SIP Jobs

To develop a VoIP calling app direct calling phone/mobile with custom caller ID Admin portal- 1. User management 2. Custom caller ID 3. Calling history with IP 4. Manage User details, Name, Email, User ID, Showing password, Caller ID with have change custom caller ID 5. Save Calling details with IP, calling number, time etc 6. Manage User by Admin (Add & remove User) 7. Admin portal development 8. User Android App with Login Email id & Password 9. Admin portal- only add new user by Admin,

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...

I need to link one incoming GSM leg with a SIP participant inside FreeSWITCH so both callers join the same live conference room without noticing any difference in media or signaling. The end result must feel like a single, seamless conversation. Key requirements • FreeSWITCH must act as the bridge and mixer. • The conference room has to support call recording (start automatically and save to disk) and allow me to mute or un-mute either participant from the console or an API. Current environment – A working GSM gateway already delivers calls to FreeSWITCH through SIP. – An existing SIP trunk is in place for the VoIP side. – Root access to the FreeSWITCH server is available. Deliverable 1. Step-by-step configuration (dial-plan snippets, conference profi...

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.

I need a proven Airtel Black SIP configuration running on my Ubuntu machine and handled through Linphone. The goal is a clean, documented initial setup—no trial-and-error learning on my system, please. If you have already registered an Airtel SIP trunk on Linux, you’ll know the exact registrar format, the unusual port mapping Airtel uses, and the little tweaks that keep audio flowing both ways behind NAT. Here’s the workflow I’m expecting: • Install or verify the latest Linphone and any required dependencies on my Ubuntu box. • Register the Airtel Black SIP account, applying the correct proxy, authentication string, and codec priorities. • Prove the setup with at least one inbound and one outbound call (I can join you on a test call). •...