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    2,000 asterisk freepbx vicidial vicidialnow trixbox elastix iş bulundu, ücretlendirmeleri EUR

    Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.

    €782 (Avg Bid)
    €782 Ortalama Teklif
    4 teklifler

    Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!

    €1089 (Avg Bid)
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    5 teklifler

    Merhaba, 40.000 Kullanıcı kapasiteli, eş zamanlı 10.000 çağrıyı kaldırabilecek Asterisk kurulumu için destek verebilecek bu konuda IP Telephony, Call Center, Voip, Network konularında tecrübeli çalışma arkadaşları arıyoruz.

    €7054 (Avg Bid)
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    4 teklifler

    Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

    €175 (Avg Bid)
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    meraba, freepbx yazılımını kendi markam ve logom ile kurmak yönetmek istiorum. yardımcı olabilir misiniz.

    €93 (Avg Bid)
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    benim için bir CRM yazın. elastix ile entegre çalışacak. alım , satım , kargo takibi , otomatik arama , data yükleme , gelen çağrıları çalışanlara aktarma ve arayan bilgisini ekranda gösterme , sms gönderimi (api kodları verilecektir.) mail gönderimi. gibi özellikler.

    €1318 (Avg Bid)
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    Merhaba kingAsterisk, profilinizi farkettim ve size projemi teklif etmek isterim. İstediğiniz ayrıntıyı sohbet üzerinden tartışabiliriz. skype=mahireker--germany i need a custom reporting after this work new work email and sms sending from vicidial to customer

    €232 - €232
    €232 - €232
    0 teklifler

    Yusuf Bey Merhaba, STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimiz şekilde çalışıyor olmasına rağmen dtmf detection sorunumuz var, ne yaptıysak asteriskin dtmfleri yakalamasını bir türlü başaramadık, bu konuda bize yardımcı olma şansınız var mıdır?Teşekkürler.

    €46 (Avg Bid)
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    1 teklifler

    Php vicidial custom reporting Wè needs a custom reporting tool for vicidial

    €27 - €227
    €27 - €227
    0 teklifler

    elastix yazılımda otomatik arama ve raporlama yapabilceğimiz ses dinletip tuşlamaları , ulaşılamayanları , cevapsız çağrıları görebilceğiniz gelişmiş raporlarma.

    €232 - €695
    €232 - €695
    0 teklifler

    ip santral üzerinden faks gönderip almak istiyoruz. elastix yada freepbx başarılı çalışmıyor yada geliştirilmesi gerekli.

    €758 (Avg Bid)
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    2 teklifler

    Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

    €28 - €232
    €28 - €232
    0 teklifler
    Asterisk - Pbx Bitti left

    Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

    €139 (Avg Bid)
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    1 teklifler

    Asterisk üzerinde çalışan crm yazılımını geliştirmek için proje desteği istiyoruz yapmış olduğumuz crm yazılımına data arama ve data kayıtlarının tutulduğu arayüz entegrasyonu yapılıcak.

    €25 / hr (Avg Bid)
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    Şuan kullanıdığımız bir elastix sistemimiz santral anonsu çektiriyoruz ama elastiğin istediği formata çevirince ses anormal bir çözümü olduğunu zannediyorum.çünkü gayet güzel ananslar var sesi bozulmamış.Elastix in istediği format 16 bits ve 8000 hz. Ben 8503404850 ye yüklediğim anonsu premierpro da yaptııma gelen iki seçenek bunun elastix tarafında bir modül tarzı bir şeyi var,yani düzenlemesi bu sesi elastiğin istediği formatta ama sesi bu kadar bozmadan dönüştürebilmemiz laızm. Örnek Yüklenmiş Bozuk ve Boğuk Ses : 0850 340 48 50 sesin orjina hali o kadar net olmayabilir ama Örnek Güzel Santral Anonsları : 444 2 444 ve 444 5 666

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    €68 (Avg Bid)
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    Merhabalar, şirketimiz elastix programı kullanıyor. Bunun için özel bir modül hazırlatmak istiyoruz. Gelen, giden çağrıları kontrol edebileceğimiz, dinleme ve konferans yapabileceğimiz gelen, giden çağrıları kaçan çağrıları görebileceğimiz ayrıca çağrı, agent raporlamasını basit bir arayüz ile alabileceğim bir modül arıyorum. Konu hakkında görüşmek isterim. Tel :[The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

    €515 (Avg Bid)
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    Merhabalar, şirketimiz asterix tabanlı elastix programı kullanıyor. Bunun için özel bir modül hazırlamak istiyoruz. Eklerde konu ile ilgili küçük bir tasarım yapıldı. Konu hakkında görüşmek isterim. Tel : 0554 884 25 33

    €720 (Avg Bid)
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    ...providing top-notch customer service and sales solutions in the healthcare industry. We prioritize excellence in every interaction and strive to deliver optimal health solutions to our valued customers. Job Description: We are seeking a dynamic and experienced Call Center / Customer Service / Sales Expert to join our team. The ideal candidate will be proficient in utilizing VICIDial and possess extensive experience in voice-based customer service and sales roles. As part of our team, you will play a crucial role in delivering exceptional service to our clients and driving sales conversion initiatives forward. Responsibilities: Utilize VICI Dialer efficiently to manage and handle customer interactions effectively. Provide exemplary customer service and sales support to client...

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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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    I 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. - Effi...

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    Hello, I'm seeking individuals to join a call center management team with two distinct roles to fill. Candidates should possess basic proficiency in Vicidial & Magnus Billing, alongside fundamental customer support skills. Role 1: Customer Support involves order processing and fulfillment using Vicidial & Magnus billing systems. Role 2: Involves sourcing a consistent supply of inbound DID Numbers for the United Kingdom, United States, Australia, New Zealand, and Finland. Weekly pay ranges from 7000 to 10500 INR based on availability. If interested, please reach out via direct message

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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 need a professional to assist with the installation and configuration of the latest version of Vicidial on a Linux operating system. The key tasks include: - Vicidial software installation: Ensure a smooth and efficient installation process, troubleshooting any complications that may arise. Familiarity with various versions of Vicidial is a plus, notably the latest. - Setting up configuration: Proactively set up and optimize Vicidial configurations to cater to our business needs, ensuring seamless functionality. Ideal candidates should have solid experience with Linux and Vicidial, including its installation and configuration. Your bid will be more favorable if you've previously completed similar projects successfully. I'm happy to provide...

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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.

    €56 (Avg Bid)
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    I need a professional to assist with the installation and configuration of the latest version of Vicidial on a Linux operating system. The key tasks include: - Vicidial software installation: Ensure a smooth and efficient installation process, troubleshooting any complications that may arise. Familiarity with various versions of Vicidial is a plus, notably the latest. - Setting up configuration: Proactively set up and optimize Vicidial configurations to cater to our business needs, ensuring seamless functionality. Ideal candidates should have solid experience with Linux and Vicidial, including its installation and configuration. Your bid will be more favorable if you've previously completed similar projects successfully. I'm happy to provide...

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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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    I am currently experiencing an issue with the FreePBX phone system that I use, specifically, I'm unable to consistently receive calls on all my soft virtual extensions or to my welcome greeting. Voicemail functionality is also part of this setup. Here's what I need: - Evaluation and Diagnosis: Investigate why call reception works only some of the time across all extensions. - Troubleshooting and Repair: Identify the problem and implement a fix to ensure consistent and reliable call reception. Ideal Skills Required: - Extensive experience with FreePBX Phone Systems, particularly concerning soft virtual extensions and voicemail functionalities. - Strong understanding of VoIP systems, and hands-on experience in troubleshooting call reception issues. It is cr...

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    Acil
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    I need assistance setting up a new SIP Trunk on my FreePBX 14 system. Here's what you will need: • Experience with FreePBX 14 • Familiarity with the setup of SIP Trunks • Knowledge of goip local voice gateway Tasks to be accomplished: • Set up a new SIP Trunk on my FreePBX 14 • Ensure the setup is correctly done using my service provider, goip local voice gateway. Only those with the prescribed skills and experience should bid, as I need this done quickly and accurately. Thanks!

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    As fast as possible, I need a professional to assist in setting up FreePBX. Key tasks essential for this project will include: - Installation and configuration - Call routing and extensions - Integration of FreePBX with a CRM system that is yet to be determined In addition, assistance is required for setting up VoIP softphone. Familiarity with popular CRM platforms like Salesforce, HubSpot, Zoho CRM is an asset, as one of them may be chosen for integration. Absolute proficiency in setup, call routing, extensions and integration with CRM systems is a must for this job. I am looking for a fast turnaround on this project, so previous experience achieving quick deployment is highly recommended. It is also necessary to be able to advise our mobile developer if he has questi...

    €160 (Avg Bid)
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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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    ...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.

    €16 / hr (Avg Bid)
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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 project, I encourage...

    €6868 (Avg Bid)
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    ...looking for a skilled freelancer to establish and configure a FreePBX telephone system with the following features: - Call Recording: Implement reliable call recording for quality assurance and training purposes. - Voicemail: Set up voicemail boxes that are easy to access and manage. - Call Forwarding: Enable seamless call forwarding to ensure no call goes unanswered. I am anticipating fewer than 10 extensions for my organization at this moment. Key Skills: - Extensive experience with FreePBX systems and VOIP technology. - Capability to deliver a secure and user-friendly setup. - Strong understanding of potential integrations (CRM or ERP) is a plus though not immediately required. Please highlight your experience with FreePBX in your proposal. I am looking for s...

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    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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    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, potential co...

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    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 p...

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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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    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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    I'm experiencing issues with the current function of Vicidial Survey's human detection. Specifically, we're facing: - False positives - False negatives - Inaccurate detection These issues are largely present when we call English speakers based in the United Kingdom. Consequently, I need enhancements specifically for the detection of UK English. Although perfection is not the aim, the updated system should be capable of detecting English words accurately, such as simple greetings like "Hello". Skills and Experience Required: - Proficient in Vicidial Survey's software - Excellent understanding of UK English, particularly accents - Prior experience with improving speech detection accuracy

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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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    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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    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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    As a Part-Time Vicidial Developer, you will be integral to the continuous improvement and maintenance of our Vicidial system, ensuring optimal performance to meet our organization's unique requirements. This role is perfect for a committed developer with a passion for telephony solutions and a solid background in Vicidial development, particularly in GoHighLevel integration. Responsibilities: Customization and Configuration: Tailor, configure, and uphold the Vicidial system to align seamlessly with organizational needs. Collaboration: Work collaboratively with cross-functional teams to comprehend business requirements and implement Vicidial solutions accordingly. Troubleshooting: Identify and resolve issues related to Vicidial system perfo...

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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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    I am seeking a skilled developer who can build an Asterisk solution on Amazon Web Services (AWS). The system should be proficient in handling a load capacity of 10-50 concurrent calls, ensuring efficient performance and high availability, thereby exceeding the traditional communications infrastructure. Key Features: - Providing VoIP services - Enabling Conference Calling - Incorporating Call Recording Preferred Skill Set: The ideal freelancer for this task should be experienced with not only telephony engineering, but also cloud environment configuration. Knowledge in Asterisk operation, VoIP, and AWS platform are absolutely essential. Proven experience in similar projects is highly appreciated. The goal is a robust and scalable Asterisk environment that leverages th...

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