SIP Jobs

Buscamos Ingeniero de Telecomunicaciones / Administrador de Sistemas Senior con experiencia avanzada en VICIdial y Asterisk PBX (Habla Hispana). Responsabilidades y tareas clave: Optimización de infraestructura: Configuración, mantenimiento y tuning de servidores VICIdial (arquitectura Single o Cluster). Bases de datos: Optimización fina de base de datos (MySQL / MariaDB) para soportar alto volumen de llamadas simultáneas y evitar bloqueos de tablas. Desarrollo e IVR: Configuración de campañas predictivas, gestión de IVR avanzado, enrutamiento inteligente y scripts AGI personalizados. Soporte y conectividad: Resolución de problemas de latencia, calidad de audio y configuración de troncales SIP / carriers. Requisitos indis...

I have a local Twilio number that must reliably reach me wherever I am. Your task is to build a clean call-flow that forwards every inbound call to both my mobile handset and my SIP softphone, with time-of-day rules and the flexibility to add or remove extra destinations later. Here is what I need: • A Twilio SIP Domain (or compatible Elastic SIP Trunk) that hands calls off to my chosen softphone client, while simultaneously or sequentially ringing my mobile depending on the schedule I provide. • Time-based routing so after-hours calls follow a different path. • Clear call-quality tuning—codec selection, jitter buffer settings, and any necessary NAT or firewall advice—to guarantee stable, low-latency audio. • TwiML or Studio Flows documented so I...

My FreePBX server is already connected to a Twilio Elastic SIP Trunk using a Credential List. Zoiper extensions register without issue and the PJSIP endpoint for Twilio shows “Available,” yet every outbound attempt returns “All circuits are busy now.” Nothing at all appears inside the Twilio call logs, so the traffic never reaches their side. I suspect the PJSIP trunk settings are to blame, though the outbound route or some subtle SIP-signaling mismatch could also be involved. No recent changes have been made to the trunk configuration, so whatever is wrong has likely been there from the start. What I need from you • Trace and isolate the fault across the outbound route, PJSIP trunk, and SIP signaling (sngrep or tcpdump, Asterisk CLI, etc.). • Apply the ...

I am ready to launch a new voice service and need a fully verified USA-based SIP trunk that lets me place and receive both domestic and international calls. Monthly traffic will sit in the 1,000 – 10,000-minute range and the trunk must support several concurrent calls without choking, so adequate channel capacity is essential. Here is what I need from you: • Introduce or provision a reputable provider that can activate an account in my company’s name with all necessary KYC completed. • Ensure the trunk is enabled for two-way voice, CLI presentation, 911 compliance, and international reach. • Supply full connection details (SIP URI, username/password or IP auth, codec list) plus concise step-by-step setup instructions for popular soft-switches or an Aster...

I’m building a straightforward voice-only calling app dedicated to Android. The core requirement is rock-solid one-to-one voice communication, so there’s no video feed or chat module to worry about—just clean, low-latency audio calls. Users will sign up and log in with phone-number verification (OTP/SMS). Once authenticated, they should see a simple contact list, tap a name, and place a call. Think of an experience as seamless as a standard dialer but inside its own sandboxed app. What I need from you: • Full Android Studio project (Kotlin or Java) with well-commented source code • Functional APK for testing on Android 10+ devices Acceptance: • Phone verification must complete within two steps and fail gracefully on bad codes • Voice connection ...

Project Overview We are looking for an experienced FreePBX / Asterisk developer to install, configure, and develop a complete REST API for automating telephone bridge calls. We will provide: SSH access to a Linux server. SIP trunk credentials. Access to a remote server where call recordings will be stored. The final solution must be production-ready, stable, secure, and well documented. Phase 1 – Installation & Configuration The developer will: Install and configure FreePBX (preferred) or Asterisk if there is a clear technical advantage. Configure one or more SIP trunks. Our SIP provider allows multiple DDI (Caller IDs) on the same SIP trunk. The system must support multiple SIP trunks simultaneously. Configure automatic call recording. Secure and optimize the PBX for prod...

Design, configure, and maintain Asterisk-based telephony systems. Develop and manage SIP trunk configurations, IVRs, call routing, DID management, and dial plans. Integrate telephony systems with REST APIs, webhooks, and third-party platforms. Build and maintain API integrations for the Neurova platform. Configure inbound and outbound call flows, click-to-call, predictive dialing, and intelligent routing. Work with AI Voice Agents and ensure seamless human call transfer (human failover). Integrate WhatsApp, SMS, Email, CRM, and other communication channels where required. Troubleshoot voice quality, SIP signaling, RTP issues, and carrier connectivity. Collaborate with the AI development team to integrate Speech-to-Text (STT), Text-to-Speech (TTS), and LLM-based conversational workflows. Op...

I already have a Grandstream UCM-series IP PBX on site and reachable for remote work. The hardware is up, the , but the system is still running on factory defaults. I need a Grandstream-savvy engineer to jump in and complete the core setup: • Create and test all user extensions (SIP and any analog FXS ports that may be required). • Build a clean inbound route so external calls land where they should. • Design and activate a simple operator/IVR menu that greets callers, routes them to the right extension or ring group, and falls back to voicemail if unanswered. Please handle dial plans, basic voice prompts, time conditions, and any minor tweaks that are typically part of making the above work smoothly. I’ll supply extension lists, Remote access is available via Te...