Asterisk callback openwrt işler

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

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!

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.

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

Selam, bir 4G Router projemiz var. Bu proje kapsamında OpenWrt kullanımını planlıyoruz. Bu konuda bizlere yardımcı olabilir misiniz? Bu projeyi sizlerle nasıl görüşebiliriz?

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.

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

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

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

...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...

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...

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...

...details via a MySQL queue, and communicate transaction results to a specified callback address. Key Responsibilities: System Design: Design a robust architecture that can handle trade data input, processing, and output efficiently with high reliability. API Integration: Integrate with various decentralized exchanges (DEXes) APIs to execute trades on BASE, Solana, and BSC. Database Management: Implement a MySQL database to queue trade requests and log transaction details. Server Deployment: Configure and deploy the application on a WHM cPanel server ensuring high availability and security. Callback Handling: Implement a system to send trade transaction details back to a user-specified callback address after trade execution. Specific Require...

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.

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...

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 have a website that sells physical items. I am looking to open it up to digital items also and need an IPN to handle the callback. I was unable to get the IPN working though. The button works fine, people can order items, but nothing hits my IPN. The site is and a payment page is at The site is all PHP with mysql db and custom built so no CMS.

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

...apps for our CPEs (Customer Premises Equipment). Reproduce, isolate, and debug complex issues. Define APIs for IPCs (Inter-Process Communication). Work on tools for collecting statistics and metrics from different hardware. What we are looking for: Embedded Linux experience: building, deploying, profiling, and testing. Networking knowledge. Understanding of Wireless Networking. Experience with an OpenWrt-based build system, preferably QSDK. Ability to read and write simple scripts in common languages such as Bash, Python, etc. Deep understanding of C and C++ programming for a Linux runtime environment. Nice to have: Experience with Quectel or other vendor LTE. Experience with DeviceTree. Experience with Wireless Technologies. Experience with Luci. Experience with I2C, GPIO, PWM....

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...

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.

I'm looking for an expert freelancer who can assist me in integrating a slot casino game API into my CodeIgneter website. Key Requirements: - Seamless API integration allowing website users to play the game - Provide necessary support and maintenance for the game API -Write callback function Example code and documentation also available

I am looking for an experienced freelancer to install IPFire on a Banana Pi BPI-R3 and create an easy-to-follow guide. My lack of experience with both IPFire and Banana Pi, combined with challenges in installation, lead me to seek professional help. The ...Create a beginner-friendly installation and configuration guide, detailing each step. Completion Criteria: Successful running of IPFire on Banana Pi BPI-R3. Provision of an easy-to-understand installation guide. Validation of basic firewall functions and a demo for confirmation. Information: To run the Linux IpFire install command, is not my main challenge basically. My main challenge is the OpenWRT compatibility, check of hardware, the check of compatibility of WiFi channels and the installation depending hardware like Dip sw...

...~ Backend: Implementation of payments in the script, adding the ability to activate and deactivate in the admin panel, edit, and change authentication data ~ Backend: Implementation of callback from the payment server to confirm payments and/or another communication protocol confirming the transaction ~ Frontend: Adding a payment button during order checkout 2. Payments ~ Backend: Implementation of payments in the script, adding the ability to activate and deactivate in the admin panel, edit, and change authentication data ~ Backend: Implementation of callback from the payment server to confirm payments and/or another communication protocol confirming the transaction ~ Frontend: Adding a payment button during order checkout Payments (at the end,

I'm looking for an expert freelancer who can assist me in integrating a slot casino game API into my CodeIgneter website. Key Requirements: - Seamless API integration allowing website users to play the game - Provide necessary support and maintenance for the game API -Write callback function Example code and documentation also available

For this project, I require a proficient API integrator. We have web site with CodeIgniter. We need - Integration casino API to our platform ASAP. -Write CALLBACK function. Ideal Skills: - Proficiency in API Integration - Prior experience with CRM systems - Familiarity with casino platforms, specifically slot machine functionality The right professional will not only ensure seamless integration but also enhance the overall user experience on our casino platform. If you have experience in such integrations and a passion for casino gaming technology, then I'd love to see your bid.

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

...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: ...

I require the expertise of a seasoned web developer with a demonstrated knowledge of HTML/CSS, Rust, AWS/Kubertes, and Bash to create specialized web pages. The function of these pages is primarily to update OpenWRT routers and AWS databases. Key requirements include: - Designing and developing the pages according to a specific layout I have - Expertise in HTML/CSS, Rust, AWS/Kubernetes, and Bash is a must - High level proficiency in AWS and Kubernetes deployments - Experience with OpenWRT or other router firmware is highly desirable - Excellent attention to detail and the ability to deliver a user-friendly interface Please respond with interest and availability to open the discussion, if this project aligns with your profile and capabilities. I look forward to worki...

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

I am seeking an experienced telemarketer to set appointments with residential homeowners for solar panel installations. Your target market is widespread; any area is acceptable. Ideally, you would have a strong record of successful appointment setting in the past and a professional phone demeanor. WE w...is not cold calling, only interested people. We provide CRM, Lead, Everything. You need to schedule appointment and connect it to zoom with the sales rep that we have ready for you. Key Responsibilities: - Contact homeowners to introduce solar services - Schedule appointments for solar evaluations - Connect client through zoom app Deliverables: - Detailed report of each conversation - Agreed callback date and time with homeowners -Call recording Looking forward to your prompt...

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.

I'm in need of an experienced OpenWRT developer proficient with RALINK RT5350F and MT7628N Chipsets. Primary Tasks: Require changes to existing code to facilitate dial in options. Require capability to write applications in this processor Specialise in USB interface and Router Development Be able to write and make image for upload. Firmware Customization requires adding new features, particularly in integrating dial-in options. Performance Optimization is also needed to ensure the smooth operation of new and existing features. Ideal Skills and Experience: - Expertise in OpenWRT Firmware - Experience with RALINK RT5350F and MT7628N Chipsets - Proven record of successful Firmware Customization and Performance Optimization projects - Strong backgroun...

...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...

...username: "", password: "", }; ('ready', () => { const ses = ; ({ proxyRules: `http://${proxyConfig.proxy_address}:${}`, proxyBypassRules: "localhost,*.local", }).then(() => { // Code to create BrowserWindow }); }); ('login', (event, webContents, request, authInfo, callback) => { if ((proxyConfig.proxy_address)) { (); callback(, ); } }); Despite this setup, the ('login') event does not get triggered for proxy authentication requests. This method is not preferred due to its manual nature and the fact that it doesn't properly handle the proxy's authentication challenge. Seeking Advice Has anyone successfully implemented

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...

...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...

...gateway log or elsewhere, so we're unsure exactly what the issue is. Normal payments i.e. not repeat payments, are working ok. We'd also like to make some improvements suggested by our merchant provider. - Add a new field to tblclients table, to store a secondary 'token' in WHMCS which will be generated by the payment gateway (which you would retrieve in the response, which is dealt with in the 'callback' code). This secondary token will need to be sent to the payment provider for any recurring payments (using WHMCS capture function), in addition to the primary token stored in the gatewayid field in tblclients. - A HTML Form is generated within the code, some logic needs to be added depending on whether a token is already stored or not i.e. if no t...

...Payments ~ Backend: Implementation of payments in the script, adding the ability to activate and deactivate in the admin panel, edit, and change authentication data ~ Backend: Implementation of callback from the payment server to confirm payments and/or another communication protocol confirming the transaction ~ Frontend: Adding a payment button during order checkout 2. Payments ~ Backend: Implementation of payments in the script, adding the ability to activate and deactivate in the admin panel, edit, and change authentication data ~ Backend: Implementation of callback from the payment server to confirm payments and/or another communication protocol confirming the transaction ~ Frontend: Adding a payment button during order checkout 3. Payments ~ Backend:

...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...

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...

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.....

...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...

I need an expert help to configure the Firewall settings on OpenWrt router. Extensive experience with OpenWrt, namespaces and iptables configurations is preferable. Using standard OpenWrt functionality, I need provide the following: - The WLAN0 and WLAN1 interfaces must connect to WiFi networks that are connected to the Internet - The WLAN2 and WLAN3 interfaces are used to create 2x WiFi networks ssid:"NET2" & ssid:"NET3". - The ssid:"NET2" network (WLAN2) should provide Internet access for clients only through WLAN1 connection. - The ssid:"NET3" network (WLAN3) should provide Internet access and access to Wireguard network for clients only through WG & WLAN0 connection - The OpenWrt router itself must use the WL...

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...

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 ...

dalam proyek ini saya ingin memindahkan script dari firmware reyre openwrt ke firmware redmi openwrt... di dalam firmware reyre ada fitur cpu temp dan cpu usage dari Cezary Jackiewicz, script ini mau saya pindahkan ke firmware openwrt saya... lebih diutamakan freelancer yg bs remote dan paham akan html dan javascript, biar bisa edit langsung filesnya sama mau edit 3 html di interface openwrt - ngilangin teks status di menu overview - ngedit 2 hyperlink di menu uutama