Asterisk meetme mute jobs
I'm in need of a knowledgeable and experienced Asterisk expert for the implementation in our company We need an Asterisk specialist to help us set up a carousel of different numbers and provider accounts. The essence of the task, when calling, automatically should be substituted random number from our list. Details and scheme will be discussed individually
I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.
I'm seeking a proficient Asterisk developer who can build a custom PBX system with Issabela or something comparable. The PBX system requirements are as follows: - Less than 10 extensions. - Features such as call recording, auto-attendant, and call routing. - Compatible with VoIP phone lines. Ideal candidate should possess a deep expertise in Asterisk and a good understanding of VoIP technology. Knowledge of Issabela or similar software is an added advantage. Let's connect if you can guarantee a seamless and efficient communication system. For more information, sent the requerimients - The calls received would be answered by several receptionists who would log in/unlog in with their landline to receive the call. If there are more calls than receptionists, a message w...
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...
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...
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 ..
...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.
You Have To Cutout Sexual Scenes From My Videos And I'll Give You Per Video 8₹. 1 Video Take 15 Minutes To Edit Proper. Videos Requirements:- 1. Size 1:1 2. No Water Mark 10 Second no copyright Music With Best scenes but hidden all private parts. 4. See sample video before making videos. 5. Make same like sample video. 6. Video length 3-4Minute 7. Mute all sex sounds like (fuck,boobs,uhh,aah,ooh,fuck me) sound mute please. 8. Video must be 1080p 9. Subtitles must be included don't remove that. ✅ Done I'll Pay You Weekly 800-1000₹ For 150 Videos. Depends On Your Editing. If You Can't Make Videos On This Price Then Don't Bid.
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 have a 30 sec mute video in hindi .I need someone who can read lip moment in hindi
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 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'm in need of an electronics engineer who can develop a custom two-way intercom system for a medical pressure tank, for communication with patients. indoor use only. The system should have the following features: - Allways communication (from inside to outside) - Push-to-talk from outside to inside (will mute the other direction) - Ability to mute all together - Ability to buzz from inside, for the outside speaker (tone generator) - 2 Condenser Mic Inputs (nice to have: brakeout bypass of Condenser Mic circuit for other microphones) - 2 Speaker Outputs (50W, per output) - individual speaker Volume control - control signals are all 24V with comman GND - 0-24V PSU Ideal Skills: - Electronics Engineering - Circuit Design - PCB Design - Audio Systems T...
Objective: Create a dynamic and visually appealing pop-up for the Lignum website () to promote our new product, WoodXEL. The pop-up should feature a compelling video to engage visitors immediately. Details: Video: into stunning, personalized wooden art. Perfectly blending modern technology with traditional craftsmanship." Call-to-Action (CTA): A prominent button labeled "Create Your Panel Now". Design Elements: Use colors and fonts that reflect WoodXEL’s branding (). Ensure the design is clean and professional. Functionality: Responsive design, autoplay video without sound (with a mute/unmute button), and an easy way to close the pop-up. Final design files. Mock-up showing the pop-up on desktop and mobile.
I need a skilled Discord developer to help me setup my server and create custom bots with specific functionalities. Requirements: - Server Setup: You will be responsible for the initial setup of the Discord server. - Custom Bots: I need custom bots to be created with the following functionalities: - Moderation: The bots should be equipped to perform actions like kick, ban, and mute. - Auto-moderation: The bots should prevent spam. - Custom Commands: Implement custom commands as per my requirements. Additional: - Integration: The custom bots need to integrate with Twitter and TikTok. Ideal skills and experience: - Proficiency in Discord API and server setup procedures. - Experience in creating custom bots with moderation and auto-moderation capabilities. - Familiarity with ...
...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...
...web-based media players - Knowledge of modern UI/UX design - Previous work with live audio streaming. I want the radio player to support icecast / shoutcast and allow users to switch between multiple radio stations. Each radio station should have: * Meta info such as whats currently playing on air like images, song name, artist name, podcast name. * name of station and location * Pause, play, mute, volume, * What previously played * Whats playing next (small upcoming embed schedule) AN API will likely need to be created, we will pay extra for api to be created. An api is needed for meta whats playing from radio stream. I want the design to have header and footer. I do want back-end admin could be simple UI or a text file to control player such as adding, deleting, editing...
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...
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.
...and detail-oriented individual who can accomplish a specific task for me. With your proficient Windows operating system skills, browsing acumen, and technology savviness, you will navigate a particular website while a screen-record captures the process. Your job will simply entail: Deliverable: Two screen recording videos (maximum of 5 minutes each) where you act as a user. Important Notes: • Mute your microphone while screen recording. • Do not reveal your identity. If you log in from your account, cut that part from the screen recording. • Do not reveal your location. Choose a random location within Australia. • Do not complete the payment on the site during checkout. Website 1: Moon Phases by WillyWeather Task 1: Current Weather Check: • Scenari...
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
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...
...automatically play on mute, and vice versa. No play or pause button should be visible; the video should play automatically when it appears on the timeline. If the video is initially played on 'mute,' and the user taps on the video to watch it as a reel for the first time, it will play with audio. However, for subsequent views, if the user mutes the video and then taps to watch it as a reel again, it will play on mute consistently. Timeline Interaction: User scrolling should stop the audio of the previous post automatically. Once a video on the timeline ends, a click on the video should allow the user to watch it again in the reels. While a video is playing on the timeline, a user can click on it to enter the reel session, and a back button must be provid...
...analyst navigating spreadsheets and avoiding office politics; the next, he's thrust into an ancient realm of magic and mayhem, summoned by a desperate sorcerer seeking a champion to save their world. But Silas? He's anything but a fighter. His battles are against deadlines, not dragons. Yet here he stands, chosen to be a guardian in a world that's as beautiful as it is deadly. With the help of a mute child, who is as deadly as she is silent they must navigate treacherous terrains while learning to master his new abilities in order to survive. True power isn't just about mastering your weapon, but also mastering yourself. With an ancient evil stirring and the fate of the realm in his unsteady hands, Silas must confront his fears, embrace his destiny, and disc...
...multiple domains for me. The bot's duty would be to monitor the websites' state via three simple APIs(also checking downtime state and response time), with the ability to send different notifications as Telegram messages to the subscribed users. The frequency of alerts and other variables should be customizable. The bot should have UI menu to manage (add, delete domains, change customized variables, mute notifications for some domains or change monitoring mode) Alerts should be dispatched at specific times and also triggered by certain events. Experience with Telegram bot programming, API integration, and alert systems is essential. Key Requirements: - Monitor multiple websites' states using simple APIs - Send alerts to subscribed users (whitelisted in the in...
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...
...minimal effort. The plugin will support various customization options for video embedding, ensuring flexibility and compatibility with different website designs and layouts. Features - Easy Video Embedding: Users can embed PeerTube videos by simply inserting a shortcode or markdown into their page content. -. Customizable Embed Options: Provide settings for video width, height, autoplay, loop, and mute options, among others. . PeerTube API Integration: Fetch video details and generate embed codes using PeerTube's API. Responsive Design Compatibility: Ensure that embedded videos are responsive and work well on devices of all sizes. - Privacy-Focused: Support for privacy-enhanced options that limit tracking by third-party services. Documentation: Comprehensive user and develo...
I'm currently facing a challenge with my Instagram videos. After posting, they immediately go mute for reasons I can't understand. These are finance-related videos that are important for my content strategy. Thus, I need skilled assistance to resolve this issue and monitor my future posts to prevent recurrence. Additionally, I require help with posting 4 videos per month. Ideal qualifications for the job: 1. Proven experience in managing and troubleshooting Instagram content – especially videos. 2. A strong understanding of Instagram's guidelines and best practices. 3. Background knowledge or interest in finance is a plus. 4. Excellent problem-solving skills. In summary, I'm not only in need of an immediate solution but also seeking long-term assistance...
...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...
...signup process (optional, consider social media login integration). Request permission to access webcam and microphone. Optionally: Allow users to set basic preferences (age range, interests - future expansion). Matching & Video Chat Room: Utilize efficient algorithms for rapid user matching. Integrate high-quality video chat functionality with clear audio and visuals. Provide basic chat controls (mute microphone, turn off camera, end chat). Optionally: Offer text chat alongside video chat (future expansion). Safety & Security: (Optional): Age verification system to deter minors (depending on your target audience). Clear reporting mechanisms for inappropriate behavior. Outline user privacy and safety guidelines....
We need to create an Asterisk aplication (v18) for Service at workshop by appointment for vehicles. This aplication must have voice recognition in English /Spanish language and Text to speach language with Google technology. Functionality: i) Welcome. ii) Select Languague. iii) Request Data: * Type of vehicle * City * Car licence plate * Telephone number. * Date request. * Time request. d) System will confirm first date/time available and customer will confirm. At this time, application will not have conectivity with real system....only must confirm next day and time users told. But it will have errors control, confirmation recognized data, etc.....
...engineer to implement an Opus encoder and decoder in C# for seamless integration with Asterisk. The project involves handling voice audio from Asterisk, decoding it, incorporating Text-to-Speech (TTS) functionality, and encoding the synthesized speech before sending it back to Asterisk. The ideal candidate should possess the following skills and experiences: - Proficiency in C# programming language. - Extensive experience with audio processing and Opus codec. - Familiarity with Asterisk, SIP, and IVR systems. - Knowledge of Text-to-Speech (TTS) integration. - Ability to deliver high-quality code within specified timelines. Main Tasks: - Implement Opus encoder and decoder in C#. - Integrate the solution with Asterisk for audio processing. - Incorporat...
In our quest for an outstanding and interactive VoIP web app, we're in need of an experienced developer, ideally one who is well-versed in Asterisk. The main features we're after, though not limited to, are call logging and reporting, an IVR system, and call routing and forwarding capabilities. Key Skills and Experience: - Extensive experience in VoIP development. - Proficiency in Asterisk. - Ability to develop an IVR system. - Experience in call routing, call forwarding, and call logging mechanisms. Given the nature of our project, it’s crucial to have a level of experience in these areas. Understanding the intricacies of these features is what will drive the success of our project. Though we have not specified it in the initial questions, potential co...
I have install Asterisk (with freepbx), for gui to a Raspberry pi 4. I need a DEVELOPER TO WRITE a succesfull CODE and configurate the system to: Make a AUTO-VIDEOCALL from extention ''0'' to extention ''1'' or ''2'' or BOTH at the same time. The exctention ''1'' or ''2'' are log in to androip application, called ''PortSin'''or another app called ''Linphone''. Extention ''0'' is already loged into Raspberry with the programm ''Linphone''. When i press a button connected to GPIO 21 i want the audiocall to be started for 9 seconds. If the call is anwsered I also need to OPEN the DOOR by pushing ...
I'm seeking a skilled professional to set up a new VoIP asterisk system with primary features of call routing and call recording. Must have the ability to have custom caller ID for outgoing calls. This project entails configuration for a small scale operation with less than 10 users/phone lines. Key Job Requirements: - Proficiency in setting up and configuring VoIP systems - Outstanding knowledge of asterisk - Experience with call routing and call recording This task requires an efficient and effective approach, understanding the needs of a smaller user-base. Someone with a strong track record of setting up VoIP systems and the knowledge to troubleshoot potential issues would be perfect.
We used Asterisk as our phone system And we used Flutter's Sip-UA as a client The communication platform is WebRTC The problem we have is that when the internet suffers a few packet losses during a call, the client leaves the channel and then it is completely silent until the call is disconnected. We simulate the same scenario with a softphone, after the internet is disconnected and reconnected, the call continues and there is no problem. Are there any friends who can guide me?
Seeking a professional to integrate Asterisk VoIP into my existing Odoo system for my call center operations. Key tasks include: - Installing Asterisk VoIP - Setting up integration with Odoo - Implementing Call Recording functionality - Configuring Interactive Voice Response (IVR) - Implementing Automatic Call Distributor (ACD) The system should be able to handle 21-30 concurrent calls at peak times without any lag or quality degradation. Skills & Experience Required: - Proficiency in Odoo and Asterisk VoIP setup and integration - Proven track record in call center technology setting-up - Ability to work under strict deadlines and handle pressure - Understanding of call center operations and key processes - Excellent problem-solving abilities - Attention to d...
