Freeswitch voicexml asteriskงาน
เขียนหน้า GUI เพื่อสั่งงาน Elastix , Asterisk
Hi, Please understand the problem we are facing with a specific use case. You need to have knowledge of Twilio, SIP, and Asterisk. We are trying to achieve the following use case: A caller dials a Twilio number. After some time or under certain conditions, Twilio forwards the call to an Asterisk server. Asterisk receives the call from Twilio and immediately connects the caller to sip:test (essentially bridging the caller directly to the SIP address). Once the call is connected between the caller and sip:linphone, Twilio should no longer be involved in the call (to avoid duration charges). The desired flow is: Caller → Twilio → Asterisk → sip linphone At the end, the connection should be: Caller → sip inphone We understand t...
...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'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 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...
hello, I have a very old CentOS 5.5 installed only for a single application : Asterisk. I need to move the current server including Asterisk (1.6) to a new environement (on AWS). I'm looking for someone who knows Asterisk and can help migrate keeping the same OS & Asterisk releases.
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
For Asterisk, in php, checking AGI/AMI scripts
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
I need an expert to integrate my Asterisk IP PBX with WhatsApp for receiving customer support calls via SIP. This project involves both setting up new WhatsApp accounts and integrating with existing ones.
I'm dealing with intermittent connection drops on my Asterisk system, which is running on a cloud-based service. Ideal Skills and Experience: - Proven expertise in diagnosing and resolving Asterisk connectivity issues. - Experience working with cloud-based Asterisk services. - Strong understanding of SIP protocols and VoIP networks. - Excellent problem-solving skills and attention to detail.
I’m working on a project that aims to automate calls, includi...PBX Account: I have a Skyline account with a phone number, username, and password. - Issabel: Using Issabel to manage calls, connect Skyline PBX, and handle SIP configurations. - Python for Call Handling: I need Python to interact with Asterisk, answer calls, and handle STT/TTS and GPT processing. Current Issues 1. Asterisk ARI Compatibility: - I found ari-py, but it’s outdated and works only with Python 2. - Tried asyncari, but faced numerous compatibility issues with anyio and ARI event handling. 2. Goal: I need to connect Python with Asterisk to: - Answer incoming calls and handle events. - Process voice input (STT) and generate responses (GPT)....
I'm in need of an expert who can carry out advanced, custom configurations on my Asterisk SIP server. The job will also involve integrating the server with various other communication platforms. Experience with Asterisk and SIP server configuration is a must, as is proficiency in system integration.
I need to stream live call audio from Asterisk to a WebSocket and vice versa. The purpose of this is to facilitate AI-based interactions during the calls, specifically for conversation analytics. Key Requirements: - Set up a system to stream live call audio to a WebSocket and back to Asterisk. - Enable AI-based interaction for conversation analytics. - Capture and analyze various data points, including sentiment trends, keyword frequency, and call duration patterns. Ideal Skills: - Experience with Asterisk and WebSocket. - Familiarity with AI and conversation analytics. - Proficient in setting up live audio streaming systems. The system must fully integrate with the AI to provide real-time analysis and responses.
De acuerdo a la información suministrada por el cliente, el trabajo consiste en diagnosticar el origen del problema al momento de marcar desde el terminal Bitrix24 usando el siptrunk integrado con asterisk. Requerimientos: - Acceso a terminal Bitrix24 vía anydesk, con posibilidad de instalar software Wireshark para realizar captura de paquete y emulación del problema. - Acceso a FreePbx. Procedimiento: - Acceso a Freepbx para validar operatividad/configuración de troncal SIP. - Conexión a terminal Bitrix24, se procederá a instalar software Wireshark para captura de paquetes. - Emulación del problema y captura de paquetes. - Análisis de captura y tráfico de paquetes SIP entre el endpoint, PBX y CRM. Costos: - El costo ...
