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    2,000 asterisk web dialer travaux trouvés au tarif de EUR
    Goip and freepbx S'est terminé left

    **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...

    €117 (Avg Bid)
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    3 offres
    Configuration de Kamailio S'est terminé left

    ...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 ...

    €254 (Avg Bid)
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    Developpement Android S'est terminé left

    une application Dialer qui peut faire un appel GSM ou SIP. Au moment de l'appel il vous demande si vous voulez passer par le GSM ou le SIP. Si vous choisissez le GSM c'est l'appel normal qui est lancé et si vous choisissez le SIP, l'appel passe par un PBX. Le SIP supporte le codec G729g. Les parties GSM et SIP sont toutes intégrées dans l'appli. Vous devez fournir le code source pour nous permettre de modifier certaines parties de l'application.

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    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

    €240 - €719
    €240 - €719
    0 offres
    predictive dialer avec asterisk S'est terminé left

    je recherche à acheter, a louer ou à faire créer un predictive dialer pour les callcenter

    €131 (Avg Bid)
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    2 offres

    ...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) : connaissance des produits et services de notre société - prise en charge d'un problème avec le client ou le visiteur et avec l'assistance de l'équipe technique, résolution du problème via les outils de CRM - aides techniques - Aide au développement de la soci&eacu...

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    Urgent
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    Support francophone pour site VoIP S'est terminé left

    ...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) : connaissance des produits et services de notre société - prise en charge d'un problème avec le client ou le visiteur et avec l'assistance de l'équipe technique, résolution du problème via les outils de CRM - aides techniques - Aide au développement de la soci&eacu...

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    Configuration d'un serveur KAMAILIO S'est terminé left

    ...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. ...

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    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)

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    Codez un logiciel S'est terminé left

    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,

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    Asterisk PBX S'est terminé left

    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.

    €591 (Avg Bid)
    Urgent LDN
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    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.

    €201 (Avg Bid)
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    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

    €28 / hr (Avg Bid)
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    Excellente connaissance de la gestion des ports USB, de UDEV, de asterisk et de sous linux centos et ubuntu

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    à la recherche d'un spécialiste VoIP pour l'installation d'un serveur asterisk, a2billing et fail2ban insertion des des rate plan...

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    Installation de opensips sur un serveur et asterisk/a2billing sur 2 autres serveurs

    €337 - €539
    Local
    €337 - €539
    0 offres

    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

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    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.

    €7 / hr (Avg Bid)
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    Mise en place d'un serveur Asterisk S'est terminé left

    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)

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    Installer un serveur Asterisk S'est terminé left

    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)

    €492 (Avg Bid)
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    Codez une application iPhone S'est terminé left

    Réalisation d'un SDK embarquant un dialer et implémentation d'un protocole permettant d'envoyer des clic to dial vers ce dialer

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    Création d'un dialer speciale S'est terminé left

    Premièrement vous devez être sure de pouvoir avoir les etat des appelles sortant (Outgoing call states), (en d'autres termes si le correspondant est occupé, hors réseaux, si ça sonne, ...Etc), on a 7 application à faire, pour un lancement d'un smartphone dans une région ou ça marchera forcement, nous pouvons payer, ou donner un pourcentage sur les bénéfices des ventes du smartphone, contactez moi pour plus d'information.

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    installation ipbx asterisk S'est terminé left

    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

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    Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip

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    ...conversion initiatives forward. Responsibilities: Utilize VICI Dialer efficiently to manage and handle customer interactions effectively. Provide exemplary customer service and sales support to clients, ensuring their needs are met with professionalism and efficiency. Conduct data entry tasks in Excel with accuracy and attention to detail. Performance Metrics: The Call Center / Customer Service / Sales Expert shall be evaluated based on the achievement of CONVERSION quotas set forth by our company. Quality of conversion calls, including professionalism, clarity, and adherence to scripting guidelines. Customer satisfaction ratings based on post-conversion interaction surveys. Requirements: Proven expertise in utilizing VICI Dialer for call center operations. Extensive ...

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    I am urgently seeking an experienced telephony and data processing specialist to configure my Grandstream UCM6302A with Asterisk. The core functionality required includes receiving calls, playing a welcome message, meanwhile working with Caller ID and Web API to determine where to forward the call. When a call comes in, • first a welcome message is played () • in the meantime the caller ID will be sent to web API preferably POST, but get can be if POST is not possible () •The API will respond json array: - {forward_to: 33356853 } - Forward the call to 33356853 - {forward_to:0 } - Play message and terminate the call • If forwarded call is not answered by Agent in three rings, another call to API will

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    ...am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Efficiency Goal: ...

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    i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much

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    I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.

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    I'm looking for a top-notch professional with extensive experience in API Integration, web-based applications, and direct server formats. This project requires setting up cold bulk SMS and Email marketing campaigns for both individual customers and businesses. Our predominant content will be promotional offers. Auto dialer is also required read the description below: I am launching a project that requires an expert in SMS marketing, auto dialer and email marketing for customer engagement. The project's goal is to distribute product updates to the general public, instilling premium brand integrity and visibility. Email marketing querry: I want to send 1mil emails a day on filtered and clean data. Now this data is not cleaned manually it has gone through a...

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    I have a particular project requiring expertise in Freeswitch, preferably with exposure to a Debian Linux environment. The key objective is to use Freeswitch for answering machine detection. The main requirement is to capture and analyze voicemails. We want to build a custom module or ready solution which can detect human or machine on the outbound dialer. The solution should work on any freeswitch version and without any licensing restriction. Please note, we have already used AVMD module and it just detects beep. We want to detect the machine or human from the initial screening of voice only. Skills and Experience Needed: - Solid experience with Freeswitch answering machine detection - Familiarity with working on Linux systems - Ability to effectively capture and analyze audio...

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    ...featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can provide the necessary Asterisk API documentation. Ideally, you are able to show me a s...

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    Hello, I operate a fax communication system leveraging Hylafax, integrated with an Asterisk server and iaxmodem, all running on Alpine Docker. While our outgoing fax functionality performs flawlessly, we are encountering persistent issues with incoming faxes. Specifically, incoming fax pages frequently get cut off midway, resulting in incomplete document reception. We are in search of a seasoned Hylafax professional who can diagnose and rectify this particular issue. Expertise in managing Alpine Docker environments and Asterisk/iaxmodem configurations will be highly regarded. Desired Expertise: Demonstrable experience with Hylafax, especially in fixing issues related to incoming faxes. Deep knowledge of Asterisk and iaxmodem. Proficiency with Docker containers, pre...

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    VoIP Efficiency & Quality Upgrade S'est terminé left

    ...seeking a VoIP consultant for improvement of my existing computer-based VoIP system. The purpose of the project is twofold - improved communication efficiency and enhanced call quality. Key Tasks: - Analyzing the current computer-based system setup - Implementing the connection of Physic SIP to asterisk on the cloud for enhanced call quality Ideal Skills and Experience: - Proven experience as a VoIP consultant - Excellent knowledge of IP PBX system - Experience with connecting Physic SIP to asterisk on the cloud - Ability to improve communication efficiency and call quality. Kindly submit your proposal outlining your plan to achieve these two goals along with your previous relevant work. Looking forward to finding a VoIP specialist who can provide a swift and efficie...

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    Using another dialer now and want to see if vici can improve our stats in terms of connection, answers, voicemail detection, autorotate to local area codes, spam detection etc.. Basically looking past the basic setup and someone who has used vici and knows how to add code into the system. Put "I can help" in the first line so I know you read the entire job post. Also please detail some things you could suggest in terms of adding codes. We are calling home owners in the US.

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    ...developer experienced in WebSocket/AudioSocket technologies and Asterisk integration to develop a solution that enables real-time transcoding with OpenAI Whisper through gRPC. Requirements: 1) WebSocket/AudioSocket Integration: Develop WebSocket/AudioSocket functionality to facilitate real-time audio communication with OpenAI Whisper. 2) gRPC Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [a...

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    I am searching for a skilled software developer with a strong background in Asterisk, Dialer, IVR and VOIP technologies. Although I haven't specified particular functionalities, general familiarity with call routing, call recording and interactive voice response (IVR) would be beneficial. The ideal candidate for this job should be proficient in: - Designing, implementing, and maintaining Asterisk software - Developing dialer functionalities, with emphasis on auto dialing, click-to-dial, and predictive dialing - Ensuring system is up-to-date and secure Freelancers who apply should provide any past work, detailing their experience and including project proposals, if any. If you believe that you have the expertise to effectively take on this projec...

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    Application for Asterisk 18 S'est terminé left

    We need to create an Asterisk aplication (v18) for Service at workshop by appointment for vehicles. This aplication must have voice recognition in English /Spanish language and Text to speach language with Google technology. Functionality: i) Welcome. ii) Select Languague. iii) Request Data: * Type of vehicle * City * Car licence plate * Telephone number. * Date request. * Time request. d) System will confirm first date/time available and customer will confirm. At this time, application will not have conectivity with real system....only must confirm next day and time users told. But it will have errors control, confirmation recognized data, etc.....

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    ...engineer to implement an Opus encoder and decoder in C# for seamless integration with Asterisk. The project involves handling voice audio from Asterisk, decoding it, incorporating Text-to-Speech (TTS) functionality, and encoding the synthesized speech before sending it back to Asterisk. The ideal candidate should possess the following skills and experiences: - Proficiency in C# programming language. - Extensive experience with audio processing and Opus codec. - Familiarity with Asterisk, SIP, and IVR systems. - Knowledge of Text-to-Speech (TTS) integration. - Ability to deliver high-quality code within specified timelines. Main Tasks: - Implement Opus encoder and decoder in C#. - Integrate the solution with Asterisk for audio processing. - Incorporat...

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    I need a specialist to develop an inbound call center dialer to optimize call efficiency. The key elements of this task include: - Maximizing efficiency: The primary objective of this project is to streamline call handling to boost efficiency, greatly reducing idle time between calls. - Email Integration: The dialer needs to have robust email system integration capabilities to facilitate seamless outbound communication. - Call Recording: The dialer should possess call-recording functionality, facilitating quality assurance and improving our customer service efforts. Prospective freelancers are ideally experienced in telecommunication systems, dialer technology, integration of systems, and understand general call center operations. Experience specifically wit...

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    In our quest for an outstanding and interactive VoIP web app, we're in need of an experienced developer, ideally one who is well-versed in Asterisk. The main features we're after, though not limited to, are call logging and reporting, an IVR system, and call routing and forwarding capabilities. Key Skills and Experience: - Extensive experience in VoIP development. - Proficiency in Asterisk. - Ability to develop an IVR system. - Experience in call routing, call forwarding, and call logging mechanisms. Given the nature of our project, it’s crucial to have a level of experience in these areas. Understanding the intricacies of these features is what will drive the success of our project. Though we have not specified it in the initial questions, poten...

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    Raspberry pi 4 video-doorbell S'est terminé left

    I have install Asterisk (with freepbx), for gui to a Raspberry pi 4. I need a DEVELOPER TO WRITE a succesfull CODE and configurate the system to: Make a AUTO-VIDEOCALL from extention ''0'' to extention ''1'' or ''2'' or BOTH at the same time. The exctention ''1'' or ''2'' are log in to androip application, called ''PortSin'''or another app called ''Linphone''. Extention ''0'' is already loged into Raspberry with the programm ''Linphone''. When i press a button connected to GPIO 21 i want the audiocall to be started for 9 seconds. If the call is anwsered I also need to OPEN the DOOR by pushing ...

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    I'm seeking a skilled professional to set up a new VoIP asterisk system with primary features of call routing and call recording. Must have the ability to have custom caller ID for outgoing calls. This project entails configuration for a small scale operation with less than 10 users/phone lines. Key Job Requirements: - Proficiency in setting up and configuring VoIP systems - Outstanding knowledge of asterisk - Experience with call routing and call recording This task requires an efficient and effective approach, understanding the needs of a smaller user-base. Someone with a strong track record of setting up VoIP systems and the knowledge to troubleshoot potential issues would be perfect.

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    Asterisk Phone System Enhancement S'est terminé left

    We used Asterisk as our phone system And we used Flutter's Sip-UA as a client The communication platform is WebRTC The problem we have is that when the internet suffers a few packet losses during a call, the client leaves the channel and then it is completely silent until the call is disconnected. We simulate the same scenario with a softphone, after the internet is disconnected and reconnected, the call continues and there is no problem. Are there any friends who can guide me?

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    Seeking a professional to integrate Asterisk VoIP into my existing Odoo system for my call center operations. Key tasks include: - Installing Asterisk VoIP - Setting up integration with Odoo - Implementing Call Recording functionality - Configuring Interactive Voice Response (IVR) - Implementing Automatic Call Distributor (ACD) The system should be able to handle 21-30 concurrent calls at peak times without any lag or quality degradation. Skills & Experience Required: - Proficiency in Odoo and Asterisk VoIP setup and integration - Proven track record in call center technology setting-up - Ability to work under strict deadlines and handle pressure - Understanding of call center operations and key processes - Excellent problem-solving abilities - Attention to d...

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    Veuillez vous inscrire ou vous connecter pour voir les détails.

    À la une Urgent Scellé LDN

    I am in need of a specialist for customizing my Asterisk PBX system. The primary tasks will focus on two areas: call routing/forwarding and call recording/monitoring. Call Routing and Forwarding: I require an extension based routing system to be developed. Call recording and Monitoring: The second part of this project revolves around call recording and monitoring. Specifically, I require an on-demand call recording system, coupled with real-time call monitoring. I am expecting the freelancer engaging with this project to have a deep understanding of Asterisk PBX system customization, particularly in call routing, forwarding, recording, and monitoring. Prior experience in similar projects will be a great advantage. Strong communication and troubleshooting skills will be ke...

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    I am setting up asterisk with Amazon Chime for sip trunking, but am very new to this and am having difficulty where making a test call doesn't work. If you think you can help, please bid the amount you would charge, and use the word "alarm clock" somewhere in your bid so I know you read the description and the bid isn't automated. If things go well, I will likely need more help in the near future setting up other details of the Asterisk config. For now though I am just trying to get a test call to work.

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    AWS Asterisk VoIP and SIP Setup S'est terminé left

    ONLY APPLY IF YOU HAVE EXPERIENCE IN VoIP and SIP. I am in need of a skilled freelancer with intermediate experience in Asterisk to set up a VoIP and SIP trunk on AWS. The successful candidate should have solid background with AWS infrastructure and Asterisk application. Key Deliverables: - Set up VoIP on AWS using Asterisk - Implement a SIP trunk functionality Requirements: - Must have experience in setting up and managing Amazon Web Services (AWS). - Asterisk application experience is essential. - Previous work on similar projects will be advantageous. Please make sure to include your experience with Asterisk and AWS in the application. I look forward to receiving your proposals and working with you on this exciting project.

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    I am seeking a specialist to help integrate AWS with our Asterisk PBX system. Key responsibilities will include: -Diagnose and resolve organic AWS to Asterisk PBX connection issues -Assure AWS and Asterisk PBX platforms interface correctly Ideal skills: -Extensive experience with both AWS and Asterisk PBX is essential -Experience with system integration and troubleshooting -Ability to work independently and find creative solutions to complex problems Whilst the specific functionalities and preferences for this project have not yet been finalized, these will be communicated upon initiation of the project. Your experience and suggestions will be valued in these discussions.

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