Freeswitch voicexml asteriskemplois
...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...
**Project Description:** **Overview:** We are in need of an experienced system administrator with expertise in VoIP, Asterisk PBX, and Linux to configure our GoIP8 device. Our goal is to set up GoIP8 as a SIP trunk within FreePBX 14. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in configuring GoIP devices and SIP trunks. - Examples of previous projects where you successfully configured similar VoIP setups. - An outline of your approach to configuring GoIP8 as a SIP trunk within FreePBX 14. - Your proposed timeline for completing the configuration. - Your pricing structure for this configuration project. **Note:** We are looking for a reliable and efficient configuration of GoIP8 as...
...d'un freelancer pour m'aider à configurer Kamailio sur mon serveur et orchestrer plusieurs serveurs Asterisk. Mon objectif est de recevoir des appels de Twilio et Telnyx, et de mettre en place des règles entrantes et sortantes pour gérer ces appels. J'ai déjà installé un docker de Kamailio sur mon serveur, mais je ne sais pas comment configurer les premiers réglages. J'aimerais utiliser le mode file plutôt que la base de données, ce qui signifie que j'aurai un fichier de réglages contenant les adresses IP de mes serveurs, les adresses de Twilio et Telnyx, ainsi que les règles entrantes et sortantes pour gérer les appels. Je suis à la recherche d'un freelancer ...
mettre en place un cahier des charge pour un sophtephone ou webphone j'aimerais développer un shotphone, j'ai déjà un serveur Asterisk en place qui fonction et il me manque que le softphone pour mon application
je recherche à acheter, a louer ou à faire créer un predictive dialer pour les callcenter
...exceptionnelle de rejoindre une société belge de eCommerce et une équipe jeune et enthousiaste qui connaît une croissance très rapide. Profil recherché : - Sérieux : intégrité et stabilité. Références obligatoires - Attitude professionnelle, attitude le client est roi - Autonome (capacité de propre initiative) - désireux d'apprendre - bonne connaissance du web - Connaissance technique VoIP / Asterisk / SMS / Réseaux et serveurs - Langues : français et anglais écrits. Très bon niveau exigé Description du travail via Chat: - conseil client (soit le client initie le chat, soit l'agent invite le visiteur sur le site) : connaissa...
...exceptionnelle de rejoindre une société belge de eCommerce et une équipe jeune et enthousiaste qui connaît une croissance très rapide. Profil recherché : - Sérieux : intégrité et stabilité. Références obligatoires - Attitude professionnelle, attitude le client est roi - Autonome (capacité de propre initiative) - désireux d'apprendre - bonne connaissance du web - Connaissance technique VoIP / Asterisk / SMS / Réseaux et serveurs - Langues : français et anglais écrits. Très bon niveau exigé Description du travail via Chat: - conseil client (soit le client initie le chat, soit l'agent invite le visiteur sur le site) : connaissa...
besoin d'avoir un script php(perl/lua) pour créer un bridge en freeswitch pour initialiser un appel. si on appelle le script avec un numéro de téléphone alors on crée un appel entre une extension (3000) et le numéro appelé
...numéro de téléphone SIP fourni par OVH 1 serveur dédié sur lequel est installé KAMAILIO. Mon téléphone IP était directement connecté à ma ligne SIP chez OVH (Je pense que ma ligne SIP chez OVH est installée sur un serveur ASTERISK chez OVH) Mais depuis peu, mon FAI bloque les enregistrements et appels SIP. Mon projet est de configurer un serveur intermédiaire KAMAILIO pour l'utiliser en tant que PROXY SIP pour enregistrer mon téléphone IP sur KAMAILIO sur un port différent du 5060 et que ce dernier relais les appels entrants et sortants entre le serveur ASTERISK d'OVH (sur lequel est installer ma ligne SIP) et mon téléphone IP. ...
La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à - installer un trunk SIP sur un serveur Asterisk - installer minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel - installer un panel de gestion (XIVO ou autre)
Bonjour, Je chercher un expert Asterisk et le protocole SS7 pour mettre en place un script qui permet de terminer des appelles venant d'un serveur asterisk en utilisant un API SS7 : Trafic Voip -----> Asterisk ---->>API SS7 ----> Destination Cordialement,
Bonjour, Nous souhaitons déployer un système Asterisk avec des besoins de base précis mais particuliers. Nous avons surtout besoin d'un expert sur Asterisk et PBX en général, à qui on pourra poser des questions claires et avoir des réponses et des résultats. Pour des raisons de confidentialité ,nous ne publions pas ces besoins ici. Un accord de non divulgation devra être signé. Il est nécessaire d'avoir une très bonne connaissance d' Asterisk et d'être force de proposition afin de suggérer des solutions alternatives.
L'objectif est de connecter Vtiger CRM cloud edition avec un AsteriskNow. Le travail est en partie réalisé mais certains détails ne fonctionne pas. Nous recherchons quelqu'un pour peaufiner l'installation. Vtiger et Asterisk se trouve dans le même réseau et l'installation reste basique.
Nous avons un serveur asterisk configuré avec quelque sip trunk ovh. Nous avons des problemes dans la presentation des alias dans les appelles sortant. Nous avons besoins de quelqu'un qui connait bien se fournisseur Merci
Excellente connaissance de la gestion des ports USB, de UDEV, de asterisk et de sous linux centos et ubuntu
à la recherche d'un spécialiste VoIP pour l'installation d'un serveur asterisk, a2billing et fail2ban insertion des des rate plan...
Installation de opensips sur un serveur et asterisk/a2billing sur 2 autres serveurs
j'ai besoin de installer un serveur opensips avec une proxy rtp (rtpproxy) + serveur Freeswitch le tout avec debian jessie x64 comme sur le plan: Merci
Concevoir une page web (disponible dans l'intranet de la société) pour permettre de saisir un numéro de téléphone. Ce numéro doit être utilisé par asterisk pour activer une déviation de numéro. Actuellement le numéro en question est codé en fixe dans le fichier de configuration
Besoin des competneces pour des scripts lua avec freeSwitch et avec le mode SMS et chat
Je souhaite que le site soit mieux référencé naturellement pour mettre en avant nos compétences en développement "Asterisk", "VoIP" et "CRM"...et autres à définir.
LWS, société spécialisée dans l'hébergement web, recherche un développeur apte à monter un serveur asterisk complet. Une fiche explicative complète vous sera fournie. Ce projet conviendra à un développeur confirmé, ayant déjà monté un serveur Asterisk évolué : menu, logs, file d'attente etc INDISPENSABLE : Langue française (toutes les explications vous seront données en français)
LWS, société spécialisée dans l'hébergement web, recherche un développeur apte à monter un serveur asterisk complet. Une fiche explicative complète vous sera fournie. Ce projet conviendra à un développeur confirmé, ayant déjà monté un serveur Asterisk évolué : menu, logs, file d'attente etc INDISPENSABLE : Langue française (toutes les explications vous seront données en français)
Je voudrais installer un ipbx asterisk dans mon entreprise qui comporte 3 sites. Le projet comporte l'installation sur paris et un transfert de compétences pour pouvoir se débrouiller avec les pbs maintenance globale serait affectée à celui qui remporte le projet Cordialement
Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip
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.
...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.
