使用Freeswitch 作为服务器搭建一个呼叫中心，并编写iPhone 和Android 移动客户端App 供客服人员使用 - 呼入 + 呼出 - 语音信箱功能 - 看到其他客服人员的状态和姿态 （在线，离线；在线接听中，在线空闲中，在线转接中... 等等） - 看到队列中的等待客户，并接听来电 - 队列中有等待客户时，自动检测在线且空闲的客服并分配呼叫 - 在App 中从Freeswitch服务器端下载预配置的分机号码，每个分机号码对应注册用户名 - 接听电话过程中进行呼叫转移 - 接听电话过程中拉入第三方客服加入通话 - 文字消息功能 - Email 通知客服有语音或者文字留言
主要指导elastix asterisk 的安装配置工作，熟悉Asterisk的模块（包括体系结构，配置文件，Log日志,History）,能重新打包ISO文件
I'm looking for a skilled Android app developer to create an application for my company. Key Features: - The app will display the current time and company name at the top center of the screen. - Users will log in using a personal number and password, both entered into text boxes. The password will be concealed with an asterisk. - The app will include a button to submit the login details, labelled "Anmelden". - There will be a copyright notice at the bottom of the screen, aligned to the right and in a font size smaller than 16. CSS: Background: #000 Font Color #fff Button Color: orange Authentication: - The app will utilize an NFC system for authentication, along with a clear username and password scheme. - Users will authenticate using a text-based password or NF...
...already partially developed: The system is built using Issabel and Asterisk. The call center module in Issabel is set up and functional. OpenAI APIs have been integrated for dynamic conversational responses. Current Issue: The project needs to be finalized. Specifically, we need the call center module to handle the following: Import large .csv files containing 1,000-5,000 phone numbers. Automatically call customers listed in the .csv file. Offer services and respond to their questions dynamically using OpenAI API. We are seeking a freelancer who can complete this project efficiently, ensuring the system is capable of: Handling high-volume call operations without errors. Maintaining seamless integration between Issabel, Asterisk, and OpenAI APIs. Providing clear docum...
...to help me recreate a UI for both Android and iOS. This app will connect to an existing FreePBX/Asterisk server for international calling and integrating with APIs for other functionalities like billing. All necessary APIs and a Figma design for the UX/UI are available. Current Status: - An Android version of this product exists, but we are redesigning and recreating it with Flutter for cross-platform compatibility. Required Features: - User registration/login - Call history tracking - Push notifications -app to phone calls (international) Visual Style: - The app should have a modern and sleek aesthetic. Ideal Skills and Experience: - Proficiency in Flutter - Experience with FreePBX/Asterisk - Familiarity with API integration - Ability to work from Figma designs - Prev...
I'm looking to integrate JSSIP with Asterisk to provide a web-based client interface. The client interface should include call history, specifically focusing on missed calls. Key Requirements: - JSSIP and Asterisk integration: Proficiency in these technologies is crucial for this project. - Development of a Web-Based Client Interface: The primary goal of this integration is to create a user-friendly, web-based client interface. - Call History Feature: The interface should have a 'Call History' section that records missed calls. Ideal Skills: - Strong knowledge in VoIP technologies, particularly JSSIP and Asterisk. - Web development skills with a focus on creating interactive and intuitive user interfaces. - Experience with developing telecommunication...
... Implement a PHP-based web form that triggers outbound calls to users, utilizing CallFluent’s AI via Twilio. Enable bulk call initiation for sales campaigns from a list of phone numbers (CSV or DB). Ensure smooth SIP trunking between Dinstar and Twilio/CallFluent. Optimize call routing, logging, and reporting. Technical Requirements: Experience with VoIP and SIP (Asterisk, FreePBX, or similar platforms). Hands-on experience with Twilio API (voice calls, webhooks, and outbound call triggers). Understanding of Dinstar GSM Gateways and SIP trunking configurations. CallFluent AI integration expertise (or similar AI-based telephony platforms). PHP Development skills to implement web forms and initiate calls. Familiarity with JSON...
I'm facing issues with my Asterisk 22 dialplan. The problems are primarily with extension dialing. Specifically, call forwarding for the extensions is not functioning as it should. Key Requirements: - Diagnose and correct the issues within the dialplan - Ensure all extensions are correctly set up for call forwarding Ideal Skills and Experience: - Extensive experience with Asterisk, particularly version 22 - Strong understanding of dialplans and call routing - Excellent troubleshooting skills
...concurrent calls and scale with increasing user base. 6. Security: Ensure secure SIP signaling and web access. --- Key Deliverables 1. Fully configured and scalable VoIP system (Kamailio + FreeSWITCH). 2. User portal for trunk management and billing. 3. Real-time monitoring system with dashboards. 4. Billing integration for user payments and provider reconciliation. 5. Comprehensive documentation for deployment and maintenance. --- Development Steps Phase 1: Infrastructure Setup 1. Provision Servers: 4 servers (Linux OS) for: Kamailio (SIP signaling and routing). FreeSWITCH (media handling). Billing System (AstPP). Monitoring and Analytics (Prometheus + Grafana, HOMER). Hardware Specifications: CPU: 16 cores. RAM: 128 GB. Storage: SSD/NVMe ...
...connected to an Asterisk instance on my Linux home server. This project involves integrating the intercom to call Linphone accounts and executing specific actions based on inputs from the software's numeric keypad during a call. Key Tasks: - Configure the intercom to call Linphone accounts. - Implement a system where pressing digits on the software numeric keypad triggers pre-defined bash or Python scripts. (The primary action is to open the gate when a certain digit is pressed by my own API) I have accounts on Linphone and my Asterisk instance is set up. Importantly, I want to keep my Asterisk ports secure and not exposed to the internet. Therefore, the Asterisk should function as a client relative to Linphone. Ideal candidates should have: - Extens...
...Modification 4: Add the following data fields for note uploads (seller instructions): University Name* Lecture Notes for* (Subject Name) Course Name* Course Code Semester Teaching Professor* Year of Study* How to Prepare for This Subject* Price Type (Fixed/Negotiable) Price Upload Images of Notes Your Name Mobile Number Agree to Terms and Conditions Fields marked with an asterisk () are mandatory. 4. Modification 5: Add a WhatsApp button for direct communication between buyers and sellers of notes, similar to the textbook feature. 5. Modification 6: Add a login/registration requirement for students to upload or access notes. ________________________________________ Feature 3: University Events Management • Current Functionality: Students can uploa...
Actualmente en mi empresa utilizamos una terminal de conmutación telefónica las líneas telefónicas externas son provistas por un IPS. Debido a que la terminal de conmutación PBX ya no puede soportar mayor cantidad de extensiones me veré forzado a modernizar a tecnología VOIP quizá utilizando Asterisk en un servidor rocky linux 9 que se encuentra de forma física. Además necesitare un archivo de los comandos utilizados a manera de instructivo ya que se necesita replicar lo mismo en otro edificio.
...professional who can set up an Asterisk server for me. This server is intended to manage both alarm-with-voice and alarm-without-voice functionalities. It will facilitate communication between clients through custom XML message exchanges via Asterisk server. Key Requirements: 1. Asterisk Server Configuration: The server should be configured to handle both types of alarms. 2. Real-Time Client Registration: Implement a system for instantaneous registration of clients from a MySQL database. This system should include an authentication mechanism and client status monitoring. 3. Testing Guidance: Provide instructions on how to assess the server's performance using SipSoftphone. Ideal Candidate: The perfect fit for this project is someone who possesses extensive kn...
...configure an Asterisk server for me. This server will handle both alarm-with-voice and alarm-without-voice, where clients send and receive XML-type messages with each other. The server needs to support real-time registration of clients from a local MySQL database, as I expect to serve a large number of clients. Key Requirements: 1. Configure the Asterisk server for both types of alarms. 2. Implement a feature for real-time registration of clients from a local MySQL database. 3. Provide guidance on how to test the server using SipSoftphone. The ideal candidate should have extensive experience with Asterisk server configuration and a strong understanding of MySQL. Skills in network configuration and VoIP technologies will be highly appreciated. Please note, I'...
I need a professional to set up my Asterisk server. The primary purpose is to connect it with a Postgres database and an Android Linphone On-Board Unit (OBU). Key Tasks: - Setting up an Asterisk server - Connecting Postgres with the Asterisk Server and Android Linphone OBU - Configuring Linphone to work seamlessly with Asterisk and Postgres The ideal freelancer should have: - Extensive experience in Asterisk server setup - Proficient in integrating Asterisk with Postgres - Familiar with Linphone - Knowledge of Android platform integration - Skills in VoIP technology
I'm in search of a seasoned Asterisk & FreePBX developer to create a comprehensive call center CRM. This web-based interface system should prioritize Direct Inward Dialing and incorporate essential call handling features such as Call Barge, Call Whisper, and Call Mute. Additional functionalities should encompass Call Queues, Call Transfer, Call Pause, Lead Upload, Recording, and Call Scheduling. Key Features: - Direct Inward Dialing as the top priority - Web-based interface for user accessibility - Critical call handling features: Call Barge, Call Whisper, Call Mute - Other functionalities: Call Queues, Call Transfer, Call Pause, Lead Upload, Recording, Call Scheduling Ideal candidates should have: - Extensive experience with Asterisk & FreePBX - Strong web dev...
I'm facing an issue with my HT814 that's causing incorrect call rates for international calls. They are being mis billing when landing on the Grandstream HT814 via Asterisk. I need this problem diagnosed and resolved. Ideal Skills: - Extensive experience with Asterisk - Proficient in troubleshooting Grandstream HT814 - Knowledgeable in VoIP billing systems - Familiar with international call rate regulations Please provide a comprehensive plan on how you would approach this issue.
...ou d’une API pour récupérer les leads générés selon les différents scénarios d’appels. Reporting clair et structuré pour faciliter le suivi des performances. Compétences recherchées : Expertise en speech-to-text (Vosk, Whisper ou autres). Compétences avancées en NLP open source, notamment pour le traitement en langue française. Connaissance des solutions de téléphonie VoIP, telles que Jambonz, Asterisk, ou équivalents. Maîtrise des architectures scalables et des environnements cloud. Connaissance des bases de données pour le traitement des leads (PostgreSQL, MongoDB, etc.). Capacité à fournir une solution clé en main avec do...
...already partially developed: The system is built using Issabel and Asterisk. The call center module in Issabel is set up and functional. OpenAI APIs have been integrated for dynamic conversational responses. Current Issue: The project needs to be finalized. Specifically, we need the call center module to handle the following: Import large .csv files containing 1,000-5,000 phone numbers. Automatically call customers listed in the .csv file. Offer services and respond to their questions dynamically using OpenAI API. We are seeking a freelancer who can complete this project efficiently, ensuring the system is capable of: Handling high-volume call operations without errors. Maintaining seamless integration between Issabel, Asterisk, and OpenAI APIs. Providing clear docum...
...usuários, domínios, rotas, configurações de NAT e regras de segurança. Requisitos Técnicos 1. Kamailio: • Configuração de proxy SIP para gerenciar múltiplos domínios. • Suporte para NAT traversal usando módulos do Kamailio (como rtpengine ou nathelper). • Implementação de controle de tráfego para evitar sobrecarga do servidor. 2. Asterisk: • Integração de tráfego entre Kamailio e Asterisk via SIP ou PJSIP. • Capacidade de configurar múltiplos servidores Asterisk para escalabilidade e redundância. 3. Segurança: • Configuração de firewall para proteger o servidor (iptables ou fa...
I need a professional translator to help me translate 51 to 100 audio files from English to Swedish. The original audio files are in WAV format. Details: - The audio files are VitalPBX sounds (Asterisk sounds). - The scripts or text for each audio file can be accessed via Google API. - You will need to download the audio files from a Google Drive link. - The final translated audio files should be in the same WAV format. Ideal Skills: - Proficient in Swedish and English. - Experience with audio file translation. - Able to use Google API for script access. - Knowledge of WAV format. The translation project should be completed within 1-2 weeks.
hello, I have a very old CentOS 5.5 installed only for a single application : Asterisk. I need to move the Asterisk to a new environement. I'm looking for someone who knows Asterisk and can help migrate datas & configuration.
I need a comprehensive step-by-step PDF guide that details the installation, configuration, and troubleshooting of Asterisk 20 with pjsip and Kamailio 5x on Ubuntu 22.04. The guide will be used to provision a lab with Vagrant for Virtual Box. Key Requirements: - Installation Steps: A detailed account of the installation process, ensuring even the most novice can follow. - Configuration Details: Clear and concise configuration guidance, covering all necessary aspects. - Troubleshooting Tips: Common issues and their solutions to help in smooth operation. Please ensure the document includes detailed explanations for each step.
I'm looking for a skilled developer to enhance my Asterisk system. The project has two main components: 1. **Caller ID Capture**: I need the system to capture the Caller ID as soon as the call starts. After the Caller ID is retrieved, it should be stored in a database for future reference. 2. **Survey Plugin**: I also need a survey plugin that records user feedback during the call. The plugin should be integrated seamlessly into the call flow, allowing it to capture feedback without interrupting the call. Ideal skills for this project include: - In-depth knowledge of Asterisk and its APIs - Experience in developing and integrating survey plugins into telephony systems - Proficiency in database management and storage solutions - Understanding of Caller ID systems and ...
I need a specialist to resolve serious call routing issues in my telephony system based on FusionPBX and FreeSWITCH. Issues: - Calls not connecting - Incorrect call destinations - Dropped calls These problems emerged after a recent system upgrade.
I'm seeking an IT expert with a stro...with a strong background in the telecommunication field, specifically with PBX systems, Magnus billing, or the VOOS3000 switch. Key Responsibilities: - Familiarity with Asterisk and Magnus billing or the VOOS3000 switch is crucial - Maintenance and troubleshooting of the PBX system - Integrating the PBX system with other platforms - Implementing a Round Robin configuration for Direct Inward Dialing (DID) and call conversion Ideal Skillset: - Profound knowledge and experience in telecommunication IT - Expertise in PBX systems - Proficiency in using and configuring Magnus billing and VOOS3000 switch - Familiarity with Asterisk - Strong troubleshooting skills - Experience with Round Robin configuration The expected timeline for the...
...in-depth knowledge of Vicidial/Asterisk and AGI script to help me troubleshoot a problem with my system. Currently, when a call comes in to a DID, it enters the queue and waits to be answered by a telecaller. However, after 20 seconds, the call gets disconnected automatically. Your task will involve: - Diagnosing and fixing the 20 seconds call drop issue. - Making necessary adjustments to the /etc/asterisk/ file. and agi script /usr/src/astguiclient/trunk/agi/ I believe the following lines may need to be removed: ; DID forwarded calls ;exten => _99909*.,1,Answer() exten => _99909*.,1,AGI() exten => _99909*.,n,Hangup() Skills and experience required: - Extensive experience with Vicidial and Asterisk. - Proficiency in AGI scripting
required a developer who has the knowledge about perl language and agi scripting platform: vicidial / asterisk
I'm looking for a professional with experience in setting up a minimalist Asterisk for Caller ID spoofing on AWS. Key Requirements: - Design and implement a system for Caller ID spoofing, specifically focusing on the ability to set Custom caller IDs. - Configure a secondary VOIP line as part of the project, but the main emphasis is on the Caller ID spoofing capability. - The system should be accessible via SIP TRUNK. Ideal Skills: - Extensive knowledge and practical experience with Asterisk. - Previous work with AWS configurations. - Expertise in VOIP and SIP TRUNK setups. - Understanding of Caller ID spoofing regulations and ethical considerations. Please note, I'm looking for a minimalist approach to this project. The system should be efficient, streamlined an...
I'm in need of a skilled professional who can configure and customize both Asterisk and Issabel for my project. Key Responsibilities: - Set up both Asterisk and Issabel - Configure call routing and IVR - Customize voicemail and call recording features - Set up user extensions and permissions - Implement security measures to protect the system from unauthorized access - Configure SIP/SIP Trunks for external connectivity Please only apply if you have substantial experience with Asterisk and Issabel, specifically in configuration and customization.
I need an efficient, scalable SIP-based inbound call center set up using Asterisk. The system should support a 'press 1' function and allow 3-5 people to be on the line simultaneously to handle incoming calls. Key Requirements: - Setting up an Asterisk SIP-based inbound call center - Implementing a 'press 1' function - Ensuring the system can support 3-5 people on the line simultaneously - Making sure the setup is efficient and scalable Ideal Skills and Experience: - Extensive experience with Asterisk - Previous work setting up call centers - Knowledge of SIP technology - Ability to create efficient and scalable systems.
I'm looking for a professional with specific experience in bulk calling systems, RTP optimization, and Asterisk configurations. The primary goal of this project is to effectively handle a high volume of calls. Key features of the system include customizable call routing. The main purpose of this routing system is to direct calls based on time zones. Essential Skills: - Expertise in bulk calling systems - Proficiency in RTP optimization - Asterisk configuration experience Desirable Skills: - Experience with time zone based call routing - Knowledge of customizable call routing systems - Skills in designing systems for high call volume handling
I'm looking for a developer to create a custom Android softphone for me. This softphone will primarily be used for voice calls and needs to be compatible with an Asterisk PBX system. Key Requirements: - The softphone should support high-quality voice call capabilities. - It should be fully integrated with the Asterisk PBX system. - Experience with Android development and VoIP technology is a must. - Understanding of Asterisk PBX systems will be a significant advantage. Please note that the softphone does not need to support video calls or messaging at this stage. The focus will be solely on voice call functionality.
This project is a text data analysis project. The goal of the project is to convert a file containing 36,000 book authors into a table of unique “wildcard” search terms ranked by their frequency. A “wildcard” search term is a character string ending in an asterisk. For example, the characters “ins*” would be a wildcard search term. When submitted into a search engine that accepts wildcards, “ins*” would return any word starting with the characters “ins”. For example, “insurance”, “insulation”, “instance”, etc. The project will involve the following main steps: * Parse the set of 36,000 book authors into separate words so that each word is a separate record. *&n...
I'm looking for a seasoned professional who can set up an Asterisk server for me. This server should be capable of recording calls, integrating with ChatGPT for customer service automation, and transcoding call recordings to email them to specific users. Key Requirements: - Set up Asterisk server with call recording capabilities. - Integrate ChatGPT for enhanced customer service automation. - Transcode call recordings and email them to designated users with the recordings attached. Ideal skills for this project include: - Extensive experience with Asterisk server. - Proficiency in integrating AI tools like ChatGPT. - Strong understanding of audio transcoding and email systems. Your expertise will help improve our customer service efficiency. PLEASE NOTE: I W...
...handle the signaling for voice calls. It's crucial to use SSL certificates to enable secure WSS connections. - Access Server via SSH: You will remotely access our server via SSH to carry out the entire setup. The entire process should be recorded as a screen recording, including accessing, setting up, configuring, and testing the server. - SIP Server Integration: The project involves configuring an Asterisk SIP server for call signaling. Ensuring integration between the WebSocket server and SIP to enable communication with Odoo's softphone is essential. - Nginx Reverse Proxy Setup: You will be setting up Nginx as a reverse proxy to forward secure WebSocket traffic (WSS) from clients to the backend server on port 8443. Ideal Skills and Experience: - Extensive experience...
I'm seeking a skilled developer to create an Asterisk AMI-based API for my SIP VoIP system. Key Requirements: - The API will need to interface with multiple third-party systems, including CRM software, billing systems, and softphones. - Proficiency in SIP is a must, as the system must strictly adhere to this VoIP protocol. Ideal Skills: - Extensive experience in Asterisk AMI and API development. - Prior work with SIP and its integration in VoIP systems. - Familiarity with interfacing APIs with CRM software, billing systems, and softphones. - Strong understanding of API design and software development principles. I look forward to receiving your bids.
I'm looking for an expert to configure ClearIP's inline proxy on my Asterisk VPS. The primary purpose of this proxy is to facilitate testing of my hosted VoIP setup. The selected professional will be responsible for: - Setting up the inline proxy on the VPS - Conducting tests on network performance and system integration of the VoIP - Ensuring the proxy is configured for optimal content filtering, performance improvement and security enhancement - Providing a comprehensive report on the tests results and any issues found Ideal candidates should have: - Extensive experience with configuring inline proxies - In-depth understanding of VoIP systems, particularly hosted VoIP - Proven track record in testing network performance and system integrations - Excellent skills in pr...
I am looking for a C programmer who is also familiar with Freeswitch to patch up a bug that freeswitch themselves have not given any attention to. I found what I needed within the source code, it is just a matter of now making it work correctly. This is not anything complicated, should not take more than 2-3 hours at the most from my experience. Key Responsibilities: - Debugging existing C code within the Freeswitch framework - Identifying and resolving functionality bugs impacting call routing Ideal Skills: - Proficient in C programming - Familiar with Freeswitch - Strong debugging skills - Excellent problem-solving abilities
Preciso de um app VoIP em react-native para conectar no meu servidor asterisk para comunicação entre os ramais por audio e vídeo
I am experiencing configuration errors with my Asterisk and VoIP setup, specifically with the dial plan. The main issue is incorrect call routing. I need an expert who can troubleshoot this issue, identify the root cause, and implement the necessary fixes. Ideal skills and experience for the job: - Extensive experience with Asterisk and VoIP - Strong understanding of dial plan configuration - Proven track record of resolving call routing issues - Excellent problem-solving skills - Ability to work efficiently under pressure
I'm looking for an expert who can seamlessly integrate Asterisk 18 with Salesforce via a CTI Connector. Key Features: - Enable click-to-dial functionality from within Salesforce - Ensure automatic logging of calls - Configure screen pop-ups on incoming calls - Setup an auto-dialer and voice broadcasting - Enable call transfer functionality to forward calls to other representatives - Integrate voicemail system for missed calls - Implement automatic call recording for quality assurance - Provide detailed call analytics and reporting within Salesforce - Set up an Interactive Voice Response (IVR) system - Enable custom caller ID for different departments or agents - Send real-time notifications for missed calls and voicemails - Create a system for call escalation to higher-level ...
I need assistance with my Asterisk setup. The system is not integrating correctly with my custom billing solution, causing multiple issues. The ideal candidate for this project should have extensive experience with: - Troubleshooting Asterisk - Integration of Asterisk with billing systems - Working with custom billing solutions - Understanding and resolving call quality issues Please only apply if you are capable of diagnosing and fixing the integration issues promptly.
I'm looking for an expert for Vicidial and configure it with GoIP for VoIP call routing. Key Requirements: - GoIP setup for VoIP call routing - Experience with Asterisk and Vicidial Additionally, I need security measures for the SMS Gateway. Specifically, implementing IP whitelisting to ensure only approved IP addresses can send and receive messages. Ideal Skills: - Asterisk PBX configuration - Vicidial setup and management - GoIP configuration - SMS Gateway security measures implementation
Necesito buscar a alguien que pueda instalar webrtc en un hosting y este se conecte con un asterisk para finalmente la web pueda hacer llamdas telfonicas mediante una llamada de telefonia IP. hay documentacion pero se necesita alguien que tenga experiencia.
Hello: I need your help to - Install and configure Freeswitch on my Linux server (Debian) - Ensure the system is set up for optimal performance by using open source libraries Please only apply if you have a proven track record with similar projects.
Configure Asterisk dialplan, so it can be integrated with Python scripts the client already has
