Jitsi asterisk trabalhos

Seeking an experienced developer to enhance our existing dongle voice SMS termination system with chan dongle and asterisk on Linux using custom firmware. The main focus will be on three key aspects: voice call termination, SMS termination as well as custom firmware development. While keeping the focus on the dongle voice SMS termination system, you will also need to utilize your Java skills to ensure seamless integration with our Linux server control system. Skills & Experience - Developing firmware - Working with chan dongle and asterisk on Linux - Java programming I invite applicants who can showcase both their past experiences relevant to this role along with any other skills and competencies that might be beneficial for the execution of this project.

I'm looking for a...someone with experience in Asterisk and SIP integrations. Key Project Aspects: - Asterisk Setup: Your task will be to set up Asterisk with multiple features including monitor calls + transfer + hang up calls, an IVR system, and call recording. I require a high level of accuracy and efficiency in the configuration of these features. - SIP Service Connection: Additionally, I need to connect Asterisk to a SIP service. We have an AI that speaks and give response to your recorded message. So the idea is to connect this AI to the Asterisk that is connected to the SIP so it can speak with people on the phone. Ideal Skills: - Experience setting up and configuring Asterisk, IVR systems, and call recording - Knowledgeable abou...

I am looking for a skilled Asterisk developer to customize and integrate my existing AI systems into Asterisk. Specific tasks include: - Integrating Asterisk with intelligent speech recognition and natural language processing systems. - Incorporating chatbot functionality for automated customer support. - Implementing AI-powered call routing and prioritization. The ideal candidate should have prior experience in Asterisk customization, AI integration, and development. Knowledge of speech recognition, natural language processing, chatbot technology, and AI-based routing is crucial for this project.

...PBX principles. In particular, your role will require the development of a Telegram bot, with integrated services improving the functionality of Asterisk. This will be a comprehensive project, necessitating precision and a thorough understanding of programming. Key features should include: - Both DTMF and Any Caller ID functionalities. - Implementations for these features should enable Call Recording. - The bot should also provide hold music capabilities, specifically using pre-recorded tracks. Ideal candidates will have a solid foundation in both Python and PBX systems. A background in telecommunications, specifically interacting with Asterisk systems, is also beneficial. Your expertise should encompass Bot development, particularly within the Telegram platform, as the...

In the search for an experienced developer, I require a web2call application based on Asterisk. This web2call should focus on enabling international call capabilities. Specifically, the created application should run adeptly on Linux platform systems. Additionally, the application needs to support one-on-one call functionality for its users. Hence, any freelancer bidding on this project should preferably have: - Proficiency in Asterisk software - Experience with building web2call applications - Knowledge of Linux platform compatibility issues - A background in creating systems for making international calls Your understanding of one-on-one call functionality support will be highly valued in this project. This project will be a significant contribution to connecting people ...

Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

Requiero la instalación y configuración de Festival y Cepstral en varias voces en español, así como hacer la configuración necesaria para que las grabaciones que se están grabando en WAV se graben en wav, todo lo anterior en asterisk 11.25 en Issabel.

...of Work I will build website and theme - DONE I will install Bookly - DONE I will install Bookly deposit plugin - DONE I will install Bookly STRIPE plugin - DONE - Connect - PENDING I will add funds to SMS feature - PENDING I will create a new Gmail email for the Website to use for bookings - DONE I will also connect it to the google Cloud Calendar like the last time - DONE Work for you 1. Get Jitsi working with Bookly 2. Video Link in the email notification (like last time) 3. Sms - Also add the Video link to an SMS (we can test with our numbers, i will load credits) Deposit System: 1. Deposit when you create appointment 2. Remaining amount payment after video meeting. Extra notes/questions:(i will help with this) - change bookly appointment date selections to be hours only a...

I'm searching for a proficient SIP/Asterisk engineer who has ample experience in performing the following tasks: - Setting up and configuring an Asterisk server - Ensuring that the server is set up for high security and can support telemarketing or sales calls - Adding a specific function of calling and spoofing numbers, to be used exclusively for telemarketing and sales purposes Ideal skills and experience for the job: - Extensive experience with Asterisk and SIP protocols - Past-experience in setting up IVR systems - Understanding of server security measures - Familiarity with telemarketing systems Please provide a summary of your relevant work experience and your approach to this task in your bid.

I resolved issues with developed pbx based asterisk with dial plan and fax over sip

...real-time speech-to-text functionality within the Asterisk ARI, leveraging the snoop channel feature. This project's goal is to create a seamless and accurate method of converting spoken words into text in real-time during calls, focusing on English language conversations. **Requirements:** - Proficiency in developing with Asterisk ARI and understanding of snoop channel operation. - Strong background in AI and speech recognition technologies, specifically tailored for real-time applications. - Solid experience with handling real-time audio streams and converting them to text. - Capable of integrating speech-to-text solutions in a Linux environment; though the operating system choice was skipped, Linux is our preferred platform for running Asterisk. - Ability ...

Necesito instalar jitsi meet en una VPS bajo ubuntu . En este servidor estàn corriendo diversas web bajo litespeed y quiero que en uno de estos dominios pueda correr jitsi meet para organizar videoconferencias, webinars, etc. Los recursos que tenemos son limitados y zoom nos parece prohibitivo. Sería una instalación sencilla en la que hubiera diversas funcionalidades en este tipo de páginas: chat, moderador, posibilidad de subir archivos por chat, capturar información del usuario (mínimo nombre, tlfo, correo-e, problema en el que está interesado...), grabar la videoconferencia... y a partir de aquí, me abro a las ideas que tú también puedas dar. Un saludo esperando tu respuesta.

Hello Asterisk Developer - Engineer - Tech, 1. I need an Asterisk auto-attendant - IVR with a TTS text-to-speach greeting generator - plugin. Canned greetings are so last century - customers want tailored greetings in a life-like male and female voice.2. No cumbersome GUI to handle this. I currently have a GUI running my Asterisk Voicemail that is over 10 years old...simple, liteweight and has what I need.3. If you are not experienced with Asterisk, you probably should not you.

...I am seeking a developer with experience in Asterisk PBX and PHP to create a custom dashboard that interacts with the Asterisk Manager Interface (AMI). The dashboard should allow real-time monitoring and management of calls, as well as generating reports and statistics on system usage. Required Features: Real-time visualization of incoming, outgoing, and ongoing calls. Ability to initiate, pause, and terminate calls from the dashboard. Generation of reports and statistics on call traffic, duration, etc. Integration with Asterisk PBX administration functions via AMI. Intuitive and user-friendly interface. Required Skills: Demonstrable experience in web application development with PHP. In-depth knowledge of Asterisk PBX and its ...

nesecito crear una configuración en Asterisk la cual todas las llamadas salgan por trunk diferentes estas de forma aleatoria., este proyecto lo nesecito lo mas rápido posible. nota: este plan de marcado tiene que tomar el calleid de cualquier plataforma o aplicación y asignarlo como callerid a la llamada saliente. le dejare adjunto lo que intente, sin suerte, porque no asigna el callerid

Experiencia en desarrollo y mantenimiento de sistemas con Asterisk, FreePBX, Issabel. Dominar el idioma español.

I am experiencing an issue with the Asterisk 13 Audio upload and volume control API, particularly the volume control function. It is unresponsive and isn't altering the volume at all, despite my attempts to modify it. I've tried rectifying this issue by using alternative extensions to manage the volume, but with no success. The ideal freelancer for this project would have a strong knowledge and previous experience with the Asterisk 13 API, and a solid understanding of volume control functions. Troubleshooting and debugging skills are essential. The goal of this project is to ensure that the volume control functionality works impeccably, and responds accurately to adjustments made.

Completing this project demands expertise in Kubernetes, Jitsi, and infrastructure integration – backed by a keen understanding of high scalability, low latency, and high availability features. The successful candidate will be tasked with: • Incorporating Jitsi Video Bridge (JVB) with high scalability, high availability and OCTA setup. Your skills to ensure performance and reliability in a complex network environment will be crucial. • Implementing automatic scaling on the Jitsi Video Bridge (JVB) based on user load. This requires a deep understanding of dynamic resource management and load balancing. • Integrate the TURN server with our current infrastructure. This calls for skills and experience in networking and system integration. Our focus ...

I require an expert to install and configure Asterisk on my Linux system. Specifically, I need the setup to efficiently manage and handle a maximum number of concurrent calls between 21-30. Given that I haven't specified the exact end-user features, my preference is for a freelancer who has comprehensive experience in deploying diverse functionality on Asterisk systems. Key skills: • Strong experience in setting up and configuring Asterisk on Linux platform. • A solid understanding of developing and managing Asterisk features. • Familiarity with call concurrency and optimization for 21-30 concurrent calls. I am looking for a competent freelancer who can ensure a seamless and efficient setup to meet my project demands. If you've got t...

...solution, utilizing both FreePBX and Asterisk servers to streamline incoming, outgoing, and call monitoring processes. Here's a glance at my requirements with the BSNL SIP Direct Truck : 1. FreePBX AND Asterisk Server Setup: I am looking for an experienced tech consultant who can not only advise but also assist in setting up both FreePBX and Asterisk servers. 2. Custom Bot Development: The bot should be responsible for calling, receiving, and monitoring calls directly from the software with features such as: - Automated call routing - Call recording - Call analytics - Call forwarding - Interactive Voice Response (IVR) - Real-time call monitoring Ideal skills would include a deep understanding of VOIP technologies, especially FreePBX and ...

I am looking for an experienced developer proficient in Asterisk, particularly its API. The role involves integrating an established frontend UI web app with a backend PHP and Asterisk API. Refer to the attached document for comprehensive details. Only respond if you possess strong Asterisk and web development skills. Thank you!

I'm seeking a skilled developer for a crucial integration project involving Asterisk, ERPNext, and Vodaphone API. My main goal is to bolster my customer service capabilities. I'll need an expert who can blend these components seamlessly, ensuring a smooth workflow and real-time interaction that elevates user satisfaction while streamlining my business processes. What I require: - Implementation of Call Recording: Capturing and storing every customer interaction. - Intelligent Call Routing: Directing calls smartly for optimal customer service response. - Detailed Call Analytics: Insights into call patterns to refine our communication strategies. - Real-time Data Synchronization: Keeping our ERP system and telephony platform in perfect harmony, to reflect updates instantly...

We are seeking an experienced freelancer to help integrate Asterisk with Node.js for our communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP...

I'm looking for an experienced full-stack developer knowledgeable in Asterisk and React. The project scope involves building a VOIP system focusing on Asterisk. We need to integrate a vital functionality: call routing. This system will mainly be dealing with my business's customers. Hence, it's essential for it to be user-friendly and have a robust functionality. Ideal Skills: - Proficient in Asterisk and React - Previous experience in VOIP system development - Experience in customer-focused application development - Expertise in implementing call routing functionality This role requires technical and domain skills. You should have a strong understanding of the best practices in Asterisk systems and be able to efficiently implement a call...

I'm seeking an experienced developer proficient in Asterisk configurations and AGI scripting to enhance our customer support telephone system. **Essentials:** - AGI scripting for Asterisk - AI Voice Response integration **Key Goals:** - Develop an AGI script to automate voice responses for incoming and outgoing calls. - Integrate the system with openAI APIs to deliver natural, engaging voice interactions. **Ideal Candidate:** - Previous experience with Asterisk telephony systems - Strong proficiency in AGI script development - Familiarity with AI-driven voice response technologies - Proven track record with APIs We need to transcribe caller's voice and send it to openAI via api and get responses from there and convert it to voice again and play it ou...

I'm looking for a skilled Asterisk developer who can create an effective call tracking and routing system. This task will be from scratch, so you will not have any previous resources or guides to reference. Key project requirements: - Configure and implement call routing and forwarding -Develop GUI in react, backend in MySQL - Strong PHP skills are required to handle this task successfully - Proven experience in Asterisk and telephony systems Ideally, you should have a substantial background in setting up telephony systems from scratch. Demonstrated ability to solve complex problems in this area is highly desirable.

...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 am seeking a skilled developer to construct a simple call forwarding system using Asterisk. Key functionality required includes forwarding calls to several numbers. This function should be dynamic, with the ability to add or edit the numbers to which calls are forwarded via a user-friendly web interface. Moreover, the web interface should give users the power to customize the call routing settings. Ideal candidates should have significant experience in VOIP technology, Asterisk systems, and developing intuitive web interfaces.

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

I'm looking for an experienced software engineer or IT specialist who can assist me by setting up real-time voice transcription on the VitalPBX platform. Key requirements and details of the project: - Integration of Google Cloud Speech-to-Text API: As I have already started this process, the successful freelancer will need to continue this work and ensure a successful integration. -...needs to be configured in such a way that it activates during multi-user conference calls. Your expertise in real-time voice transcription systems, particularly on VitalPBX, and thorough understanding of Google Cloud Speech-to-Text API will be crucial to the success of this project. I need it real quick and the idea is simple, user A calls user B and the conversation should be displayed in as...

We are seeking an experienced freelancer to help integrate Asterisk with Node.js for our communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP...

...solution It must install on an Ubuntu Server environment. This must be an opensource solution. The following client environments must be able to interact with it: * Desktops - Windows/Apple (Linux maybe?) * Mobile: Apple IOS / Android The chat software must authenticate to our LDAP server. We are using Samba 4 Active Directory / Domain Controller For the voice / video portion, it must use Jitsi technology. However the chat software is where user authentication will initiate and chat software will be the first / initial interaction between users. Once they are engaged in chat they will have the ability to engage via voice or video. When completed you must provide clear step by step documentation / instructions on how to install and configure the end product on our ser...

Project replay due to account issue I am looking for a skilled Android developer to create an application that streams video from a USB external camera. This app should also integrate the Jitsi meetings SDK and provide additional features for Jitsi meetings. Key Requirements: - The ability to stream video from a specific USB camera model. - Integration with the Jitsi Meetings SDK. - Additional features such as using an external USB (OTG) camera in the Jitsi Android client. Ideal Skills: - Proficiency with Android app development and understanding of video streaming. - Experience with the Jitsi SDK. - Familiarity with integrating USB hardware with mobile applications. Knowing that you have used a specific USB camera model with Android apps before ...

...create exporters for custom metrics when needed and represent the data in grafana dashboards. Grafana should be embeded in the existing web app either as public dashboards or iframes. Security concerns are a MUST. Things to monitor: system stats (node exporter): cpu, ram , disk, network... usual processes like sshd, cron, systemd-timesyncd, rsyslog... solution specific: mysql, nginx, kamailio, asterisk,nodejs, php-fpm, redis-server... Own developed services need some custom metrics. We can provide scripts that will return the data when executed. Data is meant to be presented to non-it people as well as it people. Meant to be visually atractive and easy to understand. Less easier graphs rather than many complicated ones The deployment in the product must be via configuration f...

I am looking for a freelancer who can assist with configuring my Issabel Asterisk system; especifically OUT AUTO DIALLER The specific configuration changes I need include: - Dial plan modifications - Voicemail setup - Call routing adjustments I have a specific deadline for completing the configuration, so it is important that the freelancer can work within that timeframe. I am already familiar with the system, so I do not require additional training or documentation on maintaining the new configuration. Ideal Skills and Experience: - Strong knowledge and experience with Issabel Asterisk system - Proficiency in dial plan modifications, voicemail setup, and call routing adjustments - Ability to work within a specific deadline If you have the necessary skills and experienc...

I am looking for an experienced freelancer to configure my Issabel Asterisk system. The project involves setting up specific features and implementing a detailed call flow. The ideal candidate should have expertise in call routing and be able to work with a call volume of less than 100 calls daily. Skills and experience required for this job: - Proficiency in Issabel Asterisk configuration - Strong knowledge of call routing - Ability to implement a detailed call flow - Experience working with a low call volume environment If you have the necessary skills and experience, please submit your proposal outlining your approach to configuring Issabel Asterisk and your previous relevant projects.

I'm currently in search of an experienced developer who can deliver a seamless integration of personalised ID setup on my existing VoIP services. I currently use 3CX, Asterisk, Vicidial, and Eyebeam for my business communications. Key Requirements: - Significant experience working with VoIP services including 3CX, Asterisk, Vicidial, and Eyebeam. - Proven expertise in VoIP system integration. - Ability to test for and ensure smooth communication across all platforms upon completion. Candidates with a strong background in VoIP software/platforms and successful project completions in similar setups will jump to the top of my list. Looking forward for a professional with a solid track record who can deliver an efficient solution.

I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

Need help to be able to set up an Android app(preferably React Native) and Server software runnng on my laptop that uses open source XMPP(example Tigase), WebRTC(example Jitsi) to demonstrate 1. Voice call, Video call, Group Voice and Video calls, Push to Talk, Push Video 2. Chat, Group-chat, Audio Message, File sharing. Video Message These applications should be usable in a Webapp also. Deliverables required 1. Opensource and any custom Source code required 2. Installation on my laptop and demonstration 3. Documentation for installation and configuration. 4. Support me in build the Android APK.

Requirement: Video chat integration into my existing system. - Platforms: Some options are Zoom, Skype, Google Meet, Agora, Jitsi, Twilio etc depending on what suits us the best. - Maximum Participants: The video chat function should be tailored for a virtual fair context, accommodating initiation of chats with a flexible number of customers. Ideal candidate possesses strong experience with several video chat platforms

***Do not bid outside the budget I specified!!!!*** Adding native jitsi video & audio calls for android (java) dating app. Codes can be kotlin/java. Jitsi 1to1 calls will be made. Sample codes on Github can be applied. A maximum of 2 hours of work for a knowledgeable programmer. Please do not make costs that will take days. Do not make offers outside my budget range!!! The fee is paid at the end of the work. No interim payment is made! Price update is rejected! Sample: Main App:

I am looking for a skilled Android developer to create an application that streams video from a USB external camera. This app should also integrate the Jitsi meetings SDK and provide additional features for Jitsi meetings. Key Requirements: - The ability to stream video from a specific USB camera model. - Integration with the Jitsi Meetings SDK. - Additional features such as using an external USB (OTG) camera in the Jitsi Android client. Ideal Skills: - Proficiency with Android app development and understanding of video streaming. - Experience with the Jitsi SDK. - Familiarity with integrating USB hardware with mobile applications. Knowing that you have used a specific USB camera model with Android apps before will significantly enhance your candida...

We are seeking a skilled freelancer with expertise in Asterisk to assist us in implementing outbound call functionality to a mobile number using PJSIP. The current channel originate command is not working. and we are open to alternative methods that successfully initiate calls. Earlier, when we used older version of asterisk 18.1, we can get calls but newer version doesnt have that command and it gives below error pi*CLI> originate sip/ application Playback hello-world No such command 'originate sip/ application Playback hello-world' (type 'core show help originate sip/' for other possible commands) We are using asterisk 20.5 and system is rpi with ubuntu 23 installed. Thank you!

I'm seeking an expert with a deep understanding of Asterisk and Magnus Billing Portal to resolve a critical issue we're facing. Our system currently fails to establish connections for incoming calls right at the beginning of the call process. This persistent problem is resulting in missed communications and potential business losses. Ideal Candidate: - Proficient with Asterisk PBX systems - Experienced with Magnus Billing setups - Strong diagnostic and problem-solving skills Scope of Work: - Analyze and pinpoint the cause of the call connection failure - Troubleshoot and fix the routing issue to ensure call connectivity - Test and confirm that the issue is fully resolved Timeline & Communication: - Immediate availability preferred - Consistent updates require...

Stick to your offer. Do not bid if the price range does not match. A java dating app will add a jitsi video and voice chat feature. That's it. No details. Because this is the work. I think there is no need for any other details.

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

Remove DND Button Add Login/Logout Button (Asterisk Queue) Add Pause/Unpause Button (Asterisk Queue) Add our logo Show Waiting Calls in Queue Show Agent Pause Status Show Agent Pause Count Show Agent Pause Time

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

We need to connect our CRM with our phone system in place. We need integration between Zoho CRM and WAZO (Asterisk 15) Features: - Click to call from Zoho Interface: callback user extention + keep time duration - Pop up when incoming call from WAZO and open customer record or if customer is not registred, open record creation and keep call duration It worked with ZOHO Phone bridge before annd it worked proprely but ZOHO CRM stopped the service few years ago. Thank you

Responsibilities: • Develop and maintain Asterisk-based solutions for real-time transcription in an AI call center environment. • Integrate Whisper API for seamless speech-to-text and text-to-speech functionalities. • Implement real-time transcription capabilities for incoming customer calls. • Collaborate with the AI team to integrate intent recognition models for dynamic call routing. • Ensure smooth communication between Asterisk, Whisper API, and internal AI systems. • Optimize system performance to meet real-time transcription requirements. • Troubleshoot and resolve issues related to speech-to-text and text-to-speech integration. • Stay updated on advancements in Asterisk and speech recognition technologies. Requi...