Asterisk softphone symbianJobs
Integrate Asterisk + UniMRCP + AWS LEX
I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9
We must have set up a new asterisk server that follows the normal asterisk guidelines. We also need a script where we provide the telephone number (DID), opening hours also agents who are queuing, then it must do the setup in configs.
Vi søger freelance konsulenter til telefonisk fra egen lokation eller vort kontor at arrangere møder indenfor forsikring, IT og reklame. Der faktureres fra din side, når mødet er gennemf...din side, når mødet er gennemført. Provisionen er en procentuel del af vor pris for mødet. Vi arbejder ud fra etiske retningslinier, f.eks. at vores organisation skal være så transparent som muligt. Derfor er de priser vi tager for møderne selvfølgelig frit tilgængelige for dig som samarbejdspartner. Vi har brug for både dansk-, engelsk- og svensk talende personale. Vi sørger for al software, softphone og holder personalemøde [Removed for encouraging offsite communication which is against ou...
Vb6 ivr asterisk dialogic programme
Sonetel & Asterisk Configuration
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Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.
The current configuration application speed of my Issabel Asterisk system is consistently slow, regardless of recent updates or changes. This affects my daily operations as I apply configuration changes every day. - I need an expert who can diagnose and fix this issue, ensuring that configuration changes can be applied swiftly and efficiently. - Ideal candidates for this project should have extensive experience with Issabel Asterisk, and a strong understanding of VoIP systems. - Previous work in optimizing system performance and troubleshooting similar issues would be a significant advantage.
I'm looking for a skilled technician with proven expertise in building and configuring a VOIP system. This system needs to support call routing and be compatible with softphones on Linux. We are planning to do this business. Some of our customers need to use th...Design and build a custom VOIP system - Integrate the system with a SIP trunk provider - Implement call routing functionality - Ensure compatibility with Linux-based softphones The ideal candidate will: - Provide the necessary source code for the VOIP system - Offer long-term support and cooperation - Have a deep understanding of VOIP systems and SIP trunk providers Experience with Linux and softphone configuration is a must. Please only apply if you can provide a custom-built VOIP system and are open to a long-term...
...follows: The call starts from the Twilio SIP trunk. Under this SIP trunk, I have configured my Asterisk server. Every call begins when the user dials the Twilio number, and it is received by the Asterisk server. My Asterisk server immediately calls the Twilio SIP domain. Under the Twilio SIP domain, I have a webhook that routes the call to my worker. If my Twilio client’s workers do not respond within 30 seconds, I use Twilio Refer to send the call back to my Asterisk server, which then connects to my backup SIP. The backup SIP is a simple SIP for Linphone. Flow: Caller → Twilio SIP Trunk → Asterisk Server → Twilio SIP Domain → Worker (if available) If no response from workers: Twilio SIP Domain → Asterisk Server...
...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...
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 that we will ...
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
I'm looking for a talented developer to create a softphone app for iOS and Android. The core functionality of the app should be voice calling. Key requirements: - A strong background in mobile app development, particularly with VoIP apps. - Proficiency in iOS and Android development. - Experience with implementing voice calling features. - Ability to deliver a user-friendly, high-quality app. Please note, there are no requirements for additional features like call recording, voicemail, or conference calling at this stage.
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 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'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
Description: We seek an experienced developer or team to implement a softphone app using Linphone for iOS and Android platforms. The app will need to integrate seamlessly with our existing Flutter modules. Key Details: • Designs are ready: All UI/UX designs will be provided. Your task is strictly implementation. • Platform: Native iOS and Android apps, with Flutter compatibility for integration with our current app modules. • Framework: Utilize the Linphone library for VoIP functionality. More Details on the Attached PDF
...templates for common tasks (e.g., logo replacement). A live session on performing customizations, with Q&A. 3. Adding RabbitMQ Pub/Sub Support Scope of Work Integration of RabbitMQ: Set up RabbitMQ server and configure the MicroSIP client to interact with it. Implement RabbitMQ pub/sub messaging: Publish messages for call status updates. Subscribe to external events to trigger actions in the softphone (e.g., auto-dialing). Testing & Validation: Unit testing of RabbitMQ messaging within MicroSIP. Documentation on configuring RabbitMQ in the deployed app. Deliverables Source code updates to include RabbitMQ support. Documentation detailing: RabbitMQ setup and configuration. Integration within MicroSIP. A live training session on: Understanding RabbitMQ int...
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.
