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    5,000 asterisk pbx proiecte găsite, la prețul de USD

    Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!

    $200 (Avg Bid)
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    Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,

    $300 (Avg Bid)
    $300 Oferta medie
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    Project for Anton F. S-a încheiat left

    Asterisk script

    $135 (Avg Bid)
    $135 Oferta medie
    1 oferte
    Server FreePBX Asterisk S-a încheiat left

    Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.

    $188 (Avg Bid)
    $188 Oferta medie
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    Project for Mucahit T. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.

    $1000 (Avg Bid)
    $1000 Oferta medie
    1 oferte
    Project for Macovei M. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.

    $250 (Avg Bid)
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    1 oferte
    Project for Ionut R. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de maxim 1000$

    $1000 (Avg Bid)
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    1 oferte
    Project for slk S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de la RDS

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    Project for MikeRRR S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la RDS

    $30 (Avg Bid)
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    1 oferte
    Project for ionut0709 S-a încheiat left

    Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.

    $150 (Avg Bid)
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    Specialist/Suport IT S-a încheiat left

    2-5 ani experienta in suport nivel 1 si 2 pentru sistemul de operare si aplicatiile Microsoft Cunostinte LAN, VPN, WLAN si Remote Connectivity, TCP/IP, la nivel de instalare si troubleshooting Cunostinte de diagnoza hardware si reparatii minore pentru laptop, desktop, imprimante, monitoare. proiectoare, solutii de videoconferitna, ...experienta in suport nivel 1 si 2 pentru sistemul de operare si aplicatiile Microsoft Cunostinte LAN, VPN, WLAN si Remote Connectivity, TCP/IP, la nivel de instalare si troubleshooting Cunostinte de diagnoza hardware si reparatii minore pentru laptop, desktop, imprimante, monitoare. proiectoare, solutii de videoconferitna, instrumente backup, alte periferice Cunostinte si experienta suport Telecom (PBX, RAS) Experienta in managementul stocurilor...

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    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 - $3000
    $1500 - $3000
    0 oferte
    FreePBX Project S-a încheiat left

    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 (Avg Bid)
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    In the search for an experienced developer, I require a web2call application based on Asterisk. This web2call should focus on enabling international call capabilities. Specifically, the created application should run adeptly on Linux platform systems. Additionally, the application needs to support one-on-one call functionality for its users. Hence, any freelancer bidding on this project should preferably have: - Proficiency in Asterisk software - Experience with building web2call applications - Knowledge of Linux platform compatibility issues - A background in creating systems for making international calls Your understanding of one-on-one call functionality support will be highly valued in this project. This project will be a significant contribution to connecting people ...

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    Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

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    Requiero la instalación y configuración de Festival y Cepstral en varias voces en español, así como hacer la configuración necesaria para que las grabaciones que se están grabando en WAV se graben en wav, todo lo anterior en asterisk 11.25 en Issabel.

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    I'm searching for a proficient SIP/Asterisk engineer who has ample experience in performing the following tasks: - Setting up and configuring an Asterisk server - Ensuring that the server is set up for high security and can support telemarketing or sales calls - Adding a specific function of calling and spoofing numbers, to be used exclusively for telemarketing and sales purposes Ideal skills and experience for the job: - Extensive experience with Asterisk and SIP protocols - Past-experience in setting up IVR systems - Understanding of server security measures - Familiarity with telemarketing systems Please provide a summary of your relevant work experience and your approach to this task in your bid.

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    Currently I have a problem where the VOS3000 cannot be used with the topology in the picture I attached. 1. NIC Card 1 (IP ) Is the IP of the Leased ...(IP ) Is the IP of the Leased Line that leads to the VOS in the Data Center. 2. NIC Card 2 (IP ) Is the IP of the Customer Router which is connected to the Customer IP Pbx. Currently VOS is installed on IP and the connection can be used to make outbound calls to the VOS3000 in the data center. when the customer makes a call via the IP segment on NIC Card 2 () via the Customer's IP Pbx (IP ) the call is immediately disconnected and there is no login on the VOS3000. Can you help me so that customers can make calls on NIC Card 2 in their IP Pbx Segment. Thank you very much

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    Voip Project 5 zile left

    I resolved issues with developed pbx based asterisk with dial plan and fax over sip

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    ...real-time speech-to-text functionality within the Asterisk ARI, leveraging the snoop channel feature. This project's goal is to create a seamless and accurate method of converting spoken words into text in real-time during calls, focusing on English language conversations. **Requirements:** - Proficiency in developing with Asterisk ARI and understanding of snoop channel operation. - Strong background in AI and speech recognition technologies, specifically tailored for real-time applications. - Solid experience with handling real-time audio streams and converting them to text. - Capable of integrating speech-to-text solutions in a Linux environment; though the operating system choice was skipped, Linux is our preferred platform for running Asterisk. - Ability ...

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    Free pbx Setup remote phones 1 zi left
    VERIFICAT

    We have a free pbx box and the phones work but we cannot get a phone to work outside of the network. We need help setting that up.

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    ...company providing managed wifi, managed video surveillance, MSP and hosted PBX, we require a freelancer or a team of freelancers to help us with consistently updating and improving our services. Areas of focus will primarily be: - Managed Wifi: Although we skipped the specifics on wifi service improvements, it is safe to say that overall enhancements in performance and reliability are essential. Relevant skills in network security, signal coverage, and user management might be required. - Managed Video Surveillance: Practical experience in IP camera systems and cloud video surveillance would be beneficial. - MSP: Familiarity with IT service management and broad knowledge in system infrastructure is preferred. - Hosted PBX: Understanding of telecommunications, VoIP and clo...

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    I'm seeking an expert in Asterisk PBX and Automatic Speech Recognition (ASR) to develop a real-time speech-to-text transcription system utilizing the Snoop channel in Asterisk RESTful Interface (ARI). The purpose of this project includes: - Enhancing the quality of customer service - Improving the accuracy of call transcriptions - Actively monitoring call quality The volume of calls for transcription has not been specified, which calls for a system that can efficiently handle variable loads. The freelancer should ideally have: - Experience in developing ASR systems - Proficiency in working with Asterisk PBX and ARI - Robust knowledge of Snoop channel, its functionalities, and potential The project's timeline is urgent, and it needs to be...

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    Hello Asterisk Developer - Engineer - Tech, 1. I need an Asterisk auto-attendant - IVR with a TTS text-to-speach greeting generator - plugin. Canned greetings are so last century - customers want tailored greetings in a life-like male and female voice.2. No cumbersome GUI to handle this. I currently have a GUI running my Asterisk Voicemail that is over 10 years old...simple, liteweight and has what I need.3. If you are not experienced with Asterisk, you probably should not you.

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    ...am seeking a developer with experience in Asterisk PBX and PHP to create a custom dashboard that interacts with the Asterisk Manager Interface (AMI). The dashboard should allow real-time monitoring and management of calls, as well as generating reports and statistics on system usage. Required Features: Real-time visualization of incoming, outgoing, and ongoing calls. Ability to initiate, pause, and terminate calls from the dashboard. Generation of reports and statistics on call traffic, duration, etc. Integration with Asterisk PBX administration functions via AMI. Intuitive and user-friendly interface. Required Skills: Demonstrable experience in web application development with PHP. In-depth knowledge of Asterisk ...

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    nesecito crear una configuración en Asterisk la cual todas las llamadas salgan por trunk diferentes estas de forma aleatoria., este proyecto lo nesecito lo mas rápido posible. nota: este plan de marcado tiene que tomar el calleid de cualquier plataforma o aplicación y asignarlo como callerid a la llamada saliente. le dejare adjunto lo que intente, sin suerte, porque no asigna el callerid

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    ...am currently seeking a skilled professional to execute essential configuration changes to my existing Issabel IP PBX setup. The project specifics include: 1. Changing the current SIP provider from Tata to Airtel. 2. Updating IP PBX configuration settings to seamlessly support the new Airtel SIP service. 3. Making necessary modifications to the SIP trunk settings to ensure optimal compatibility with Airtel's service. 4. Configuring additional SIP trunk registrations as required. The ideal candidate should possess proficiency in SIP trunk configurations, with a solid track record of experience in working with both Tata and Airtel SIP services. A clear understanding of Issabel IP PBX setups and the ability to configure SIP settings accordingly are crucial for th...

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    Experiencia en desarrollo y mantenimiento de sistemas con Asterisk, FreePBX, Issabel. Dominar el idioma español.

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    PBX Server backup & Sync S-a încheiat left

    ...perfectly synced includes user accounts and extensions, as well as call routing rules. This guarantees seamless user experience and operational consistency across both platforms. **Ideal Skills and Experience:** - Proficient with Asterix/PBX systems, particularly in backup and synchronization processes. - Strong understanding of disaster recovery protocols and system scaling strategies. - Expertise in real-time data synchronization techniques and tools. - Familiarity with the specific configuration nuances of user accounts, extensions, and call routing within PBX systems. - Proven track record of successfully upgrading systems to newer versions while ensuring data integrity and system stability. This project aims not just at making a functional copy, but also upgrading t...

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    ...Integrations:** - **Messaging Platforms:** WhatsApp, Facebook Messenger, email, and SMS. Additionally, PBX system integration is required to bridge our voice call capabilities with digital communication channels. - **Features:** Essential features we're looking for include: - **File Sharing:** Ability to share documents, images, and videos across all channels. - **Chat History:** Seamless access to previous conversations for context and continuity in customer support. - **Chatbot Integration:** Capability to integrate AI chatbots for automated responses and assistance. **Ideal Skills and Experience:** - Experience with API integration for WhatsApp, Facebook Messenger, email, SMS, and PBX systems. - Proficient in developing chat platforms with functiona...

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    I am experiencing an issue with the Asterisk 13 Audio upload and volume control API, particularly the volume control function. It is unresponsive and isn't altering the volume at all, despite my attempts to modify it. I've tried rectifying this issue by using alternative extensions to manage the volume, but with no success. The ideal freelancer for this project would have a strong knowledge and previous experience with the Asterisk 13 API, and a solid understanding of volume control functions. Troubleshooting and debugging skills are essential. The goal of this project is to ensure that the volume control functionality works impeccably, and responds accurately to adjustments made.

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    I require an expert to install and configure Asterisk on my Linux system. Specifically, I need the setup to efficiently manage and handle a maximum number of concurrent calls between 21-30. Given that I haven't specified the exact end-user features, my preference is for a freelancer who has comprehensive experience in deploying diverse functionality on Asterisk systems. Key skills: • Strong experience in setting up and configuring Asterisk on Linux platform. • A solid understanding of developing and managing Asterisk features. • Familiarity with call concurrency and optimization for 21-30 concurrent calls. I am looking for a competent freelancer who can ensure a seamless and efficient setup to meet my project demands. If you've got t...

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    ...construct an Asterisk application tailored for call center management. Plan for building Asterisk (SIP server). 1. Gather information about hardware and software requirements. In this case, we focus on CPU speed, memory size, and bandwidth. You will need a router or firewall to forward SIP traffic to your server. We use Asterisk as our SIP server software. User scenarios and support documentation are also provided. (Day 1). 2. Procure and build cloud servers according to specifications, such as the volume of calls to be processed. Emphasizes factors such as OS, storage, and security. Aim for 300 calls. We also set up a firewall and domain. (Day 2 and 3). 3. Install the latest version of Asteriks. You do this on the official website. (Day 4). 4. A SIP trunk con...

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    I'm seeking a skilled professional to create an efficient virtual PBX system with extensive features for my needs. Key Responsibilities: - Setup call forwarding feature to ensure smooth communication. - Implement voicemail functionality, allowing for easy message retrieval. - Enable an auto attendant feature, to direct calls intelligently and effectively. Must-haves for this role: - Experience in setting up and managing complex virtual PBX systems. - Proven ability to implement call forwarding, voicemail, and auto attendant features. This project involves managing over 11 phone lines and requires the capability for international calling. Previous experience with large-scale PBX systems and international telecommunication would be a significant advantage.

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    ...with AWS API and PBX systems to execute a project on a PHP 8.1 platform. The job primarily involves the creation of DNS Zone records for each sub-domain and making dynamic DNS records updates based on PBX activity. Responsibilities: - Create DNS Zone records for each sub-domain in a multi-tenant PBX, using AWS API with Route53 connection. - Update DNS records dynamically based on PBX activity. Key Requirements: - Understanding of AWS API and Route53 service. - Familiarity with PBX systems, PHP 8.1, and multi-tenant environments. - Ability to propose and implement a hybrid mapping scheme for sub-domains to each tenant, using a provided basis. Security: - Implement SSL encryption for API communication. Ideal Skills: - Extensive AWS API Experience. - ...

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    ...solution, utilizing both FreePBX and Asterisk servers to streamline incoming, outgoing, and call monitoring processes. Here's a glance at my requirements with the BSNL SIP Direct Truck : 1. FreePBX AND Asterisk Server Setup: I am looking for an experienced tech consultant who can not only advise but also assist in setting up both FreePBX and Asterisk servers. 2. Custom Bot Development: The bot should be responsible for calling, receiving, and monitoring calls directly from the software with features such as: - Automated call routing - Call recording - Call analytics - Call forwarding - Interactive Voice Response (IVR) - Real-time call monitoring Ideal skills would include a deep understanding of VOIP technologies, especially FreePBX and ...

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    The default values are MixPBX, VtigerAsteriskConnector i have a voip provider and i would like to add this service on my Yefiforce instance. I will be glad if there is a solution.

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    I am looking for an experienced developer proficient in Asterisk, particularly its API. The role involves integrating an established frontend UI web app with a backend PHP and Asterisk API. Refer to the attached document for comprehensive details. Only respond if you possess strong Asterisk and web development skills. Thank you!

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    Olá, gostaria de implementar um classificador de caixa postal no meu issabel PBX (AMD) atualmente não tem nada implantado.

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    I'm seeking a skilled developer for a crucial integration project involving Asterisk, ERPNext, and Vodaphone API. My main goal is to bolster my customer service capabilities. I'll need an expert who can blend these components seamlessly, ensuring a smooth workflow and real-time interaction that elevates user satisfaction while streamlining my business processes. What I require: - Implementation of Call Recording: Capturing and storing every customer interaction. - Intelligent Call Routing: Directing calls smartly for optimal customer service response. - Detailed Call Analytics: Insights into call patterns to refine our communication strategies. - Real-time Data Synchronization: Keeping our ERP system and telephony platform in perfect harmony, to reflect updates instantly...

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    Seeking an expert web developer with a focus on creating robust e-commerce platforms with custom IVR systems. Here's what I'm looking for: - **Website Layout Design**: Crafting an intuitive, aesthetic frontend for my e-commerce business. - **Backend Development**: * Robust backend structure ...customer service. * Integration with a database for dynamic call routing and personalized responses. **Ideal Skills**: - Proven experience with e-commerce platforms. - Expertise in VoIP technologies and IVR systems. - Ability to integrate advanced database-driven features. - Strong problem-solving skills and attention to detail. By entrusting you with this project, I hope to see my platform, "Asterisk," thrive with a user-centric design and an innovative IVR expe...

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    We are seeking an experienced freelancer to help integrate Asterisk with Node.js for our communication system. The goal is to implement key features such as call recording, call forwarding, and managing SIP trunk. We are open to using any open-source alternatives like FreePBX if it aligns with the project requirements. Responsibilities: Integrate Asterisk with Node.js for seamless communication. Implement call recording functionality. Set up call forwarding features. Manage SIP trunk configurations. Provide guidance and documentation for future maintenance. Requirements: Proven experience in integrating Asterisk with Node.js. - Familiarity with VoIP technologies. - Previous projects demonstrating call recording and call forwarding implementations. - Knowledge of SIP...

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    Outbound Cold Calling S-a încheiat left

    I'm launching a new company in the industry of Google PPC Ads and Website Design services. As part of the launch, I'm doing outbound reach campaign, I require a proficient freelancer to conduct cold calling for the business. Your main responsibility will be reaching out to potential clients and i...to potential clients and introducing our services (You don't need to close on call, just get their info). You'll need to have: - Effective communication and persuasion skills - A background in marketing or sales - Good english speaking without a thick accent Not required But Ideally, you'll have experience in Google PPC Ads and Website Design services. You will be given access to our Cloud PBX system - and calls are recorded. Please only bid if you are okay...

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    I'm looking for an experienced full-stack developer knowledgeable in Asterisk and React. The project scope involves building a VOIP system focusing on Asterisk. We need to integrate a vital functionality: call routing. This system will mainly be dealing with my business's customers. Hence, it's essential for it to be user-friendly and have a robust functionality. Ideal Skills: - Proficient in Asterisk and React - Previous experience in VOIP system development - Experience in customer-focused application development - Expertise in implementing call routing functionality This role requires technical and domain skills. You should have a strong understanding of the best practices in Asterisk systems and be able to efficiently implement a call...

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    I'm seeking an experienced developer proficient in Asterisk configurations and AGI scripting to enhance our customer support telephone system. **Essentials:** - AGI scripting for Asterisk - AI Voice Response integration **Key Goals:** - Develop an AGI script to automate voice responses for incoming and outgoing calls. - Integrate the system with openAI APIs to deliver natural, engaging voice interactions. **Ideal Candidate:** - Previous experience with Asterisk telephony systems - Strong proficiency in AGI script development - Familiarity with AI-driven voice response technologies - Proven track record with APIs We need to transcribe caller's voice and send it to openAI via api and get responses from there and convert it to voice again and play it ou...

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    I'm looking for a skilled Asterisk developer who can create an effective call tracking and routing system. This task will be from scratch, so you will not have any previous resources or guides to reference. Key project requirements: - Configure and implement call routing and forwarding -Develop GUI in react, backend in MySQL - Strong PHP skills are required to handle this task successfully - Proven experience in Asterisk and telephony systems Ideally, you should have a substantial background in setting up telephony systems from scratch. Demonstrated ability to solve complex problems in this area is highly desirable.

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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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    ...my PBX system, specifically involving calls coming from my trunk provider, Anveo. They're simply not connecting as they should, and this is disrupting my business operations: - Issue at hand: Incoming calls from Anveo fail to connect. - Error encounted: Each attempt is met with an error message, not a mere dial tone or busy signal. - Call flow breakdown: Calls are meant to go directly to an assigned extension, but that's not happening. For an expert to resolve this, the ideal qualifications would include: - Extensive experience with PBX systems, particularly troubleshooting connection issues. - Proficient in diagnosing and interpreting PBX error codes. - Knowledgeable about Anveo's services and common compatibility issues. - Ability to work with di...

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    I am seeking a skilled developer to construct a simple call forwarding system using Asterisk. Key functionality required includes forwarding calls to several numbers. This function should be dynamic, with the ability to add or edit the numbers to which calls are forwarded via a user-friendly web interface. Moreover, the web interface should give users the power to customize the call routing settings. Ideal candidates should have significant experience in VOIP technology, Asterisk systems, and developing intuitive web interfaces.

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    SIP jssip webrtc demo S-a încheiat left

    I need a demo sip web RTC client built with js sip for demo. This is a quick project and will only need a demo to work with asterisk. Asterisk is configured and running.

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