Avaya sip asterisk työt

We are in the early stages of creating an AI-powered call center service tailored for businesses in Egypt, such as restaurants and car dealerships. Our goal is to leverage GPT-4 integrated with AI text-to-speech and speech-to-text models to provide seamless, efficient, and cost-effective customer service solutions. We are seeking an experienced ...Frameworks (React, Angular, Vue.js) Node.js, Django AI and Machine Learning Libraries (TensorFlow, PyTorch, Hugging Face's Transformers) Cloud Platforms (AWS, Google Cloud, Azure) API Development Docker, Kubernetes CI/CD Tools (Jenkins, GitLab CI/CD, GitHub Actions) Web and Data Security Practices Problem-Solving Skills Communication Skills Project Management Skills Telephony Integration (VoIP, SIP) Speech Processing (Speech-to-Text, ...

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.

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'm searching for a network security expert to assist in diagnosing a problem with my SIP ALG settings on my Cisco ASA5506 firewall. It's crucial because I am experiencing an issue wherein the phone rings for one second and then cuts off. Your tasks will include: - Examining the SIP ALG status on my firewall. - Troubleshooting any issues found and offering solutions to rectify them. - Implementing the solution with my approval. Relevant Skills and Experience: - Proven experience with Cisco ASA5506 firewalls. - Deep understanding of network security. - Skills in VoIP technologies and SIP ALG. - Familiarity with various VoIP service providers/software is a plus. Please note, we are not currently using Skype for Business or Ring Central, but a different prov...

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

I will install Freepbx on a local server and in Digital Ocean Cloud. I will configure it for 10 users and 3 trunks, while also ensuring that configurations can be synced between them. Finally, I will secure it against external threats and SIP flood attacks.

...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 ...

I need an imaginative designer to create a logo for my Italian Bistro, SipsAndSpices, where flavor meets elegance. Our bistro offers a curated selection of fine wines, craft cocktails, and globally inspired dishes. With a warm ambiance and attentive service, we invite you to savor the exquisite blend of flavors and sip on carefully chosen libations. I am open to all styles, be it minimalistic, vintage, modern, or a unique blend of these. Color is also your playground; creative use of blue, green, yellow, or any other shades are welcome. Just bear in mind that the logo should reflect an Italian restaurant vibe. Skills and Experience: - Proficiency in graphic design software. - Knowledge of effective color schemes in logo design. - Ability to design within different styles. ...

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'm seeking an expert Android developer to create a VoIP app with GSM integration. This pro...developer to create a VoIP app with GSM integration. This project includes, but is not limited to: - Developing an app for calling and text messaging through a SIP server, inclusive of delivery reports. - Implementing features such as call recording and instant messaging. - Ensuring compatibility with older Android versions. Since I don't have a preferred SIP server, your input and recommendations would be highly valued. Experience with SIP servers and a strong understanding of both VoIP principles and Android development are necessary for this project. Your role will be pivotal in guiding the selection and integration of the SIP server. Please consider the ...

Salut, Pe lan 1 avem reteaua locala calculatoare si telefoane voip si internet Pe lan 2 avem acces la serviciul digi sip care la care sa conectat dar nu mai vedem telefoanele din retea initial vedeam telefoanele ba chiar am gacut si teste cu un modul GOIP care a functionat bine .

...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 ...

More details: What type of engineer do you need? Structural engineer Could you please clarify what specific applications of SIP panels are you interested in? Wall system, Roof system, Floor system What are the approximate dimensions of the cottage? Small (less than 500 sq ft) , 12 feet wide by 20 feet long ICF blocks for foundation and walls. Sip panels for floor, Gable ends for roof and roof engineering for attachments See attached file to read details

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

...essential configuration changes to my existing Issabel IP PBX setup. The project specifics include: 1. Changing the current SIP provider from Tata to Airtel. 2. Updating IP PBX configuration settings to seamlessly support the new Airtel SIP service. 3. Making necessary modifications to the SIP trunk settings to ensure optimal compatibility with Airtel's service. 4. Configuring additional SIP trunk registrations as required. The ideal candidate should possess proficiency in SIP trunk configurations, with a solid track record of experience in working with both Tata and Airtel SIP services. A clear understanding of Issabel IP PBX setups and the ability to configure SIP settings accordingly are crucial for the success of this project. ...

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

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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.

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...

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...

...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 S...

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.

Hello We have an AVAYA system and we need to add some features to it. Please have a look at the attachment Thanks

instale astpp basado en astpp, el problema que presento es que luego de haberlo instalado en debian 10 no me permite realizar conexiones sip, cabe destacar que desactive el firewalld y fail2ban para ver si era el problema y aun no me permite realizar conexiones

Sales :- • Conduct market research to understand the financial needs and preferences of target customers. • Prepare sales materials, presentations, and proposals to engage with prospective clients. • Source new clients and get repeat business from existing clients. • Sell Investment products I.e. Mutual Funds SIP & Lumpsum, Bonds, FD’s.

I am looking for a Python expert, ideally with experience in VoIP, Call-center's Dialler softwares and SIP protocols. I need help in building a python application(just a script/server side python app) that will connect to VoIP/Dialler and get the audio stream for call and reply with an audio stream. I've a Dialler(MicroSip) setup on one of my servers, I have used the JS example and LinPhone application to make calls between 2 apps and it works fine. What i want to achieve is: Write a python program that should recieve, and make calls using computer's Michrophone and Speakers and should have the option to add anyother audio stream in reply to a call instead of getting the audio from system's michrophone. I've gone through plenty of libraries but as i am no...

...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...

...hardware - Implement functionality for sending commands to a remote M5Stack device over LTE - Enable the M5stamp LTE-M to connect to a web server and display online status - Integrate SIP/voice call capabilities from a PC to the M5Stack LTE-M remote device using the web application - Allow AT commands to be sent from a local PC application to the remote M5Stack Stamp-M Key Skills Needed: - Proficient in WebSockets for real-time communication - Experienced with Arduino coding specifically for M5Stack Stamp - Skilled in web application development, preferably with a focus on user-friendly interfaces - Familiarity with SIP protocol for voice communication over the internet - Ability to manage AT commands and their integration in software - Understanding of real-time data mon...

I'm seeking assistance in modifying an existing Python3 code to listen to incoming calls on a SIP server. At present, the code can only make calls. The ideal candidate for this task is experienced in Python3 and SIP server management. If your previous work includes similar projects, it will be beneficial in placing your bid. The necessary modifications include: - Adding a call recording feature - Implementing a call forwarding functionality - Integrating a call logging system While applying for the task, please consider that sharing details about your experience and past work is appreciated. Freelancers who can deliver the project quickly will be given preference. Nevertheless, I'm more interested in quality work, so feel free to negotiate a reasonable deadli...

...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 S...

...around fixing bugs in an Android app, specifically problems with its SIP functionality. The key issues I'm experiencing are: - The SIP registration automatically stops. This tends to happen when the app is runing . - Certain SIP functions related to incoming and outgoing calls and call quality issues are not working as expected. An ideal candidate would possess the following skills and experience: - Proficiency in Android app development - In-depth understanding and experience of VOIP and SIP communication protocol - Experience in troubleshooting and resolving SIP registration issues - Experience in debugging and problem-solving SIP related call issues. Your primary task will be to fix these bugs to ensure that SIP registratio...

...Key Features Required: - Call Recording: Ability to record voice calls with clarity. - Video Calling: High-quality video call functionality. - Voice Encryption: Ensuring secure, encrypted voice communication. - Real-time Language Translation: Feature enabling audio SIP calls with instant language conversion. User Interface Design: - The interface should support customizable themes, allowing users to personalize their experience. Ideal Skills and Experience: - Proficient in cross-platform software development. - Experience with SIP protocol and audio/video encoding. - Knowledgeable in implementing encryption algorithms for voice security. - Familiarity with implementing real-time language translation in applications. - Expertise in creating user-friendly, customizable UI/...

We are using SIP account using Centos for Linphone ..TCP already configured however with 5g connection sometimes calling doesn't work. We need an expert who can setup/Enable TLS on it. Linphone sdks are installed on both iOS/Android and backend.

I'm seeking an experienced tech to setup both inbound and outbound trunks on my FusionPBX system. Please note this task involves setting up Inbound/Outbound Trunk using our Provider's IP and also you need to provide us step-by-step instructions to perform this in future. The ideal candidate should have: - Extensive experience with FusionPBX. - Solid understanding of SIP trunking, as I am currently using a SIP trunk provider. - A proven record of successfully setting up both inbound and outbound trunks. I am expecting a smooth configuration process with minimal downtime. Please ensure your bid reflects your understanding of these requirements. Maximum Budget - INR 3500 / $45 USD

Looking for a good software developer for an ongoing project to develop a robust softphone app that will connect to a SIP PBX and be able to make and receive calls. - Register to a sip server - Make and receive calls - Call History: The app must track and display complete call logs. - Contact Management: Users should be capable of managing their contacts within the app. - Voice Messages: In-built feature allowing users to send and receive voice messages is required. We look to start with one OS first, most likely iOS. Then if the project is successful we look to develop the same app for Android. If both are good we would like to include Windows. Ideal Bidders: Developers with experience in VoIP app creations, cross-platform development and understanding of VoIP service pr...

...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...

un sistema de llamadas masivas con troncal SIP

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.