Asterisk amdcông việc

Tôi cần phat triển phần quản trị cho giai pháp ai smart dialer va cần nhân sự biet ve asterisk python va php mysql

Customize IVR for Asterisk: 1; Play announcements 2; Record 3; Transfer

Tích hợp plivo với freebpx/asterisk để xây dựng tổng đài support simbox và hotline

Tôi đã có sẵn tổng đài Asterisk, đã có sẵn SuiteCRM. Cần lấy API từ tổng đài Asterisk để kết nối với Suitecrm.

Hi vule83, mình đang có dự án Odoo CRM tích hợp tổng đài voIP sử dụng Asterisk 11. Bạn đã từng làm dự án nào tương tự chưa? Nội dung cụ thể thì cho mình số dt của bạn để trao đổi cho nhanh nhé

Chào các bạn, hiện tại tôi đang rất muốn kết nối vtiger opensource 7 với tổng đài assterisk, nhưng tôi không biết làm việc này, rất mong các bạn hỗ trợ tôi kết nối giữa 2 ứng dụng này, tôi sẽ trả phí kết nối cho các bạn. Trân trọng cảm ơn!

Hi nttranbao, I'm a newbee asterisk. Now, my project are doing function monitor video call with asterisk. Do you do that function? If you can, you can send email for me ( @). Thank you :) Hy vọng anh biết tiếng Việt. vì tiếng anh của em không tốt lắm. Em mới nghiên cứu asterisk và hiện tại dự án của em đang muốn làm một chức năng theo dõi cuộc gọi video của người dùng. Anh có làm được không ạ? Nếu được anh liên hệ trực tiếp với em qua email @ để trao đổi chi tiết hơn ạ.

As a sysadmin developer, I'm in need of an asterisk specialist to build a Docker Compose script or a bash script for an interactive vocal server. This project is multifaceted, carrying out outbound calls and saving responses in a database. Key responsibilities are: * Creation of an Interactive voice response (IVR) system. * Outbound calling function connected with my API for automated scheduling of phone calls. * MySQL database integration to securely store the recorded responses. For this assignment, it would be ideal if you have proficiency in using Asterisk, Docker Compose, API integration along with comprehensive database management skills It would be a cherry on top if you have prior experience constructing interactive vocal servers. Let's connect to disc...

We have moved from a VPS to a dedicated server and since then our web server becomes inaccessible every 24 to 48 hours and we have to reboot it to work. It is running on Plesk and we have our PHP Websites and Web applications running on this machine. It is a good machine AMD Ryzen 9 dedicated server. Not sure what is going on. We need help from an expert to identify and resolve this completely.

I'm looking for a professional who can install Asterisk PBX to facilitate a call routing system. Key Requirements: - Asterisk PBX will function primarily as a call router, modifying incoming caller ID's to the outgoing trunk DIDs. - The system should handle both incoming and outgoing calls efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice tran...

I'm in need of a skilled server administrator to manage my Linux server dedicated to web hosting. After moving to Dedicated Server AMD Ryzen 9 we have seen that the server goes down so often and we have to restart it. Key Responsibilities: - Ensure the server is running efficiently and troubleshoot any issues that arise - Manage and maintain the server's security, including setting up a firewall, implementing malware protection, and installing SSL certificates - Monitor server performance and make recommendations for improvements Ideal Skills and Experience: - Proficient in Linux server administration - Experience with web hosting servers - In-depth knowledge of server security measures, including firewall setup, malware protection, and SSL certificate installation - Pr...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..

As an AMD GPU owner, I'm facing severe problems with my PC not booting correctly. Troublingly, my computer is persistently crashing and there is no display on the monitor. I've tried both reinstalling drivers and checking GPU seating as initial troubleshooting steps, but these have unfortunately failed to rectify the issue. The job will involve pinpointing the root cause of these issues and implementing a solution that will let my PC boot properly. Freelancers interested in this job should ideally have: - Proficiency in troubleshooting AMD GPUs. - Knowledge on different PC hardware components. - Experience in diagnosing and resolving hardware boot issues. - A good understanding of driver installation and configuration. The objective here is not just to fix th...

...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

okay hello here, l know app devepers are highly priced! But the thing is here: My experience in the past was bad with deliver and functionality of my requesteds app's So l will make a milestone of 10 usd and not release until l can see if the Bot is functional My app is "called" XM ......of 100 usd , start making like 1 , 2 ... 10 usd daily will be more than nicier... as the times go on l'll invest my money and pay you the one affordable price for a big and well developed BOT , in the market an app costs 1k-15k$ right know my budget is 10 on front for your job and 100 after l make it with the bot , so you have to wait until your job makes succesfully good job... Good luck! amd thanks and sorry for this missanderstandings with my budget on all coders up ...

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.

I require legal services for property purchase/sale and specifically for dealing with Debt Recovery Tribunal, filing of applications in NeSL / Insolvency case for recovery amd follow up with the IRP for resolution of claims It would be preferable if the freelancer has a history of experience working in these areas. Successful freelancers should include their past work in their application. Experience and detailed project proposals are not necessary for this project. In this case, I am only seeking assistance with residential properties.

I need a dotnet expert to help sorting one .rpt file error. this file helps one function that is to run Print function which is not working amd like this some 2 to 3 models are not working. Each time I try loading 'Module A', a 'Path not found' error emerges. Thus, I assume it pertains to path issues in loading static files. Ideal Candidate: • Proficient in DotNET ERP • Expert at troubleshooting path issues • Experience in environment-specific problem-solving • Ability to understand and work with static files. Your task will involve analyzing, identifying and correcting the root cause of this error to ensure smooth loading of 'Module A'. Please only pitch if you have the required skills and experience.

Excited to work with a talented photo editor who can breathe new life into my photo. Here’s what I am looking for: - The backgound needs to be adjusted so it is more white and uniform - Need to extend the background on the sides and on the top for better composition and usability as a header on linkedin and on facebook. The more seamless, the better. - Shadows need to be adjusted and clean up a bit and extened as well to extended background The ideal freelancer for this project should proficient in advanced photo editing software. Previous experience of working on similar jobs would be a huge plus. Looking forward to your masterful creations. Final work shoud be in DNG format - same bit depth.

I would like to change our "Dedicated Server " " product...our "Dedicated Server " " product page to have a layout more like the one of our competitor "Hetzner" We use the "hostlab" template from Themeforest : The idea is to have buttons on the top of the page like 'AMD Servers" "Intel Servers" "Budget Servers" "Storage Servers" Once clicked, these buttons would load the corresponding Server and details displayed in vertical columns instead of horizontal ones as it is now. Having the same king of drop down menu as hetzner to show/hide some features is also a plus.

Estamos à procura de um especialista em Issabel e Asterisk para implementar um sistema de Resposta Vocal Interativa (IVR) na nossa plataforma de servidor Issabel 4.0.0-6. O objetivo deste sistema é realizar entrevistas telefónicas automatizadas onde os respondentes possam responder a perguntas pré-gravadas utilizando o teclado do seu telefone. O sistema deve ser capaz de carregar uma base de contactos para realizar automaticamente as chamadas telefónicas e capturar estas respostas numa base de dados para análise subsequente. Além disso, o sistema IVR deve incluir a funcionalidade de conversão de texto em voz (TTS) para facilitar a criação e atualização de prompts de voz. No entanto, também d...

...solution, I'm seeking guidance from a skilled freelancer experienced with both Raspberry Pi and Asterisk server setup. Key Project Details: - I do not require a full-fledged PBX setup, just a basic VoIP service facilitated through my Raspberry Pi. - The primary feature I'm interested in implementing is voice calling. Required Skills: - Proficiency in configuring an Asterisk server on Raspberry Pi. - Strong understanding of VoIP and related protocols. - Ability to guide and explain the setup process clearly. Your Role: I've tried to set it up but no voice can be heard on the other end. So this task is mainly a trouble shooting job. Your primary role will be to walk me through the setup of Asterisk on my Raspberry Pi, ensuring proper configuration f...

I require an expert in Asterisk and FreeSWITCH development, along with experienced system engineering skills. Specifically, I need help with: - Asterisk and FreeSWITCH development - Integrating VoIP - Setting up Call routing and IVR (Interactive Voice Response) - CTI - Skill group base call routing to agents. Apart from these, the implementation of Asterisk and FreeSWITCH clustering as well as Call Center Reporting is required. The technology stack should be Linux, PHP, MySQL along with postgresql and API knowledge. The freelancer should have considerable experience in these to meet the project's specific requirements.

...persona con experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and recei...

Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error. [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descr...

Creazione di un software che da un lato dialoghi tramite AMI (Asterisk management interface) con un centralino e dall'altro in base a delle condizioni impostabili, invii del messaggi tramite un gateway whatsappa già pronto chiamato che permette di inviare (pagando) messaggi illimitati tramite WhatsApp. inoltre questo software deve poter inviare dei messaggi anche da una lista di numerazioni che gli si fanno caricare.

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

I'm seeking a seasoned virtualization expert to configure KVM and VXLAN with a principal goal of elevating network security levels. Your responsibility will include ensuring the safety of virtual machines running databases and web servers. Key...VM's without vxlan or with one particular vxlan can not see other vxlan's or can sniff traffic from other vxlan's. Ideal skills for the task: - In-depth knowledge of both KVM and VXLAN (openvswitch or alternatives) - Proficiency in configuring linux network and bridging. Environment: - KVM virtualisation (command line only, not UI) - Ubuntu 22.04 os - Baremetal servers have enough resources (AMD Epyc, minimum 32GB RAM) - Baremetal server each has 1 public network interface to gateway/public router and 1 private netwo...

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50

...into a compact 1-2 room layout - Knowledgeable in maximizing small spaces while maintaining functionality and comfort. We own a lot in a remote baja fishing/surf town, and have already chosen a general layout for two small casitas on the lot. One of them will have an outdoor kitchen and hang out area integrated seamlessly with an indoor kitchen and hang out space. We need help finalizing layouts amd exploring ways to make the project practical, playful and cozy. We have already assembled rough sketches, a lot plan, and dozens of photos of places we like. In your proposal, please share a brief summary of your experience including examples of architectural designs you've carried out, specifically any successfully completed designs that fuse Mexican and modern concepts in t...

We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors only w...

Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.

I am setting up new code from local host using Xampp, and move to a server with Directadmin panel. But i was failed when i migrate it to server The error in log is [Mon Apr 01 22:26:03.579055 2024] [proxy_fcgi:error] [pid 643251:tid 139908475770624] [client ] AH01067: Failed to read ...[proxy_fcgi:error] [pid 643251:tid 139908475770624] [client ] AH01067: Failed to read FastCGI header, referer: [Mon Apr 01 22:26:03.579121 2024] [proxy_fcgi:error] [pid 643251:tid 139908475770624] (104)Connection reset by peer: [client ] AH01075: Error dispatching request to : , referer: Thank you. My server specs : AMD EPYC™ 7402 RAM 64GB RAM Disk 80% Free I am using Apache and Nginx with proxy mode, php-fpm 8.1

I am urgently seeking an experienced telephony and data processing specialist to configure my Grandstream UCM6302A with Asterisk. The core functionality required includes receiving calls, playing a welcome message, meanwhile working with Caller ID and Web API to determine where to forward the call. When a call comes in, • first a welcome message is played () • in the meantime the caller ID will be sent to web API preferably POST, but get can be if POST is not possible () •The API will respond json array: - {forward_to: 33356853 } - Forward the call to 33356853 - {forward_to:0 } - Play message and terminate the call • If forwarded call is not answered by Agent in three rings, another call to API will

...am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Efficiency Goal: ...

i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much

I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.

...featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can provide the necessary Asterisk API documentation. Ideally, you are able to show me a s...

Hello, I operate a fax communication system leveraging Hylafax, integrated with an Asterisk server and iaxmodem, all running on Alpine Docker. While our outgoing fax functionality performs flawlessly, we are encountering persistent issues with incoming faxes. Specifically, incoming fax pages frequently get cut off midway, resulting in incomplete document reception. We are in search of a seasoned Hylafax professional who can diagnose and rectify this particular issue. Expertise in managing Alpine Docker environments and Asterisk/iaxmodem configurations will be highly regarded. Desired Expertise: Demonstrable experience with Hylafax, especially in fixing issues related to incoming faxes. Deep knowledge of Asterisk and iaxmodem. Proficiency with Docker containers, pre...

...seeking a VoIP consultant for improvement of my existing computer-based VoIP system. The purpose of the project is twofold - improved communication efficiency and enhanced call quality. Key Tasks: - Analyzing the current computer-based system setup - Implementing the connection of Physic SIP to asterisk on the cloud for enhanced call quality Ideal Skills and Experience: - Proven experience as a VoIP consultant - Excellent knowledge of IP PBX system - Experience with connecting Physic SIP to asterisk on the cloud - Ability to improve communication efficiency and call quality. Kindly submit your proposal outlining your plan to achieve these two goals along with your previous relevant work. Looking forward to finding a VoIP specialist who can provide a swift and efficie...

...an efficient Spanish and English speaking professional in Customer Support and Virtual Assistant, to work 40 hours per week monday to friday 9 to 5 (Colombia Time), that can manage: - email handling - Calendar Management - inbound calls - proficiency in Office Suit mostly in Excel - data entry - administrative tasks IT Requirements: A Desktop pc or laptop with: Processor: Intel Core i5 or AMD Ryzen 5 or better RAM: 8GB or more Operating system: Windows 10, macOS, or Chrome OS Noise-cancelling headphones Fast internet connection: Speed: 100 Mbps or more Type: Fiber optic or cable The VA should be prepared to work 40 hours per week. Proficiency in English and Spanish is crucial to effectively bridge communication between our global customer base and our team. Prior exp...

...developer experienced in WebSocket/AudioSocket technologies and Asterisk integration to develop a solution that enables real-time transcoding with OpenAI Whisper through gRPC. Requirements: 1) WebSocket/AudioSocket Integration: Develop WebSocket/AudioSocket functionality to facilitate real-time audio communication with OpenAI Whisper. 2) gRPC Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [a...

I am searching for a skilled software developer with a strong background in Asterisk, Dialer, IVR and VOIP technologies. Although I haven't specified particular functionalities, general familiarity with call routing, call recording and interactive voice response (IVR) would be beneficial. The ideal candidate for this job should be proficient in: - Designing, implementing, and maintaining Asterisk software - Developing dialer functionalities, with emphasis on auto dialing, click-to-dial, and predictive dialing - Ensuring system is up-to-date and secure Freelancers who apply should provide any past work, detailing their experience and including project proposals, if any. If you believe that you have the expertise to effectively take on this project, I encourage...