Asterisk PBX Jobs

We are looking for an experienced VoIP engineer (OpenSIPS + rtpengine) to debug and fix a NAT-related audio issue in our production SIP platform. We have OpenSIPS 3.3 + rtpengine 1.11 behind NAT. Calls work when SDP contains public IP. No audio when SDP contains private IP. RTP visible on firewall but not on server. Need an expert to debug NAT, firewall, and rtpengine behavior. Must have real OpenSIPS + rtpengine experience. Important: This is NOT a basic SIP config task. We need a senior VoIP engineer who understands deep NAT + RTP edge cases. If you don’t have hands-on OpenSIPS + rtpengine experience, please do not apply.

I’m expanding the tele-calling team in our company and have both part-time and full-time seats open right now. All new recruits begin with a one-week, paid training program so you can get comfortable with our scripts, CRM and daily targets before moving on to live calls. If you can spare four hours a day at any slot that suits you, the part-time track pays between ₹10,000 and ₹15,000 a month. Prefer a regular schedule? Full-timers work 9:30 a.m. – 5:30 p.m. and earn ₹15,000 – ₹25,000. In both cases your training week is fully salaried. What I need from you: • Clear communication in Telugu and basic English • A mobile phone or computer with stable internet for outbound calls • Serious intent to work—this is a genuine, long-term opportunity We re...

We are looking for an experienced VoIP / FreeSWITCH Engineer to work on FreeSWITCH open-source v1.10.2 implementation and configuration. The ideal candidate should have strong hands-on experience with SIP signaling, video calling (H.264), SIP gateways, and media handling. You will be responsible for configuring FreeSWITCH to support video, resolving connectivity and codec issues, and implementing SIP to RTMP recording/transcoding. Responsibilities Configure and troubleshoot FreeSWITCH v1.10.2 (open source) Enable and optimize video calling using H.264 codec Configure and manage SIP gateways and SIP interoperability Implement SIP to RTMP recording and video transcoding Debug SIP, RTP, media, and codec-related issues Ensure stable audio/video performance Required Skills FreeSWITCH ...

Hi, I just installed a fresh ViciBox 12. I need the following done: - firewall (iptables) - set domain (already pointed) - SSL - webphone - carrier trunk (IP auth) Ready to hire now

I need a person who can help me purchase VOIP UAE number so i can call/receive. Let me know what you need from me and what is the process

I need a production-ready voice agent that can dial new prospects, speak naturally, overcome objections and lock a date and time straight into our sales calendar. The agent must be built as true conversational AI—not a decision-tree script—so it can interpret open-ended replies, ask clarifying questions, and adapt its pitch in real time. Key requirements • Cold-call focus: the system starts with completely new leads and drives toward a booked meeting. • Full NLU: intent detection, entity extraction and context management that rival a human SDR. • Multi-language capability out of the box (English first, quickly extendable to at least one other major language). • Seamless telephony integration with our existing VoIP stack and calendar/CRM hand-off. &b...

I need a working UAE-based VoIP number that I can use immediately for day-to-day business communications. The line must be able to place and receive calls reliably inside and outside the country, on both desktop soft-phones and a mobile SIP client. If the provider requires specific registration documents or configuration steps, please outline those clearly and guide me through the activation so everything is compliant with UAE regulations from the outset. My expectations are straightforward: once you deliver the number and the SIP credentials, I will test the connection for call quality and stability; payment is released as soon as incoming and outgoing calls prove crystal clear for at least 24 hours. If you can recommend value-added options like voicemail-to-email or call recording, fe...

I run a small Asterisk lab and I need to run controlled test calls where the caller ID can be set to any value I choose. The goal is strictly testing and development, so everything must remain inside a legal, closed-loop environment. Here is what I need from you: • Configure or show me how to configure my existing Asterisk 18 instance so I can present arbitrary caller IDs on outbound calls. • Supply any dial-plan snippets, AGI scripts, or CLI commands required, along with clear commentary so I understand why each line is there. • Recommend a SIP trunk or gateway configuration that reliably passes the spoofed CLI without rewriting it (I am open to using my current trunk or switching to one you suggest). • Walk me through a short live demo—screen-share or re...

My organisation needs a dependable VoIP solution focused on crystal-clear international calling delivered entirely over the internet. I want a provider who can activate service quickly, supply SIP credentials, and guide me through basic configuration on soft-phones and standard IP desk phones. Reliability and audio quality are top priorities, so please outline the codecs you use, typical latency figures, and any uptime guarantees you can share. While my immediate requirement is outbound and inbound international voice, the platform should leave room to enable options such as video conferencing or voicemail later without a major migration. Expected deliverables: • Active account with international calling enabled • Step-by-step setup documentation (including screenshots or...

I’m launching a new venture that centres on simple, one-to-one voice calls. Every user will create a profile and attach a single, fixed price per call that is shown up-front to potential callers. When someone taps “Call”, the system should connect the two parties over VoIP, log the transaction, and charge the caller that fixed amount—no per-minute maths, just a clear flat fee. To get there I need the full workflow covered: secure sign-up/login, profile editing with a “price per call” field, in-app balance or card checkout, and a reliable voice stack (Twilio, Agora, WebRTC—whichever you’re most comfortable with). The UX must make price visibility obvious before the call begins, then automatically record that a call happened so the payment can...

Expert VoIP Engineer Needed: Multi-Tenant PBX Deployment (FusionPBX / FreeSWITCH Preferred) Project Description We are an IT Managed Service Provider (MSP) looking to build a robust, scalable, white-label VoIP platform to host phone systems for multiple distinct clients. We are looking for a senior VoIP engineer to deploy, configure, and secure a True Multi-Tenant PBX System. Important Architectural Requirement: We are NOT interested in a single-instance FreePBX installation hacked with custom contexts. We require a system designed for multi-tenancy from the ground up to ensure strict data isolation and security between clients. FusionPBX (FreeSWITCH) is our preferred platform, though we are open to VitalPBX (Carrier Edition) or Kazoo. Key Deliverables * Multi-Tenant Architecture: * Se...

I run several call-center deployments on Vicidial 2.14 and need a senior-level hand to keep everything humming. Day-to-day work ranges from setting up and configuring Vicidial for new campaigns, maintaining and troubleshooting existing clusters when agents hit snags, to developing custom features or scripts that extend core functionality. Performance optimization will be your most frequent touch-point with me, so deep knowledge of dial-plan tuning, MySQL query profiling, and carrier-side fine-tuning is essential. All work is remote, ticket-based, and ongoing: sometimes a quick patch, other times a full rollout on fresh hardware. When a task lands, I’ll provide SSH or VPN access, the specific goal, and the expected turnaround. Clean documentation of what you changed and why is requir...

I need an experienced VOIP and SIP Engineer. I have developed a custom AI Voice Calling Bot and it is currently connected with Twilio. Whereas my plan is to connect this AI Bot with almost any SIP provider. When i call from Twilio using my AI Bot, then it works correctly. Here i am needing you 1. I have installed Asterisk on my AWS. It is fully configured and making outbound calls. 2. My AI Calling Bot works with web hooks, and I have created another instance where I have created an outbound webhook in Node. And the AI Calling bot's webhook is placed in this code 3. When I call from Asterisk, the outbound call works, but there is a sharp noise in the call, and nothing else. My AI Bot should be listening in the call, but i only hear a sharp noise 4. I know this is due to mis-samp...

is currently throwing a 500 during outbound call orchestration and, right now, I can’t tell whether the fault sits in Synthflow or on the Twilio side. You’ll trace that root cause, correct it, and prove that outbound calls run cleanly end-to-end. You will have full access to the Synthflow integration settings, the relevant log output and screenshots around the error timestamps, plus my Twilio account details so you can dig into Twilio Voice, Webhooks, Studio flows or any other layer that might be mis-configured. Deliverables • Written diagnosis identifying the failing component and reason for the 500 • Code/configuration fix applied and documented • Successful test calls that show reliable audio connection and call completion • Short hand-off repo...

I need an SMS messaging system set up using Twilio. The system should enable sending and receiving messages and also send messaging for whatsapp. I want a complete document built for me so I can replicate. Key Requirements: - Integrate with Twilio - Setup to send and receive SMS Ideal Skills and Experience: - Experience with Twilio - Familiarity with SMS integration and setup - Strong technical skills in messaging systems

Our voice flow stops at the very first moment it tries to place a Twilio call—the API fails right at the initiation stage. I can see the request go out, but Twilio returns an API-level error and the call never even rings. Everything else in the project is wired up and tested, so the only thing blocking us is this single integration point. I need someone who knows Twilio’s voice API inside out and can troubleshoot, patch, and verify the fix immediately. You’ll have SSH and dashboard access the moment you accept; please be ready to dive straight into SynthFlow’s webhook configuration, Twilio console, and any relevant server-side logs. The goal is simple: resolve the API call error and prove that a test number rings successfully—all within the next two hours. ...

I need an SMS messaging system set up using Twilio. The system should enable sending and receiving messages and also send messaging for whatsapp. I want a complete document built for me so I can replicate. Key Requirements: - Integrate with Twilio - Setup to send and receive SMS Ideal Skills and Experience: - Experience with Twilio - Familiarity with SMS integration and setup - Strong technical skills in messaging systems

I run a busy residential real-estate practice and I’m ready to hand off the constant stream of incoming queries to an AI-driven assistant. The goal is simple: every phone call, text, email, or WhatsApp message must be handled instantly and professionally, so no lead ever slips through the cracks. Here’s what the agent has to do for me day-to-day: • Answer common buyer and seller questions about listings, showings, pricing, and neighbourhood details. • Schedule appointments directly on my synced calendar, confirming date, time, and property address with the caller. • Take detailed messages whenever the request falls outside the approved knowledge base and send them to me by email or text. The system should prioritise voice calls first, but also mirror the sa...

I’m part of the Simplifi HR and Payroll team and need an extra pair of skilled hands to keep our recruitment funnel running smoothly. Day to day, you’ll jump on short screening calls with applicants, record concise notes in our cloud-based ATS, and make sure every offer letter, I-9, and compliance form is uploaded and verified before onboarding moves forward. You’ll receive a full software stack—ATS, HRIS, secure e-signature platform—and walk-through videos that explain our exact workflow, so there’s no guesswork. What I value most is clear communication, respect for confidentiality, and the ability to keep digital records tidy and up to date. Deliverables I measure: • Completed screening interviews with notes logged the same day • Onboa...