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    2,000 agi perl asterisk công việc được tìm thấy, giá theo USD

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

    $2250 (Avg Bid)
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    Tôi có một file viết bằng ngôn ngữ perl muốn chuyển về ngôn ngữ php. File perl mẫu được đính kèm theo. i have a perl script. i want convert to php.

    $30 - $250
    $30 - $250
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    Customize Asterisk Đã kết thúc left

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

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    NDA
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    Lập trình một file *.pl để đọc và xuất ra cột theo từng yêu cầu cụ thể. Vui lòng xem file đính kèm để biết thêm thông tin chi tiết.

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    EducaCallcenter Đã kết thúc left

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

    $500 (Avg Bid)
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    Tôi đã có sẵn tổng đài Asterisk, đã có sẵn SuiteCRM. Cần lấy API từ tổng đài Asterisk để kết nối với Suitecrm.

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    $30 - $250
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    Project for Vu L. Đã kết thúc left

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

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

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    Project for nttranbao Đã kết thúc left

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

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    Private Project 1 Đã kết thúc left

    Chào Tuấn, Mình có thể nói chuyện trên skype về Perl skill của bạn k ??? Add mình nhé :

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    Mình muốn tìm thuê một người có khả năng thiết kế và sửa lỗi một chương trình được viết bằng ngôn ngữ Perl. Yêu dự án : Tìm câu đối xứng giữa tiếng việt và tiếng trung. Hiện tại bên mình đã có code tìm câu đối xứng giữa tiếng trung và tiếng anh, nên bây giờ theo yêu cầu của dự án, các bạn sửa đổi giúp mình. Các bạn ai có khả năng thực hiên thì liên lạc theo hòm thư sau : ,yahoo: nguyenhanhtn85 (liên lạc trực tiếp), hoặc thông qua sđt: 01693009137. để trao đổi thêm các thông tin, cũng như thương lượng giá cả .

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    Asterisk PBX Installation for Call Routing 6 ngày left
    ĐÃ XÁC THỰC

    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...

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    Asterisk PBX 8 ngày left

    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

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    Asterisk SIP Server Configuration and Demo 5 ngày left
    ĐÃ XÁC THỰC

    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 ..

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    ...am looking for a PERL, HTMX, MYSQL programmer who can help me with an intermediate level project. The ideal candidate should have experience in these areas and be comfortable starting a project from scratch. I will give you a dump from my MySQL table. From which I need an interactive HTML interface, based on Perl, HTML, HTMX to this table. I need to be able to search, add, edit and delete entries as well as combine two entries into one, all via one HTML::Template file with a single perl script as interface to the MySQL table. I do want an active submit button for all changes, as well as a single interactive upload button for each field of the MySQL table. also each field needs to be validated via a given specification. All code needs to be well documented and commen...

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    ...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...

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    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.

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    I have a Perl script that searches a CVS file and generates HTML. When this script has zero results, it currently displays a page stating "SORRY there are no RESULTS". I need this behaviour to change: instead of generating a custom error message, I want the script to send a 404 call to the Apache server. Key Requirements: - Modify the Perl script to send a 404 call to the server when no results are found. - Implement a custom error message to be displayed on the 404 error page. Additional Details: - No logging of details of 404 errors is required for monitoring or debugging purposes. - The script should still function as normal when results are found, only altering its behaviour when no matches are found. - The ideal candidate for this project should have experien...

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    I'm looking for a skilled Perl expert to help me remove duplicates in an Excel dataset. The removal process should be based on multiple columns, ensuring a thorough cleanup. Ideal Skills & Experience: - Strong proficiency in Perl programming - Proficiency in data analysis - Expertise in Excel and handling large datasets - Prior experience in removing duplicates from Excel based on multiple columns Communication: I prefer to have project updates through video calls, so please be comfortable with this form of communication. Your ability to walk me through the process will be highly appreciated.

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    webrtc proxy 1 ngày left

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

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    PJSIP SIP Trunking with Python Bot 16 giờ left
    ĐÃ XÁC THỰC

    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...

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    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...

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    PJSIP SIP Trunking with Python Bot -- 2 19 giờ left
    ĐÃ XÁC THỰC

    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...

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    I'm having issues with a specific user function in my Pgsql ( postgresql 8.4) database that needs fixing. After transferring server and DB from real PC to the VM, ... 1365 | 1365 32645239 | ADMIN | _util_update_stat | 27 | 37511 | 37511 calls And new doesn't have any calls of _util_update_stat at all which makes Stats table empty You need to track why is this happening & fix it. Server is a bunch of scripts, but seems it doesn't matter cause all scripts just READ this stats table (i guess) Pgsql expirience is a must Perl expirience is a plus. Plz no "manager" bids, don't waste your time, will talk with developers only. (if you are a "manager" it will be seen after the 2nd tech question Plz don't wa...

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    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.

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

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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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    Consultor Asterisk 18 - PJSIP Đã kết thúc left

    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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    Server Perl Scripts Debugging Đã kết thúc left

    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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    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 Đã kết thúc 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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    AI Chip Engineer Đã kết thúc 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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    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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    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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    Configurar Asterisk DTMF Đã kết thúc left

    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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    Trophy icon Logo for a coffee brand - 01/04/2024 04:29 EDT Đã kết thúc left

    ***** 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 bài tham dự

    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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    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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    Simple Script (Python or Perl) Đã kết thúc left

    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 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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    Perl to PHP Code Conversion Đã kết thúc left

    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.

    $60 (Avg Bid)
    $60 Giá đặt trung bình
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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...

    $28 / hr (Avg Bid)
    $28 / hr Giá đặt trung bình
    13 lượt đặt giá