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    2,000 asterisk ivr agi perl 건의 일자리 확인, 급여 기준: USD

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

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    ...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...

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    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.

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    ...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...

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    Python Telegram Bot w/ VoIP 19 시간 left
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    I'm looking for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom VoIP libraries to make IVR calls. Here's a brief on what I need: - IVR Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integration: You'll need to integrate the bot with an existing SIP server. Ideal Freelancer: - Proficient in Python, especially in developing Telegram bots. - Experience with VoIP libraries, particularly in the context of IVR calls. - Familiar with SIP server integration. - Strong understanding of DTMF technology. If you're confident in your ability to bring this project to life, let's connect.

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

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    II. the task #2 (less important but fast to fix) email notifications do not work for some reason (on server with webUI @159.65...) The email sending works with cron there contains 1 task here /var/spool/cron/tabs/root and 4 here as i understand /var/spool/cron/tabs/serg guess user serg is not configured properly(but it is a full disk copy, so not sure) or cron tasks are not active or lunching.. also i see 3 cron procs on old sever (ps-A|grep cron) and only 1 on new server... you can change email in scripts and test (the email sending is working ok tested it with small /root/) p.s. you can check working details on old server @213.168.... I. the most important but could take more time The statistics section do not work at all on the New server here is active clients on OLD server(s)...

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    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    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...

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    Whats-2-Pbx 종료 left

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

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    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 descri...

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    AI Chip Engineer 종료 left

    I'm looking for a highly skilled and experienced chip designer to build a photonic chip tailored for AI applications. The chip should be capable of AGI, Gen AI, incorporating computer vision, natural language processing, and robotics. Key Responsibilities: - Creating a comprehensive design - Ensuring the chip supports AI functionalities including, but not limited to, computer vision, natural language processing, and robotics Required Skills and Experience: - Extensive knowledge in chip design, particularly mixed-signal chips - A strong understanding of AI and its applications such as AGI, Gen AI, computer vision, natural language processing, and robotics - Proven track record of innovative and successful chip designs This is a challenging project, requiring someon...

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    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    ...with Nice Incontact IVR system to improve my current setup. - Simplify Call Flows: We currently have 10-20 call flows which need to be simplified to enhance customer experience and boost efficiency. - Voice Recognition Improvement: The system's voice recognition capabilities need upgrading to ensure a seamless communication process for customers calling in. - Application Integration: The final task would involve the integration of the IVR system with other applications to optimize functionality. The ideal candidate should have proven experience with Nice Incontact, call flow design, voice recognition technology as well as system integration. Understanding of customer service operations will also be an added advantage. Your job will be to streamline and optimize ...

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    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.

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    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    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

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    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...

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    I'm in need of a skilled freelancer who has expertise in Perl scripting, primarily for file manipulation tasks. Key Requirements: --Gather date from a table in oracle. Based on that date find the records available in a home directory and store it in a .txt file. Skills and Experience: - Proficiency in Perl scripting is a must. - Experience in handling various file formats, specifically .txt and .csv. - Strong understanding of file operations including renaming, formatting, and sorting (although the specific task was skipped). I need the code done within 2 days and willing to pay 3000 which would be double the cost of what is given .

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

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    I'm urgently looking for a skilled professional to quickly handle an IVR-related task for me: - Make IVR calls to 120 different cell phone numbers delivering specific information. Experience in both script writing and call routing configuration is appreciated, although not mandatory. The swift initiation and completion of calls is paramount to this project. Therefore, the ideal freelancer will demonstrate adeptness in utilizing any form of IVR system, be it traditional PBX, cloud-based, or hybrid system. Please note that this project is time-sensitive and needs to be started and finished ASAP. Your adaptability and readiness to start immediately will be highly regarded.

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    looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications

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    ***** NO AI DESIGNS - I WILL REJECT ALL AI DESIGNS ******* I need a label for my coffee brand called tithe coffee. My aim is to give back 10% to the community which is what tithe means. My coffee is a superfood coffee with 14 natural ingredients. In regards to colors/design I'm not sure but I want something that stands out and has a minimalist style kind of like agi/athletic greens, ryze coffee or life boost coffee but this is only for the label not the whole bag. I have created a simple logo that I have attached that shows the english spelling as well as the hebrew spelling of tithe underneath it but the label should only have english words on it. This logo will likely be redone over but if you include a new logo in your design that I like I will pay extra for it but t...

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    342 건의 응모작

    I'm currently using PERL 5.10 system and requires a proficient expert to aid in modifying how passwords are hashed in my system. Initially, we employed SHA-256 but we're looking to switch to PBKDF2. Ideal Skills and Experience: - Proficient in PERL 5.10 - Familiarity with SHA-256 and PBKDF2 hashing methods - Prior experience in password re-hashing Task Objectives: - Identify current SHA-256 hashing method in the system - Modify hashing method to use PBKDF2 instead - Validate successful implementation of the PBKDF2 hashing method The project necessitates a strict focus on security and confidentiality given its sensitive nature.

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    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    My objective is to significantly enhance customer service efficiency and personalize customer interactions through an AI-based IVR system. Key Tasks: - The IVR system should answer frequently asked questions autonomously. - It should intelligently route calls to the corresponding department based on customer input. - Gathering customer data for more efficient call routing and enhanced personalization should be a key capability. Integration requirements: - The IVR system must integrate effortlessly with live chat to ensure a cohesive customer service offering. Ideal Freelancer: - Proficient in AI and IVR systems - Experience in implementing live chat integration - Understanding of efficient call routing mechanisms - Experience in developing personalized cust...

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    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

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    ...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: ...

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    Script Feature: - Login to a list of SSH servers by domain or IP (from text file) - Use a set of usernames + password (can be multiple passwords) to login to the servers. - Run Bash Script on the connected server (which comes from a txt file on the server running the script, not on the connected to server) - Timeout if SSH server did not answer. - Display results at the end of a run (show which servers the script did not run because of errors). Login simultaneously to all available ssh servers. Not server by server.

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    I need an expert in 3CX Pro systems to fully optimize my call flow setup. Key tasks will include configuration of: - IVR - QUEE - QUEE WAIT 2 Minits after - If Closed Play Message and redirect to Voicemail - Or All Users Bussy - Play Message and redirect Voicemail I have atached one Pic.

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    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

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    I'm in need of a talented freelancer for a voice recording project: - Unfortunately, I skipped the questions regarding the intended use for the voice recordings, what information successful freelancers should include in their application, and the preferred language for the voic...voice recordings, what information successful freelancers should include in their application, and the preferred language for the voice recordings. - Even without this information, I expect potential candidates to be adaptable and versatile with their voice talent abilities. - Regardless of the language and the purpose of the voice-over, having previous experience in voice-over, podcasting, or IVR will be advantageous. I look forward to hearing from diverse talents who can cater to multiple voice r...

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    I need to convert the following Perl code to PHP: The expected output is a PHP class that can convert a docx file into text. The attached zip contains a mockup of such class (with demo code) and a test docx file. The task is straight forward. If you place a bid, expect it to be accepted. If you fail to deliver, you'll be subjected to a bad review. This is to avoid wasting my time with autoposters and people who lowball the bid and then try to ask for more money. You have been warned. Read the brief, check the code and place a bid accordingly.

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    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.

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    ...text base messages. Can you help me build this system and how much is going to be the total cost I donot have any number to use at the moment? Autodailer: I need to do unlimited calling to usa, Canada and UAE, on raw but verified phone numbers. (sort of cold but verified data of 10million contact numbers) Secondly I need an auto dialer which is going to be linked to a dashboard along with an IVR played in three to four accents which doesn't sound robotic. I need numbers atleast 10 for dialing and routes most importantly sometime the numbers are pin pointed as spam so for after every 10k calls the numbers needs to be changed or delisted. Now the challenge: How many numbers can you provide? How many routes do you have? How many ips we need do we need for mass quantit...

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    I'm urgently in search of a well-rounded and outstanding programmer fluent in numerous programming languages. It's essential for you to excel in Python, Java, C++, C#, Scala, Ruby, Python, Perl, Javascript, Go, SQL, Rust, Swift, Kotlin, HTML/CSS, and be experienced with NoSQL and LaTeX. Ideal technical expertise includes: - Data Visualization: Proficiency in matplotlib, D3.js, bokeh, or comparable tools - Machine Learning: Familiarity with TensorFlow, scikit-learn, PyTorch, or equivalent - Unit & Functional Tests: Experience using Selenium, Cypress, Mocha, or similar - Cloud Computing: AWS, Azure, Google Cloud Platform, or equivalent knowledge - Containerization: Docker The central task is to perform insightful programming assessments and coding within my enterpris...

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    ...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...

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    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...

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    ...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...

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    inv 1 종료 left

    Completed milestone 1 for IVR testing

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    I'm seeking an expert to help cr...an expert to help create a Dockerfile that runs Perl and DBD::mysql so that I can run a legacy website. This should be a simple task if you have the knowledge. Specific requirements include: - A Dockerfile is the only output required, roughly corresponding to mine - It should build and run in wsl, i.e. docker run -p 8080:80 image_name - It must use Apache, Perl and the CPAN module DBD::MySQL - It should have been tested on a MySql database, the file is provided to run on localhost:8080/ - A screenshot of the above is required to prove that it is working (you just need a MySql Table with fields Id and Name) Candidates should ideally have: - Extensive experience in Docker and a basic knowledge of Perl - Understanding of Per...

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    I'm looking for a skilled developer to create an AI Intelligent Virtual Assistant (IVA). It will need to integrate Natural Language Processing and Machine Learning Algorithms, along with Digital Human capabilities. The main functions should include: • Understanding and processing user inputs • Developing responses using machine learning • Implementing computer vision for a digital human interface The AI has to be compatible with Windows and Mac desktop platforms, Mobile platforms like Android and iOS, and also web-based applications. Ideal freelancers should have significant experience in AI, machine learning, natural language processing, and virtual assistant development across different platforms.

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    ...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: [audiosocket] exten = s,1,A...

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    Task details - For ISUP/SS7 (E1 card implementation) along with IVR knowledge in any open source SIP servers like Freeswitch/Yate/Mobicents etc. or working experience in CRBT server. Interested candidates please apply

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    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 projec...

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    SRP Consulting 종료 left

    We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...

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    fix forms issue 종료 left

    ...working. MochaHost tech support got the Import buttons on the forms fixed but suggests they may not be able to fix the XML & XML+XSL submission buttons at the bottom of the forms. Here's what they say about that: When the server is upgraded, the Perl modules in the new server might not be fully imported. That's why your website doesn't work because some Perl module dependencies might be deprecated. MochaHost support says: We highly suggest you talk about the error with your Web Developer to ensure they can help you add the missing path on the Perl module. Based on our check, your website form processing is missing pointing to the `File::Spec` module which has been installed on the server. It seems like that might be a fairly easy fix and I have...

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    ...performance tuning, sizing and capacity planning, database backup and recovery in a Unix O/S using RMAN and Oracle export/import utilities, administration, and maintenance to provide backup and disaster recovery. • Knowledge of installation, configuration, and maintenance of multi-node Oracle RAC enabled databases, including Oracle ASM and Oracle CRS management. Knowledge of PL/SQL packages, shell, and Perl scripts for task automation, and proactive database monitoring and notification using Oracle OEM Grid Control. • Demonstrated written and oral communication skills reviewing and editing established database policies and procedures and developing new SOPs for database software installation and configuration. • Experience with Cloud(AWS/Azure/OCI) migrations (DB a...

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    autonomously AI 종료 left

    Create AGI that autonomously crawls the web, learns from its interactions, and makes decisions is an ambitious endeavor. Let's break down the key components and challenges involved in creating such a system: 1. **Web Crawling and Data Collection**: - The AGI needs to crawl websites to collect data. - Techniques like **web scraping** can extract information from web pages. - Consider using AI-powered web scraping tools that adapt to changing website designs and dynamic content³. 2. **Adaptive Learning and Decision-Making**: - The AGI should learn from the data it collects. - **Adaptive scraping** techniques allow the bot to adjust to website redesigns and structural changes. - Use machine learning (e.g., convolutional neural networks) to...

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