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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.

Hello, I am looking for someone with some knowledge about LeiaPix. The idea is to convert 2D image in video. I run a technology company for the real estate sector, creating and selling digital solutions for real estate professionals. I'm currently working on a project to create a platform that would enable estate agents to transform their photos into video using AI. Have a look on attach file. So I'm looking for people who know how to use LeiaPix and who can create quality effects to transform photos into video. My clients generally have between 8 and 15 photos per property.

O Projeto visa construir uma parte de um App em .NET MAUI que (uma pagina) para ser instalado em um POS Android de vendas (exemplo: ). Na pagina XAML do App o usuario digita um valor, escolhe Credito/Debito/PIX e ao processar as informacoes sao enviadas ao POS que retorna o status conclusivo da operação. O código fonte deve ser entregue.

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.

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 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...

...combobox for select a single product and also a datepicker for select a single day or range. This information are used to populate cards and chart 3- Balance In balance it will show the bank balance, pending balance and also a card for ask withdraw. 4- Configurations This will have a button to logout and 2 sliders for notification settings (when sale paid and when generated a new bankslip/invoice or Pix(this is a brazilian qr code for payment) 5- My Account Here will have inputs for Name, Phone, Email, Current password and newpassword (so, it will allow to change password) Notifications this is the priority need. I need to be able to send notifications in real time. For notifications, if it is necessary, we could use some library or service. We could talk for choose the bette...

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 seeking a seasoned professional to provide ongoing support for my existing Fortinet Firewall. The ideal candidate will have a strong grasp on the following areas: - **Configuration Adjustments**: I require assistance in fine-tuning the firewall rules, NAT settings, and VPN configurations. It's crucial that the candidate is well-versed in these areas, as even minor misconfigurations can have severe security implications. - **Troubleshooting Connectivity Issues**: The freelancer should be proficient in identifying and resolving connectivity issues. Quick and efficient troubleshooting skills are paramount to keep my network running smoothly. - **Implementing New Security Policies**: I need to ensure that my security policies are always up-to-date. The candidate should ...

Olá, precisamos fazer a integração com a Zoop Pagamentos, integração com o Cartão de Crédito, Boleto Bancário e Pix, a parte de cartão de crédito vamos usar a clearsale pra fazer o antifraude. documentação da zoop:

Precisamos iniciar um piloto de um projeto para efetuar pagamentos em POS Android. Esse app, precisa efetuar uma autenticação básica (usuário e senha). Com essa autenticação, o usuário será direcionado para uma tela, onde a pessoa informará um valor e, com esse valor, disponibilizará em tela as opções de parcelamento (PIX, Débito e Crédito até 18x). Para esse parcelamento, existe uma API que traz os coeficientes para esse calculo de parcelamento. Após o usuário selecionar a opção de pagamentos desejada, o app deverá efetuar os pagamentos utilizando os SDKs da Getnet ou Software Express.

Trabalho com agencia de viagem e turismo e preciso de uma site profissional com o mesmo sistema da 123 milhas, decolar, tudo completo. Por enquanto eu trabalho tudo de forma manual. Mas agora quero um site onde o cliente possa fazer a sua própria pesquisa e faça seu próprio pagamento, seja no cartão de crédito, débito ou Pix e futuramente quero colocar no boleto parcelado. Vendo passagens com milhas(pontos) aéreas e eu quero ter o controle pra mudar o preço das milhas pra que o preço saia no site em real para o meu cliente. E que os meus clientes possam receber notificações via e-mail, whatsAPP com informações do voo, promoções, a emissão com o código localizador,...

I would like a skilled Laravel developer to integrate PagSeguro's payment gateway into my website, including Boleto bancário and Pix. Key Project Requirements: - Integrate Boleto Bancário: The system should generate boleto slips, track boleto payments, and handle the necessary validations. - Include Pix: The system should support payments via Pix, a new and increasingly popular payment method in Brazil. Ideal Skills and Experience: - Proven experience with integrating payment gateways into Laravel applications. - Familiarity with PagSeguro's API and requirements. - Knowledge of Boleto Bancário and Pix payment systems. - Strong understanding of Laravel best practices. The project needs to be completed within a month. Please let m...

... This configuration should allow me to deploy the EC2 instances and also bring-down when needed. Key Requirements: An EC2 instance with Security group, Keypair, VPC components, CloudWatch logs and WAF integration. The EC2 will be launched in Private Subnet, with ALB and no direct traffic allowed to the instance. A Bastion/jump server for SSH to EC2. EC2 connects to the internet using NAT-gateway. WAF to be integrated with ALB, and ALB only listens on HTTPS. Security groups are restricted with limited access. Allocate and associate an elastic IP to the instance. The EC2 instance will run Ubuntu and serve as application server. I can provide AMI image of the web-app. It is a Python FastAPI/Nginx app. No other service is needed like RDS or S3 etc. Docker or Kubernetes are not need...

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...

...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...

Buscamos representante de vendas, com carteira de clientes, para oferecer os produtos da empresa: pix para apostas esportivas (bet), pix para varejo e máquina de cartão.

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.

Aspectos chaves: - Idioma: O site deve estar em Português. - Páginas: Estão previstas entre 01 páginas para o site. - Pagamento: O site deverá incluir Pix como forma de pagamento. Habilidades ideais: - Proficiência na língua português - Experiência no desenvolvimento de um site de e-commerce ou venda de livros - Proficiência na integração do pix pagamento. Amostras de trabalhos anteriores ou portfólio demonstrando o acima exposto serão muito apreciados.

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...

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.

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.

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

I need an expert who can assist in configuring network address translation (NAT) on iptables, particularly focusing on Static NAT. Key Project Requirements: - Deep understanding of iptables operations - Proficiency in dealing with Static NAT configuration - Ability to handle specific IP ranges during the configuration Your work will revolve around ensuring that these specific IP ranges are correctly configured in the Static NAT setup. Familiarity with various internet protocols will be advantageous. This project requires a careful and meticulous individual with a knack for detail, as absolute precision is crucial in executing this task. Please, only apply if you have suitable skills and experience. check the attached picture. the 81 port does already wor...

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...

As an enthusiast in scheme/racket programming, I'm currently building a Church-compiler. The project is majorly complete, however, there are some aspects I need professional assistance with. The key tasks inclu...I need professional assistance with. The key tasks include: - Writing and completing certain functions in the program to pass pre-existing test cases. - Enhancing syntax parsing to better compile to Lambda-calculus In order to achieve this, I'm in need of a programmer with substantial experience in Scheme/Racket, particularly in implementing algorithmic logic, constructing compilers(church->nat, church->bool,church->listof and more) and syntax parsing techniques. If you are proficient in these skills, I would love to connect with you. More informati...

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.

I want to install a nextcloud-aio instance on rocky linux behind a haproxy the nextcloud is supposed to be on a seperate vm all VMs are behind NAT

/ip route add dst-address=0.0.0.0/0 gateway= distance=1 add dst-address=0.0.0.0/0 gateway= distance=2 /ip address add address= interface=ether2-WAN1 add address= interface=ether3-WAN2 add address= interface=ether1-LAN1 /ip firewall nat add chain=srcnat action=masquerade out-interface=sstp-adsl add chain=srcnat action=masquerade out-interface=sstp-starlink add chain=srcnat action=masquerade /interface sstp-client add name=sstp-starlink connect-to=<server_address> user=vpnuser password=vpnpassword certificate=none profile=default-encryption disabled=no /interface sstp-client add name=sstp-adsl connect-to=<server_address> user=vpnuser password=vpnpassword certificate=none profile=default-encryption disabled=no I want Force Interface

Srs. Preciso fazer a comunicação com o banco de pagamentos Efí, cartão, pix, boleto via API. No site do banco tem os esquemas para conexão com diversas linguagens, mas preciso passar para ASP CLASSIC ou fazer com que meu site que está em ASP trabalhe com as informações vindas. - Tentei com oXMLHTTP = mas deu alguns erros. - Acho que estou errando em alguma linha de código. - Se for possível fazer a comunicação com Jquery também seria uma boa.

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.

...the user through that dashboard & push/deploy that configuration to the particular or multiple routers & switches automatically. The goal is to remove the use of Command Line Interface (CLI) to manually write configuration commands to each of the routers & switches. For this academic project, the configuration category is limited to Hostname, IP addresses of the interface, OSPF, VLAN, HSRP, ACL, NAT. NOTE: Similar to the below picture, you are required to use GNS3 to build a tier-3 network topology with c2600 routers & c3725 switches to demonstrate this project in a simulated environment. You are requird to use HTML CSS JavaScript for building very simple but attractive & cool dashboard, Python for backend functional/logical script, Ansible (python framewo...

...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...

...in PDF. - 2 accounts from Santander - 2 accounts form Caixa Economica Please find attached 2 PDF files as an example. For Caixa Economica statements, there are only 2 PDFs (one for each account). For Santander, the PDFs are divided by month. Details: - Critical data fields to be extracted include: transaction date, transaction description, transaction amount, number of destination accounts for PIX, TEDs and DOCs (bank transfers), and account balance. Ensure accuracy in these fields during conversion. - The Excel spreadsheet should have 4 tabs, one for each bank account. - The converter should be designed to handle non-protected PDF files. No provision for password-protected files is required. The ideal freelancer for this job would have proven experience in data extraction a...

...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...