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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.
De 21,887 opiniones, los clientes califican nuestro Asterisk PBX Developers 4.81 de un total de 5 estrellas.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.
De 21,887 opiniones, los clientes califican nuestro Asterisk PBX Developers 4.81 de un total de 5 estrellas.I need a small team of energetic cold callers who already feel at home inside ViciDial and can switch effortlessly between English and Spanish. Your three-hour calling window will run from 8:00 AM to 11:00 AM Texas (CST) time, Monday through Friday. During each session you will be dialing pre-qualified personal-injury prospects pulled straight from the cloud-based CRM I provide; all contacts, scripts, rebuttals, and disposition codes are ready for you, so you can focus purely on persuasion. Because many leads prefer to speak Spanish, true bilingual fluency is essential. Prior experience scheduling or screening personal-injury cases is a strong plus—knowing the difference between soft-tissue and catastrophic claims, basic statute-of-limitations rules, and how to spot liability red fl...
About the role: We sell a SaaS solution to eCommerce companies (mid-size and high-end online brands). We are hiring remote cold callers to book qualified demo appointments for our sales team. This is B2B calling to business owners and managers - not consumers. What you will do: - Cold call eCommerce businesses from provided lead lists and book qualified appointments - Qualify prospects against clear criteria before booking (right size, right fit, real interest) - Own every appointment you book until it happens: send calendar invites, confirm by phone/text/email, and run reminder follow-ups so prospects show up at the scheduled time - Re-book no-shows and keep the pipeline updated in our CRM/tracker - Daily reporting: dials, connects, booked, showed What we require: - 2+ years of B2B cold...
I want to create an Android application that works much like the Superfone IVR app, but tailored to my organisation’s customer-support workflow. The core goal is to make it effortless for callers to reach the right information or person without waiting on a human operator. Key functions I need built in: • Automatic IVR flow that greets callers and routes them through spoken menu options. • Branches that allow callers to: – get answers to common questions, – record and escalate complaints, – request detailed information that will be sent to them by SMS or email. • An intuitive dashboard where I can update the menu prompts and review call analytics. • Secure cloud storage for recordings plus the ability to download ...
I run a busy telemarketing agency and need a reliable, scalable way to record every outbound call my team makes from their desktop computers. I’m committed to a fully cloud-based solution so we never have to maintain on-premise hardware, yet I still want enterprise-grade security, redundancy, and easy access to recordings. The core requirements are straightforward: • Automatic call recording that captures 100 % of outbound traffic the moment a call is placed. • Built-in call analytics so I can track agent performance, campaign effectiveness, and compliance metrics from a central dashboard. Beyond those essentials, the platform should let me search, tag, and download recordings quickly, with the option to add real-time transcription down the road. Seamless integration...
STRICT RULES - READ BEFORE BIDDING 1. ZERO-TOLERANCE HARDWARE RULE: DO NOT BID if you do not have direct, hands-on experience with ALL of these specific devices: Cudy P2 5G Router DrayTek Vigor 2915 Router Yeastar S20 PBX We are a live production business, NOT a testing ground or a network engineering academy. If you do not know these exact models inside out, do not waste our time. We will not teach you how our network works. 2. THE CORE AUTOMATION GOAL (MANDATORY): We demand a 100% seamless, automated experience for the end-user. Moving in and out of the office must require ZERO manual network adjustments. Smart VPN (Always-On): It must run 24/7. When a user walks into the KSA office and connects to the local Wi-Fi, the routing must automatically prioritize local traffic without con...
I will provide fresh car-accident leads, Vicidial access, and our CRM; you log in each morning on Texas time, start dialing, and keep the line moving. Your single objective is to qualify the person for our attorney, confirming two points only—injury details and current insurance—then smoothly warm-transfer the caller to the lawyer’s line. Because some victims prefer to speak in Spanish, you must be fully fluent in both Spanish and English and able to switch effortlessly mid-conversation. Cold-calling experience, a persuasive tone, and the discipline to follow the script while thinking on your feet are crucial. Daily expectations: stick to the dial schedule, maintain clear notes in the CRM, hit the transfer quota we agree on, and flag any bad numbers or DNC requests ...
# TelesalesOS — Telnyx Telephony/ACD Contact Center Developer Our stack is , React, TypeScript, Supabase/Postgres/Auth/Realtime, Vercel and GitHub. We need an experienced programmable-voice/contact-center developer to build the Telnyx-powered telephony portion. This is not a simple click-to-call integration. We need a production-grade browser softphone, inbound ACD, queues, routing rules, supervisor monitoring, recordings, CRM screen pops, compliance controls and searchable call logs. ## Required experience Applicants should have real experience with several of the following: * Telnyx Voice API/Call Control * Telnyx JavaScript or React WebRTC SDK * SIP and WebRTC * Browser softphones * ACD/contact-center development * Call queues and routing * Telnyx conferences * Monitor, whisp...
I need an experienced FreePBX professional who will teach me what I want to learn about freepbx operation and backup You can share your freepbx installation or use mine. You will look over my current installation if needed and help me fix the issues so that I can run it smoothly. So the focus is strictly on troubleshooting and ongoing support rather than fresh installation. I plan to use your experience instead of AI support. If you routinely work with Asterisk/FreePBX CLI, have a solid feel for SIP tracing tools such as sngrep or Wireshark, and are comfortable working/teaching on production systems where downtime needs to stay minimal, you are exactly who I’m after. I would like to use a help convenient with my work hours so that I don't cheat with my boss.
I'm looking for an experienced Asterisk VoIP engineer to help with PBX configuration, specifically for call routing. Key requirements include: - Expertise in Asterisk PBX - Proficiency in configuring call routing, especially geographic routing - Experience with VoIP systems and network configurations Ideal skills and experience: - Background in telecommunications or related field - Previous projects involving Asterisk and PBX systems - Strong problem-solving skills and attention to detail Please provide examples of similar work done.
I run a service center and I’m ready to streamline two key touch-points at once: inbound customer support calls and WhatsApp messaging. I am not working on CRM and ticketing system, so the solution must plug directly into both of those platforms—ideally through secure APIs or webhooks—so every interaction is logged automatically without any manual data entry. I only need straightforward rule-based flows: IVR menus for phone calls, and button- or keyword-driven replies on WhatsApp. Natural-language understanding is not required; speed, reliability, and clear routing rules are the priorities. Please propose the tech stack you’ll use (Twilio, 360dialog, Asterisk, Vonage, Dialogflow CX in “no-NLP” mode, or any other you prefer) along with the integrati...
We are currently working on our mobile application, GCM Dialer, which handles both SIM and SIP calls. We have successfully implemented the conference screen for the SIM and SIP merge option, but we are encountering an issue where voice is not being exchanged between all parties. Below is the scenario we are testing: - User-1 has the GCM Dialer app installed and set as the default dialer. - User-2 calls User-1 via SIM, and they can exchange voice successfully. - User-3 then calls User-1 via SIP. User-1 answers, which puts User-2 on hold. User-3 and User-1 can exchange voice successfully. - User-1 clicks the "Merge" button, and all three parties are brought into one call screen. The issue is that while User-1 can hear and speak to both User-3 and User-2, User-2 and User-3 cannot...
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