SIP Jobs

I want to deploy a voice-driven concierge that guests can call at any hour to handle day-to-day requests without waiting for the front desk. The core functions I need right now are: • Answer room-availability questions in real time and complete the booking, reading and writing data directly to our RoomRaccoon PMS. • Take amenity reservations (spa, restaurant, late check-out), create or modify the corresponding records in RoomRaccoon, and send a confirmation SMS or email to the guest. • Provide concise answers about nearby attractions, directions and opening hours. • Forward room-service orders to the appropriate in-house system or by email/API if that is all that’s available. • Escalate anything it cannot solve to a live agent, handing over the call ...

We have an Android mobile dialer application that is set as the default phone app. The app supports: GSM/SIM-based calling SIP/VoIP calling We need to implement a feature that allows conference calling between: One GSM call One SIP call Requirement When a GSM call is active and a SIP call is active, the user should be able to merge them into a single conference call so that all participants can talk together. Technical Expectations Experience with Android Telecom framework Experience working with: Connection Service In Call Service SIP stack (PJSIP or Android SIP API) Handling audio routing between GSM and SIP Managing call states and audio focus Proper call merge / conference implementation Clean and stable solution Important Notes The app is already functioning for individual GSM and ...

I need real-time, screen-share guidance on Google Meet to get a brand-new Twilio Elastic SIP Trunk talking cleanly with my existing 3CX system so we can place and receive phone calls. My Twilio account is a blank slate, and the main hurdle is the integration itself—both inbound and outbound routing must work by the end of the session. Here is the flow I have in mind: we start inside Twilio, build the trunk from scratch, add the authentication details, assign a DID, and verify the voice routing. From there we’ll jump into the 3CX management console, create the corresponding SIP trunk, map the numbers, set caller ID rules, and tweak codecs or transport settings if required. Once registration is solid we will run live test calls in both directions to confirm audio quality and sig...

Project Description: We are looking to develop a VoIP calling application with direct phone-to-phone/mobile calling and the ability to use a custom caller ID. The project will include both an Android user app and a web-based admin portal for managing users and call data. Key Features: Admin Portal: User Management – Add, remove, and manage users. Custom Caller ID – Assign or change custom caller IDs for users. User Details Management – Manage name, email, user ID, password visibility, and assigned caller ID. Call Logs – Save and display calling history with IP addresses, calling number, time, and duration. Admin Control – Only admin can add new users and modify user details. User App (Android): Login – Users log in with email ID and password. D...

I need an IPRN switch brought online fast, fully wired for billing 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, ...

We are building a voice AI assistant that can automatically talk to customers over phone calls and handle real business conversations in multiple Indian languages (Hindi, English, and other regional languages). The system should: • Receive incoming customer calls • Answer queries (fees, services, timings, status, due payments, support requests) • Make automated outbound calls (appointment confirmations, reminders, lead follow-ups, collections) • Transfer the call to a human agent when required This is NOT a keypad IVR. The assistant must understand natural spoken language from callers and respond with a natural-sounding voice. Technical Scope: • SIP/VoIP integration using Asterisk or FreeSWITCH • Real-time Speech-to-Text • Text-to-Speech voice response...

We are looking for an experienced React Native developer to help build and integrate a VoIP calling SDK into an existing mobile application. This is not a basic app development task. We need someone who has real experience with VoIP / WebRTC / SIP / SDK-level work and can deliver quickly. ### Project Scope - Integrate and extend a React Native VoIP SDK - WebRTC calling using - Inbound & outbound call handling - Push notification based incoming calls - Background call handling - App-based configuration dialing logic - SDK-style packaging + documentation support Our team will handle: - OpenSIPS / SBC (Call logic) - SIP infrastructure - APIs You will handle: - React Native SDK layer - Mobile VoIP calling integration - Native module bridging if required - Call UI handling - Stabili...

I need a Japanese national DID or any other VOIP number that can reliably forward calls to my chosen destination. The line will serve both personal and business purposes, so stability, clear audio and the ability to register the caller ID on common soft-phone or SIP devices is essential. Please supply, activate and demonstrate the number working via simple call-forwarding within the shortest possible time; my preference is ASAP. If you already have an inventory of Japanese DIDs, even better—let me know the formats available and any documentation requirements. Acceptance is straightforward: once I can receive and place test calls through the number without drops or quality issues, the milestone is cleared.

I need an experienced VoIP specialist to configure my on-premises Polycom phones for use with RingCentral. Key requirements: - Configure SIP settings, network settings, and user extensions specifically for RingCentral integration. - Ensure all phones are fully operational and can seamlessly connect to RingCentral services. Ideal skills and experience: - Expertise in configuring Polycom VoIP phones. - Proficiency with RingCentral and understanding of its specific configuration requirements. - Strong networking knowledge to handle any required network settings adjustments. - Prior experience with on-premises VoIP systems is a plus.