SIP Jobs

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). •...

My Panasonic Ns500 PBX sits on but cannot “see” the rest of my network. Everything else flows through a FortiGate 60F firewall, a FortiSwitch 424E-Fiber core, and a FortiSwitch 124F-FPOE at the edge. I need someone to shape the network so this Panasonic box can handle VoIP communication smoothly. What I already know • The PBX will run pure SIP. • Dedicated VoIP rules on the FortiGate are required; simple, generic access is not enough. What I need from you • Review the current FortiGate policy set, VLAN layout, and switch port profiles. • Create or adjust firewall rules, NAT, and any SIP ALG or helper settings so that SIP registration, signalling, and RTP streams pass without one-way audio or dropped calls. • Tag or untag the appropriate swi...

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...

Your bid MUST include answers to the 5 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 Web...