SIP Jobs

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 stack (Node/Express,...

I want my new AWS account configured to host a 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...

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 this done ...

Client needs a VoIP Intercom Station operating in Standards 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 Un...

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-points &b...

Mam na własnym VPS zainstalowaną najnowszą wersję GoAutoDial 4. Potrzebuję 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 zmiana...