include: - Diagnosing the issues with the Create Account and Login APIs - Implementing fixes to ensure expected behavior - Testing the APIs to verify the issues are resolved Issue which we are facing - - When a new SIP account is created via our custom API, the user can log in, but the SIP account does not register on the mobile softphone/app. - However, when the same account is created directly from the Magnus Billing admin panel, the SIP registration works perfectly. - Once an account is created from the panel, all accounts (including those created via API) start working fine. It should be work for both as same from mobile api a...

I need a comprehensive FreePBX setup tailored for both inbound and outbound calls. The configuration should include: - Call Routing Setup - Extension Management - IVR (Interactive Voice Response) - Trunk Configuration - Inbound/Outbound Calling Setup Essential Integrations: - TG100 Gateway Locally hosted - Zoiper Softphone - Cloud FreePBX Ideal Skills and Experience: - Proficiency in FreePBX configuration - Experience with TG100 gateways and Zoiper - Strong understanding of VoIP systems and call management Looking forward to your expertise in getting this system operational.

I'm seeking a South Africa-based appointment setter to assist our marketing agency in booking property appraisals for Australian real estate agents. Key Responsibilities: - Call warm leads that ...Represent various agents and adapt to their brands - Book home appraisal appointments Requirements: - Excellent English & confident phone manner - Adaptability to different clients’ styles - Reliable internet, quiet workspace, and headset - Availability during Australian business hours Ideal Skills & Experience: - Prior experience in real estate or appointment setting - Familiarity with VoIP or softphone applications - Strong communication and organizational skills Your ability to confidently connect with clients and efficiently manage multiple agents' schedules...

I need a PBX where we only need SIP and IAX extensions to be registered and reachable from our Dialer. The extensions should not be able to make calls, but rather they accept invites from our Dialer and listen into calls. Call flow: Agent registers his softphone (Zoiper) by using credentials of FreePBX (username, PW, domain) Dialer calls the extension and Agent answers the call on Zoiper. This way the agent gets connected to the Dialer and the dialer makes outbound calls. Your job is only to make sure that FreePBX is installed and extensions are imported. Also some security measurements (different Ports for Portal and SIP) and activate Fail2Ban and some other security steps.

...endpoint are already functional. We also have a dashboard UI and a working example of our API, which you will be expected to adapt and extend. This is a focused integration task to bridge SIP audio with our AI infrastructure. ⸻ What We Need: 1. SIP Gateway Setup • Use Asterisk or FreeSWITCH • Handle both: • Inbound SIP calls from trunk or softphone • Outbound SIP calls triggered via REST API • Provide basic configuration for testing (e.g. softphone, SIP trunk) 2. WebSocket Audio Bridge • On call connect: • Capture live RTP audio and stream it to our WebSocket endpoint (/ws/audio) using the openai-audio subprotocol • Receive TTS audio responses and inject them back into the SIP call • Handle call lifecycle: connect,...

...info on call connect - Integrate with CRM (GoHighLevel / LeadConnector) and Zapier - Support call recording, live supervision (listen, whisper, barge), and basic compliance tools 4. Required Features Category Requirements Telephony Twilio BYOC support, SIP trunk config, IVR routing, call queuing, fallback routing Dialer Predictive and preview dialing, auto-pause, lead recycling Agent Tools Web softphone, screen pop, call logging, warm/blind internal transfers, wrap-up codes Monitoring Call monitoring, whisper, barge, live agent dashboards CRM Integration Zapier and/or API integration with GoHighLevel (LeadConnector) Compliance Call recording, consent logging, STIR/SHAKEN, TCPA, DNC tools, local presence Reporting Agent activity logs, call outcome reports, exportable dashboards S...

...**Netherlands Mobile Numbers** * Provision of real Netherlands mobile numbers (not landlines) that reflect our business presence in the Netherlands. * These numbers must support both inbound and outbound calling and SMS functionality. * Nice-to-have: if feasible, the ability to port over existing NL mobile numbers; otherwise, new numbers can be issued. * **Mobile Softphone App Support** * The phone numbers must work through a softphone app on both iOS and Android. * The app should support calling and SMS, with excellent call quality and reliability. * **Zoho CRM Web Calling** * Team members must be able to initiate calls directly from within the Zoho CRM web interface using click-to-call functionality. * The experience must be seamless and not rely on ex...

Hi! I’m looking for an experienced (and cost-effective) Zoho CRM expert to help set up and customize our system. Our current setup: We use Signalwire for our phone numbers. We have an answering service that handles calls and collects lead details. We'd like to use Hellosend as our softphone solution. What I need: Complete Zoho CRM setup and configuration. Integration of Zoho CRM with our answering service’s lead data. Softphone integration so I can call and text from my numbers directly in Zoho. Advice and help porting numbers from Signalwire to Twilio (if needed) — or your suggestions for the best way to make this work. Assistance with A2P messaging compliance/setup if required. UI and layout customization in Zoho to match very specific require...

***PDF description added - please read first before submitting*** I'm seeking a skilled team to build a custom Unified Communications platform to compete with 3CX. Key components include: - Telephony engine - Modern mobile and PC apps with: - PUSH notifications support - Custom tunnel connection - Comprehensive web application for users featuring: - Presence - Contacts - Voicemail - Softphone - Intuitive web UI for administration - RESTful API's to be added for realtime calls, and another for conifguration. - Licensing system - Packaged into a custom Debian image for fast deployment - Development pipeline setup Essentially, look at all the features 3CX has - we want all (well almost all) of them. Ideal candidates should have: - Proven experience in voice a...

...inbound routing per number Set up outbound routing and dial plans (e.g., number X always routes via trunk Y) Time-based call routing (based on day & hour) Store voicemails per number in separate server folders Enable push notifications for mobile apps Implement logging and basic call reporting Call Flow Logic Per number: unique greeting + voicemail Call flow: → IP Phone (Aastra 5370) → Softphone on PC (Zoiper or similar) → Mobile device → Voicemail (stored as .wav/.mp3 per number) Manual fallback routing toggle if needed Client Device Setup Assist in configuring softphones on: 1 Windows PC 1 Linux PC 2 smartphones (Android + iPhone) Configure push notifications and codecs properly Voicemail Handling Archive voicemails after 30 days (Optio...

...development (Node.js, Python, PHP, or similar) • Experience with mobile app integration, preferably for iOS/Android softphone apps • Knowledge of VoIP provisioning and configuration management ⸻ Nice to Have: • Experience working with other softphone solutions • Familiarity with telecom protocols (SIP, SDP, STUN/TURN/ICE) • Prior work with SMS gateways or telecom messaging APIs ⸻ Budget: We are a small team with limited resources but committed to quality. Please include your best rate and a rough timeline estimate in your proposal. ⸻ To Apply: Please send: 1. A short introduction of your experience with similar projects 2. Examples or case studies of previous VoIP or softphone API integrations 3. Your estimated budget and ava...

Use drachtio-srf to create a VoIP gateway via WS/WSS. The Node application should allow full communication like a common softphone, for example: MicroSIP. Remembering that the call must happen at the PSTN level. Using these technologies and libraries specifically is mandatory.

i have installed issable pbx voip system i was get a tls sip tunk and it is working good on any softphone i just need you register it on system issable pbx

I need an remote administrative assistant to help keep up with work orders and project details in two internal systems. This is a PRN (as needed) and ongoing project. Key Responsibilities: - Updating project details from emails or communication from...Willingness to learn internal systems (2 custom built web-based applications) - Proficiency in outlook and teams (for communication) - Experience with project management Pay will be by hour, kept up with timekeeping. No guaranteed hours, but the more you learn, the more I can rely on this support position. Once we establish a good working relationship, I will provide email access, teams access, and a softphone for calls (as needed). This role requires attention to detail, organizational skills, and the ability to manage tasks ef...

...per lead using: Base CF rate x CF + surcharges (fuel, stairs, etc) Buyer-specific pricing profiles Send estimate to buyer/lead via SMS or email PHASE 5: DIALER + TRANSFER SYSTEM 12. Inbound + Outbound Dialer Click-to-call from lead profile Progressive dialer for queues Voicemail drop + auto-logging Call status buttons: No Answer, Booked, VM, Not Interested WebRTC browser calling (or softphone option) 13. Transfer System Warm transfer button from dialer Call bridge: Rep > Buyer Whisper message before buyer answers Logs all transfers, time, and outcome Optional: AI qualifies and transfers calls (Phase 6) 14. Call Center Dashboard (Optional Add-on) Live view of agent calls Queue status Conversion and contact rate PHASE 6: OPTIONAL ADD-ONS Stripe billing + bu...