Huawei asterisk chan dongleemplois

**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...

Nous recherchons quelqu'un de capable de faire un logiciel de développement de solutions spécifiques aux véhicules. Pouvoir faire du DPF Off, EGR Off, IMMO Off, Dtc's Off.... Nous avons un...de solutions spécifiques aux véhicules. Pouvoir faire du DPF Off, EGR Off, IMMO Off, Dtc's Off.... Nous avons un logiciel de référence à fournir pour exemple. - Développer le logiciel sur la base du logiciel de référence et faire une refonte visuel. - Ajouter au logiciel une interface de gestion de compte et paiement. - Stocker les solutions sur un serveur. - Protéger le logiciel avec une clé dongle, voir plus avec vos compétences. Il faudra comprendre l'algorithme de modifi...

...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 ...

Nous recherchons un développeur en C# pour développer un logiciel Le tout sur une base de serveur protégé par un dongle

Création de logiciel avec clé dongle et sécurisé contre le hack

Offre d’Alternance/Stage fin d’étude Assistant(e) du Chef de Projet Chine Pré Embauche URGENCE Missions - Gestion des réseaux sociaux français et chinois des comptes de l’entreprise et de la marque de luxe (Linkedin ads, Facebook ads, Instagram, Twitter, Wechat, Tiktok et RED…) - Assister dans la préparation des...contrat d’alternance. Qui sommes-nous ? S2S GROUP est une agence de communication franco-chinoise basée en France avec une filiale à Shanghai. Nous offrons les services sur l’écosystème des réseaux sociaux chinois + Eshop pour les entreprises françaises souhaitant s’implanter en Chine ou l’inverse. Nous pouvons compter parmi nos clients fidèles de...

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

J’ai besoin de quelqu’un en mesure d’importer environ 50 produits avec leurs variantes + photo et description (iphone, samsung, huawei) en fichier CSV pour importer sur Shopify.

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...

...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

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

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

Estamos à procura de um especialista em Issabel e Asterisk para implementar um sistema de Resposta Vocal Interativa (IVR) na nossa plataforma de servidor Issabel 4.0.0-6. O objetivo deste sistema é realizar entrevistas telefónicas automatizadas onde os respondentes possam responder a perguntas pré-gravadas utilizando o teclado do seu telefone. O sistema deve ser capaz de carregar uma base de contactos para realizar automaticamente as chamadas telefónicas e capturar estas respostas numa base de dados para análise subsequente. Além disso, o sistema IVR deve incluir a funcionalidade de conversão de texto em voz (TTS) para facilitar a criação e atualização de prompts de voz. No entanto, também d...

...solution, I'm seeking guidance from a skilled freelancer experienced with both Raspberry Pi and Asterisk server setup. Key Project Details: - I do not require a full-fledged PBX setup, just a basic VoIP service facilitated through my Raspberry Pi. - The primary feature I'm interested in implementing is voice calling. Required Skills: - Proficiency in configuring an Asterisk server on Raspberry Pi. - Strong understanding of VoIP and related protocols. - Ability to guide and explain the setup process clearly. Your Role: I've tried to set it up but no voice can be heard on the other end. So this task is mainly a trouble shooting job. Your primary role will be to walk me through the setup of Asterisk on my Raspberry Pi, ensuring proper configuration f...

Have original dongle for a software with us. We need to either create soft copy and use system without hard dongle or make copy of hard dongle with new one.

I'm in need of a highly qualified and certified network and security engineer. Ideally, you should have certifications from Cisco, F5, Symantec, HP, HPE, Dell, Huawei, Avaya, Fortinet, and Juniper. Key Responsibilities and Tasks: - Network design and configuration - Firewall management - Network troubleshooting - Telecommunication system setup - Network security audits - Hardware installation and maintenance - System integration - Vendor selection and management - Providing ongoing support Your background should demonstrate extensive experience with all the above functions. I'm particularly interested in freelancers who have successfully handled similar tasks in the past. Your success will hinge on your ability to improve our overall network and security infrastructure. ...

I require an expert in Asterisk and FreeSWITCH development, along with experienced system engineering skills. Specifically, I need help with: - Asterisk and FreeSWITCH development - Integrating VoIP - Setting up Call routing and IVR (Interactive Voice Response) - CTI - Skill group base call routing to agents. Apart from these, the implementation of Asterisk and FreeSWITCH clustering as well as Call Center Reporting is required. The technology stack should be Linux, PHP, MySQL along with postgresql and API knowledge. The freelancer should have considerable experience in these to meet the project's specific requirements.

...persona con experiencia configurando la libreria de Javascript SIPML5 y asterisk. Tenemos todo instalado y configurado. El softphone web se registra al asterisk, emite y recibe llamadas, pero cuando se atiende la llamada no transmite el audio. El servidor tiene instalado una VPN. Cuando el usuario se conecta a la VPN, entonces funciona el audio de la comunicación pero cuando no se conecta a la VPN entonces vuelve el problema del audio. Se requiere que el softphone funcione sin VPN, solo por internet. Version asterisk 18 OS: Ubuntu Server 14 +++++++++++ A person with experience configuring the SIPML5 and Asterisk Javascript library is required. We have everything installed and configured. The web softphone registers to Asterisk, sends and recei...

I'm in need of a skilled programmer who can develop a software system that will replace an existing flash dongle licensing system. The primary goal of this software will be to manage and verify the licensing of our software. Essential Features: - Compatibility: The software must be Windows-compatible. - Licensing System: The software should implement a hardware-based activation system to ensure the protection of our software. Ideal Skills and Experience: - Proficient in software development for Windows environments. - Experience in developing hardware-based activation systems. - Knowledge of software licensing systems. - Strong understanding of encryption and security protocols. - Ability to create a user-friendly interface for license management. I'm looking for s...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.

...developed for delivering OTP messages through a web interface. Short codes will need to be delivered through web panel to customers. Text message delivery for multiple services used in accounts creation like facebook, instagram, gmail, whatsapp and other services.. This means we will be using physical sim cards, NOT virtual numbers. Web backend will need to communicate with physical hardware: (Moden huawei e303, 3131 FOR EXAMPLE) The system has to be the same as sms-activate, sms hub, sms pva, please if you have this experience I am searching for coders experienced is USSD/AT commands. Heavy experience with GSM/CDMA cellular networks and sms gateway integration. Look at the websites I've posted above, put together a compatible hardware list for the web frontend&backen...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error. [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descr...

...help with the setup of a Sonoff Zigbee 3.0 USB Dongle on a Synology NAS (DS920+). The device is not being recognized by the DSM 7.1 Linux-based operating system and does not show up in the /dev folder despite reinstalling the drivers. The chosen freelancer must be well-versed in Linux systems, troubleshooting device recognition issues, and experienced in the setup of Sonoff Zigbee dongles. Experience with automation using Home Assistant would also help. The task includes: - Analyzing the cause of the current recognition issue - Implementing a solution to get the Dongle recognized - Advising on further steps to avoid such issues - Ensuring the Dongle operates as expected - Ensuring Home Assistant running on Docker can see the Sonoff Zigbee dongle. For thos...

Creazione di un software che da un lato dialoghi tramite AMI (Asterisk management interface) con un centralino e dall'altro in base a delle condizioni impostabili, invii del messaggi tramite un gateway whatsappa già pronto chiamato che permette di inviare (pagando) messaggi illimitati tramite WhatsApp. inoltre questo software deve poter inviare dei messaggi anche da una lista di numerazioni che gli si fanno caricare.

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descri...

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

I need an expert who can modify my wife's Huawei Chinese phone to be fully compatible with American apps and servers. This is crucial as the phone was an expensive purchase and we can't afford for it to be useless in America. Key Requirements: - Experience in programming Huawei phones - Strong knowledge of the Android operating system - Ability to modify apps and servers to be compatible with the Huawei platform. The successful freelancer should have a track record of similar projects and provide a portfolio or examples of past work. Please do not hesitate to reach out for further information.

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50

We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors only w...

Task: Need to send a GSM command string ( USSD or SIM Tool Kit commands ) to request a pre-paid mobile credit send, credit recall, and balance check. These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Us...These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Use the Diafaan software or similar, and customize with it’s C++ / .Net scripting connector - Use other pre existing software Current Hardware: Windows Machine with remote access Port modem , port Huawei modem 3 SIM cards from L...

I require an experienced Huawei BSC 6900 expert to help with the configuration and optimization of our BSC 6900 system. Key tasks will include: - Focusing on efficient and optimal base station control. - Overseeing resource management to ensure proper utilization and efficiency. - Specifically dealing with ongoing connectivity issues. This project is a fit for you if: - You have prior experience with Huawei BSC 6900 system running R09 - You have resolved connectivity issues in the past successfully. - Your expertise lies in network configuration and optimization. Your goal will be to optimize the BSC 6900 system to ensure enhanced connectivity, resource management and base station control.

...lower in crowded cities. Therefore, we need to change the backhaul connection of our gateways to 4G mobile Internet. There are two approaches to this: 1) using an external 4G modem that has an Ethernet connection (), 2) connect a 4G USB stick to the USB connector of the Kerlink Wirnet Station, like the Huawei K5150 (), or the Huawei K5160 (). As we also have some of the aforementioned 4G USB modems, what we are looking for is a solution for the 2) option, that is the integration of the K5150 or the K5160 to the Kerlink Wirnet OS. The solution should be able to handle both public and private APN-s

Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.

Integrated Platform for Website, Control Panel, Content Management, and Mobile Apps (Android, iOS, Huawei) Overview: The project aims to develop a comprehensive platform that includes a website, a control panel, a content management system (CMS), and mobile applications for Android, iOS, and Huawei devices. This integrated platform will provide seamless access to content, management tools, and user interfaces across different devices, ensuring a cohesive user experience. Components: Website: Develop a responsive website with user-friendly interfaces. Implement modern design principles for aesthetic appeal and usability. Include essential features such as user registration, login, search functionality, and navigation menus. Ensure compatibility across various web browsers ...