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Asterisk PBX is an open source telephony framework used to create communication solutions for businesses, from voice and video calls to automated call routing. An Asterisk PBX Development professional is highly knowledgeable in the Asterisk platform and is able to deploy custom Asterisk Solutions according to a company’s needs. Services offered by an Asterisk Developer include managing, monitoring and maintaining the Asterisk system and any external services, building custom features specific to the business and improving security through authentication and encrypted communications.
Here's some projects that our expert Asterisk PBX Developer made real:
Projects like these require experienced developers who understand the Asterisk PBX platform. If you’re looking for someone with that precise set of skills, post a project on Freelancer.com to hire an Asterisk PBX Developer today. Our team of highly qualified professionals will be able to help you design, develop and deploy custom communication solutions for your business.
From 20,703 reviews, clients rate our Asterisk PBX Developers 4.93 out of 5 stars.Asterisk PBX is an open source telephony framework used to create communication solutions for businesses, from voice and video calls to automated call routing. An Asterisk PBX Development professional is highly knowledgeable in the Asterisk platform and is able to deploy custom Asterisk Solutions according to a company’s needs. Services offered by an Asterisk Developer include managing, monitoring and maintaining the Asterisk system and any external services, building custom features specific to the business and improving security through authentication and encrypted communications.
Here's some projects that our expert Asterisk PBX Developer made real:
Projects like these require experienced developers who understand the Asterisk PBX platform. If you’re looking for someone with that precise set of skills, post a project on Freelancer.com to hire an Asterisk PBX Developer today. Our team of highly qualified professionals will be able to help you design, develop and deploy custom communication solutions for your business.
From 20,703 reviews, clients rate our Asterisk PBX Developers 4.93 out of 5 stars.I’m kicking off a high-volume residential solar campaign and need skilled cold callers who can jump in right away. All outreach will be done by phone; I’ll supply the homeowner data, a VOIP dialer login, and a tried-and-tested script to keep things straightforward. Your day-to-day is simple: dial, qualify interest in a solar consultation, and book the appointment in my calendar tool. I track performance in real time, so transparency and accurate notes after each call are essential. If you’re an individual hustler or a full team with proven solar experience, I’m open to both—what matters is consistent results and a professional phone presence. Deliverables I expect: • Daily call log with contact outcome and brief notes • Qualified residential so...
I’m kicking off a high-volume residential solar campaign and need skilled cold callers who can jump in right away. All outreach will be done by phone; I’ll supply the homeowner data, a VOIP dialer login, and a tried-and-tested script to keep things straightforward. Your day-to-day is simple: dial, qualify interest in a solar consultation, and book the appointment in my calendar tool. I track performance in real time, so transparency and accurate notes after each call are essential. If you’re an individual hustler or a full team with proven solar experience, I’m open to both—what matters is consistent results and a professional phone presence. Deliverables I expect: • Daily call log with contact outcome and brief notes • Qualified residential so...
The GSM gateway (Yeastar TG400) currently refuses to register with my Grandstream UCM6308, so calls never make it through. I have remote access to both devices ready for you and can open the ports you need. I have not yet verified every network or SIP setting, so the first step will be a thorough review of both sides—IP addresses, SIP trunks, codecs, NAT, authentication and routing rules—until the two units establish and keep a clean connection. Once registration is solid, I want unanswered calls to be diverted automatically after 30 seconds. Set up and test the forwarding rule so an incoming GSM call that rings for half a minute is routed to the specified fallback destination without fail. Deliverables • Stable TG400 ⇄ UCM6308 SIP registration • Working 30-s...
I need Vapi wired up to Telnyx so my application can place and receive AI-driven voice calls through the same number. Both directions matter—outbound campaigns as well as inbound inquiries must flow through Vapi’s real-time audio stream and, when the caller needs human help, roll over instantly to a live agent. Here is what success looks like to me: • Telnyx numbers authenticated and reachable from Vapi. • Webhooks or websocket events mapped so call status, audio, and transcripts reach my backend. • Logic in place for a seamless live-agent transfer, preserving call context. • End-to-end test demonstrating one inbound and one outbound call, each handled by the AI then passed to a human. I’ve already got API keys for both services and a Nod...
I have a fresh server waiting for a clean Asterisk installation and I want it to come online already loaded with both VC Dial and a working Voice-AI module. Your mission is to take me from zero to a fully functioning system that can place and receive calls through VC Dial while handing certain interactions off to the AI engine. Core expectations • Install the appropriate Asterisk version on my Linux VPS (CentOS/Ubuntu, your preference as long as it is stable). • Deploy VC Dial, confirm the web interface, database and dialer services all start automatically, and create a sample inbound + outbound campaign so I can see calls flowing. • Integrate the Voice-AI component (Google, AWS, or another open-source alternative—let me know what you prefer) so that an IVR menu o...
I need Asterisk 22.6.0 set up on my Windows laptop. Please enable the following modules: - Voicemail - Conference calling - Call recording Ideal skills and experience: - Expertise in Asterisk, especially version 22.6.0 - Proficient in Windows OS - Experience with VoIP and PBX systems - Knowledge of configuring Asterisk modules
I am rolling out a brand-new Odoo instance together with a fresh Grandstream PBX and want them to work as one seamless system from day one. The core goals are clear: every call handled by Grandstream should be logged inside Odoo, contacts must stay perfectly synchronized between both platforms, and voicemails need to appear in Odoo with playback and status indicators. On top of that foundation, I’d like to weave in three AI capabilities: • an automated response layer that can greet callers and route them intelligently, • live voice-to-text recognition so conversations and voicemails are transcribed directly into Odoo records, and • predictive analytics that can flag high-value leads or at-risk customers based on call patterns. Because nothing is installed y...
I want to deploy an AI-driven voice agent on our 3CX phone system that can: • answer incoming calls naturally in English and Nederlands; • switch languages mid-conversation when needed, providing real-time language translation; • book or reschedule appointments by talking directly to callers and writing the results into our Odoo CRM calendar (no separate calendar app—Odoo holds the single source of truth). Key expectations – Seamless SIP or WebRTC connection to 3CX, so every extension can route to the agent when human staff are busy. – Accurate speech-to-text and text-to-speech with voices that sound professional in both supported languages. – A dialogue flow that recognises common scheduling phrases, verifies caller details, and confir...
I need a lightweight VOIP / CPaaS solution configured specifically for outbound sales calls. Call recording must be enabled and reliably archived, but beyond that I’m keeping things lean—no CRM, email, or other system integrations right now. Here’s what I’m expecting: • Recommend a reputable cloud or self-hosted provider (Twilio, Plivo, RingCentral, FreePBX, etc.) and justify the choice for high-volume sales dialing. • Configure numbers, routing, call recording, and basic reporting. • Deliver concise user instructions so my team can start calling immediately. If you’ve rolled out similar sales-focused VOIP setups before, I’d love to see it go live quickly and hassle-free.
I’m standing up a large-scale platform that blends Twilio, the WhatsApp Business API, and OpenAI/Claude into a single, fast, secure messaging layer. Dockerized Linux servers are ready to be spun up—we just need a real engineer (no middle-men, no agencies) to wire every piece together and own the stack from day one. The build will lean on both Twilio and WhatsApp. On the Twilio side we’ll be firing off SMS, Voice, and A2P traffic; WhatsApp will cover session and template messaging. Everything must sit behind a clean, well-documented API with solid auth, rate limiting, and logging. Security is non-negotiable. Deliverables I need from you: • Provision and harden the server (Docker + Linux) • Integrate Twilio (SMS, Voice, A2P) and WhatsApp so messages flow both ...
I need a voice-driven bot that can handle both outbound and inbound calls, with its main job being appointment setting. My current tech stack is centred on N8N for orchestration, and I’m happy to pair it with either an 11labs or Vapi voice layer—whichever gets us to a smooth, human-sounding experience fastest. If you already have a working solution you’re willing to licence inexpensively, let me see it; speed to deployment and overall cost matter. Otherwise, I’ll need you to build and integrate the following: • Call handling: automatic dialling for outbound campaigns plus a reliable way to receive incoming calls. • Conversational logic: a clear appointment-setting flow that can confirm dates, times, and any basic qualifying questions. • N8N work...
We are looking for an experienced VoIP engineer to configure Kamailio as an SBC together with RTPengine. The goal is to fully rewrite all SIP and RTP traffic so that the upstream provider sees only the public IP of the Kamailio server — no trace of the original Asterisk system. Connection from Asterisk to Kamailio will be over direct IP (SIP trunk), no DNS. Environment (already prepared): Clean Debian 12 (on a dedicated server, not VPS) Kamailio 5.6.3, MariaDB, RTPengine installed Proper DNS and Debian repositories configured SSH access is ready Task Requirements: Configure Kamailio as an SBC for SIP trunk traffic coming from Asterisk Integrate and configure RTPengine for full media relay and SDP rewriting Ensure all SIP headers (From, Contact, Via, etc.) and SDP contain on...
A steady flow of incoming calls means I need a calm, friendly voice on the line all day. The role is pure customer service—no selling, no technical troubleshooting—just answering general inquiries about our products, order status, basic policies, and recording each interaction in our cloud-based CRM. You’ll log in to our VoIP system for an eight-hour shift, Monday through Friday, handle every call as it arrives, take concise notes, and flag anything that needs escalation. Clear spoken English, upbeat phone etiquette, and a distraction-free workspace are essential. Familiarity with tools like Zendesk or Freshdesk is helpful, but I’ll provide scripts and onboarding so you can get up to speed quickly. What I expect • Live call coverage for the full shift (no m...
A steady flow of incoming calls means I need a calm, friendly voice on the line all day. The role is pure customer service—no selling, no technical troubleshooting—just answering general inquiries about our products, order status, basic policies, and recording each interaction in our cloud-based CRM. You’ll log in to our VoIP system for an eight-hour shift, Monday through Friday, handle every call as it arrives, take concise notes, and flag anything that needs escalation. Clear spoken English, upbeat phone etiquette, and a distraction-free workspace are essential. Familiarity with tools like Zendesk or Freshdesk is helpful, but I’ll provide scripts and onboarding so you can get up to speed quickly. What I expect • Live call coverage for the full shift (no m...
We are seeking an experienced professional to create a mirror or backup of our ASTPP server. This is a one-time project requiring expertise in server management and data integrity. ASTPP server 1 must be linked to ASTPP server 2, so that the DB of ASTPP server 2 has the same data as the DB of ASTPP server 1, both server DB's and from https interfaces. In case ASTPP 1 fails, it must not affect ASTPP 2 server and ASTPP 2 can become the main server. The same in case of mysql failure or corruption. MYSQL failture or corruption on one of those 2 server, must not affect the server number 2. Freelancer must carry out the task, then provide a detailed document on how to achieve the same result on any other 2 ASTPP servers. Someone from our side with limited technical knowledge will take the ...
I have two VOIP call-detail-record files that are at the centre of a billing dispute and I need a clear, defensible comparison of them. The goal is to line up the calls in both files, calculate each side’s duration, and show me only the cases where those durations don’t agree. The final deliverable must be a single CSV file that I can forward to the carriers as evidence. Deliverables • A consolidated CSV that pairs matching calls, shows duration A vs. duration B, and includes both the raw difference and any percentage variance. • Only the calls with unmatched durations should be highlighted or extracted; fully matched records don’t need emphasis. • My two original CDRs returned untouched, plus any script, query, or workbook you build so I can rerun the...
If you want to stay competitive in 2021, you need a high quality website. Learn how to hire the best possible web developer for your business fast.
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