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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 20,935 opiniones, los clientes califican nuestro Asterisk PBX Developers 4.9 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 20,935 opiniones, los clientes califican nuestro Asterisk PBX Developers 4.9 de un total de 5 estrellas.Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all details: Range, number, carrier, carrier payout, carrier pay term*, client payout**, to which client it's allocated if it is, and Notes -Search options: search per client allocated numbers, per carrier, per range, per country, per rate (more than or less than or =), per number, etc. -Option to download CSV file numbers of the searched result, or for all if search is reset for all -Select numbers (...
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 want to replace the manual US Calling sales calls my team is making with a fully automated, AI-driven outbound dialer. The system has to plug straight into our existing CRM so that every call is launched from, and written back to, a single customer record. Core workflow • The dialer should pull a lead from the CRM, place the call automatically, open a tailored script, and capture everything the prospect says through real-time speech recognition. • Natural language processing must understand intent and sentiment, while AI-driven responses keep the conversation moving until a meeting is booked or a hand-off to a human rep is required. • Once the call ends, the transcript, outcome, and next-step task should save to the same CRM record without anyone touching a keyboard...
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 ...
I need to walk through, configure, and fully test an after-hours rule on my RingCentral call queue. Outside of business hours every call should go directly to voicemail, unless the caller flags it as urgent—in that case it must be forwarded instantly to the team lead’s phone. I want this built live with me so I understand each step. Once the escalation path is working, we’ll spend the remainder of the hour fine-tuning other features I use every day, specifically the IVR menu settings and the newer AI receptionist options. A short screen-share session is perfect; one focused hour should cover everything. Deliverables (completed during the call) • Functional call-queue rule that sends after-hours calls to voicemail • Verified urgent-call forwarding to the team...
Hi Vicidial is installed and working. Calls are being made too. Issue is i need to make calls to different campaigns. But it calls to one campaign. Here is work flow I have AI bot that works with vicidial via node bridge. When i make calls from vicidial, it connects to bridge and bridge then finally connects to AI bot. I have created 2 campaigns medicare and final expense. It must dial both campaigns and i must receive calls on both numbers. Currently i can receive call on medicare campaign only. I need to fix this issue. So this job is only for them who can resolve this issue.
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...
We are building a simple but reliable automation system designed for driving instructors. Driving instructors often miss calls while they are teaching lessons. When a potential student calls and the instructor cannot answer, that enquiry is usually lost. The goal of this project is to automatically recover those missed enquiries. The system should automatically respond when a call is missed, send a message to the caller, and notify the instructor if the caller is interested in booking lessons. This is not a prototype or experiment. We need a stable working system that runs automatically and reliably. Main Goal The main objective is to make sure driving instructors never lose potential pupils when they miss a phone call during a lesson. When someone calls and the call is not answered, ...
I want to replace the manual US Calling sales calls my team is making with a fully automated, AI-driven outbound dialer. The system has to plug straight into our existing CRM so that every call is launched from, and written back to, a single customer record. Core workflow • The dialer should pull a lead from the CRM, place the call automatically, open a tailored script, and capture everything the prospect says through real-time speech recognition. • Natural language processing must understand intent and sentiment, while AI-driven responses keep the conversation moving until a meeting is booked or a hand-off to a human rep is required. • Once the call ends, the transcript, outcome, and next-step task should save to the same CRM record without anyone touching a keyboard...
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...
I need a complete IVR built and configured to streamline our customer-support line. The system must greet callers, then automatically route each call to one of five departments through a simple, voice-prompt menu. In addition to this core routing, I also want the IVR to trigger bulk messages (SMS or WhatsApp blasts) and hook directly into Meta Business Suite so our phone flows, social inbox, and chat automations all stay in sync. You are free to choose the underlying platform—Asterisk, FreePBX, Twilio, or a comparable cloud service—so long as it supports reliable call routing, bulk-messaging APIs, and Meta’s official integrations. If a ready-made template speeds things up, that is fine; the voice prompts, language, and menu logic must still be customised to our brand. D...
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...
I’m looking to partner with a reliable call-centre operation that can dedicate 1 – 5 agents exclusively to inbound customer support over the phone. All interactions will be in English, so clear diction and a neutral accent are essential. The agents will be the first point of contact for my customers, answering questions, resolving basic issues, and routing complex cases to my in-house team when needed. Average volume is modest right now, but I need a team that can scale quickly if call traffic grows. Core expectations • Live phone coverage during the agreed shift window • Courteous, solution-focused conversations that follow my brand guidelines • Real-time ticket or CRM updates so I always know the status of each call • Daily call logs and a conc...
Project Title: Full Issabel 5 PBX Deployment: Installation, Trunks & Extensions Project Overview I am seeking a VoIP specialist to perform a complete installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architecture, including SIP trunks for external connectivity and internal extensions for users. Detailed Scope of Work 1. Server Installation & Hardening Perform a clean installation of Issabel 5 on AlmaLinux 8 using the official net-install script. Configure Fail2ban and firewall rules to block unauthorized SIP and SSH attempts. Set secure passwords for the Linux root, MariaDB, and Issabel web admin. 2. SIP Trunk Configuration Connect the PBX to my chosen VoIP provider using a SIP trunk....
I don't have FusionPBX installed on any server yet. You will install and setup fusionpbx on a server I will provide credentials access into. I want to be able to present a specific caller ID on every outbound call placed from my softphone. The softphone I will be using is Linphone, so everything you build must be fully tested from that app before we wrap up. What I need you to do • Configure the relevant SIP profile(s) and outbound route so that the number I supply appears consistently, regardless of the device extension that initiates the call. • Make any dial-plan or gateway adjustments required for a typical commercial SIP trunk. I have not fixed on one provider yet, so keep the setup provider-agnostic and document the few fields I will have to tweak once I decide. ...
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