Ami dial asterisk 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.

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.

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

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

I'm searching for a skilled designer with a keen eye for detail and expertise in creating realistic patch designs. This project involves the creation of a decorative patch, with no specific theme to be incorporated. Here's what I'm looking for: - Mastery in designing realistic patch designs. - Ability to...- Ability to innovate and create from scratch, without dependence on particular themes. - Expertise in capturing the essence of detail for decoration. - A portfolio demonstrating previous work related to my needs. Your main task would be creating a patch for our 100,000 miles, following a realistic style, rather than a minimalistic one. I would like to see a motorcycle throttle with the 100,000 dial across where the rider would hold the grip. Please share your wor...

... I do not know anything about robotics and have never played with it. I just know that what I want done is capable of being done. We are looking for someone that is very familiar with building task that we can bring in to our existing software possibly through a d l l. Our software is already built and performing but it requires a human being to make a turn of the dial or a change of the range. Whoever we hire will be familiar with the process and be able to advise us on solution that will get the job done. The items that we are looking to work with our small test and measurement devices and we basically need the robotic arm to make the then when done, trigger and let our software take over from our software finishes its

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.

THE JOB I'm seeking a skilled designer to adapt an existing logo for my new clothing brand. Please note, the company logo already exist and will be sent before the job starts. The job consist of adapting the logo so it loo...a new font. The name of the company is EKEROE. DELIVERABLES 1. Deliver the logo files 2. Deliver examples how the new logo looks on the clothes (sweatshirt) ABOUT US We are a startup creating a new clothing brand - sporty clothing for office use. The first collection will consist of hoodies, sweatshirts and t-shirts for men and women. Price point higher than H&M but lower than Ralph Laurent, Ami Paris, Stone Island, Comme de Garcons etc. OTHER The job must be delivered in an iterative process where we can see examples and give feedback Payment upo...

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...

...their understanding of this platform through a demo project. Looking for a simple ESP32/Wifi project for color mixing with LED lightbulb. This is a simple demo project to show WiFi connectivity using the ESP32 microporcessor Using ESP32, 64x128 oLed, one button and 3 LEDs, 10K potentiometer, 1602 LCD (non I2C) Dial in R color using potentiometer 0-255 – Press button, show brightness on RED LED Dial in G color using potentiometer 0-255 – Press button, show brightness on GREEN LED Dial in B color using potentiometer 0-255 – Press button, show brightness on BLUE LED Colors will show on individual LEDS after each button press Press button again to “send” color setting the WiFi LED light bulb and show final mixture values on the 1602 LCD ...

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.

...- Providing assistance with CCTV systems - Offering support for access control systems Your job will be to receive inbound phone calls, listen carefully to the user's concern, help diagnose the problem, and provide the simplest and most effective solution. Ideal Skills and Experience: - Technical Support - IT Support - Experience in Security Systems (CCTV, Access Control) The customers will dial a public phone number which will be routed to you; all inbound calls will be routed in a queue-like system. Therefore, being comfortable with communication is important. You will need to know English and ideally some Indian languages. To be successful in this role, you need to be patient, articulate, and a good listener. An understanding of customer service and prior experience...

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

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

** If you are a company I am not interested in you applying. This is for freelance developers. ** I'm looking for a dev to work on a project. I need an Android application created which displays automotive gauges which will display data passed to it via Bluetooth. Those gauges might have a "dial" which needs to move based on data sent to it or it can be just a "text based display" depending on which UI is chosen by the end user. What language it is written in doesnt matter much. It just needs to be able to have user selectable UI's which can be designed by me after you complete the project. If you know how to write this application but are not a graphics designer, thats OK, I have a graphics designer that can work with you on the UI. Require...

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.

...Smiths, Echo and the Bunnymen, etc. These can be live shots and/or promotional. Each pic duration should only be about a second or two. The audio should correspond to each band and feature a 2-3 second sound byte of their biggest songs, i.e. Pic of Oasis/sound byte of chorus of live forever "Maybe I just wanna-", Pic of "Blur"/short clip of Song #2 - "Woo-hoo!" etc etc Include sfx of a radio dial changing the station between each audio clip, as if someone's going from station to station looking for a song.a The final clip should be our logo flickering on/off like a neon sign for a second or two, until it blows up, fading into Austin Powers (attached) saying "Yeah, baby, Yeah!" - fade out. We are also open to any creative ideas/thi...

Hi Ami J., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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 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 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 need a freelancer for a three-hour a day gig focused on telemarketing for lead generation. The services required include: - Calling 100 US based B2B or residential numbers daily. - Submitting daily reports detailing the outcome of each call and the leads generated. The ideal freelancer should preferably be based in the Philippines, have excellent...especially in lead generation. Your work will help expand my client's reach and develop potential client relationships. Your ability to communicate effectively and professionally will be key in nailing this project. Get ready to hit the ground running on this exciting opportunity! This bid is for 100 calls to the USA! You must speak good English and be able to send a voice sample. Must be able to dial USA. Can be full time i...

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 background in networking a...

I'm in need of an expert to configure FusionPBX for 1-5 tenants with Twilio. This includes: - Call Routing functionality - Recorded welcome - Voicemail capabilities An important aspect of this project is customization for each tenant, to include: - Individual Dial Plans - Custom Voicemail setups - Tenant-specific features and settings (something to start with so we can grow packages) - Potential ongoing support as we grow Prior experience with FusionPBX is essential for this job. This project requires a deep understanding of client-focused VoIP solutions. Sound technical skills in FusionPBX platform setup with Twilio, VoIP configurations, and tenant-centric customization are key in this role. The deliverables should not just meet the standard FusionPBX platform requirement...

...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...

Szeretnék létrehozni egy szabadúszókra szakosodott weboldalt, mint az Uppwork, sproutgigs, és szeretnék hozzáadni egy új funkciót, ami a rövidített linkek cseréje.

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...

...contact called would then go to the lowest priority, which would move the first contact one space up in priority, and of course the cycle of prioritizing contacts called would continue in this fashion as calls are made. when a contact in the database is called, the contact's information will show up so that the user can see it. there is a button to hang up the call so that it will automatically dial the next contact in the database to be called according to priority. there is also a pause button so the user can pause the calling process in case they need a break or whatever else they might need it for. there will need to be a button for sales made. when a sale is made a user clicks this. then they can enter the customer's information ( first and last name, billing ad...

...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 en...

...visitors in website , consider the following scope of work: 1. Collection of API from Respective department: To reach our milestone our first job is collection data from various department like, • Ministry of Corporate Affairs [Register Companies data collection] • Goods and Service Tax [GST Register Companies data collection] • Income Tax [PAN Information Search] • Just Dial [All new business listing data collection] • Any other govt sites [all other available database collection] 2. Control Panel for Data Validation and edition or delectation. Need to develop a control panel where all collected data can altered / remove / manual addition before live in page. 3. Customer ID Allocation After daily basis data collection, and before live in...

...design, taking inspiration from the provided logo for color scheme consistency. Content Inclusion: Incorporate the menu items and prices from the images provided. A text file will also be attached with the typed out text from the images to facilitate the process. Additional Content: Include a WiFi ID and Password section for each floor of the restaurant. Add a note that says "Dial 9 on intercom to reach reception." Deliverables: A print-ready restaurant menu design in A4 size. Source files in editable format (e.g., PSD, AI, INDD). Skills Required: Proficiency in graphic design software (e.g., Adobe Photoshop, Illustrator, InDesign). Strong layout and typography skills. Ability to work with brand guidelines and color scheme...

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...

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 ...

Sei un esperto nel filtrare e qualificare lead? Hai una passione per la creazione di contenuti video coinvolgenti? Sei la persona che stiamo cercando!
 Responsabilità Principali:
 Lead Qualification: * Utilizza strumenti come ... 
 Requisiti: * Esperienza comprovata nella qualificazione dei lead e nella creazione di contenuti video. * Eccellenti competenze comunicative e capacità di gestire relazioni con i clienti. * Orientamento ai risultati e capacità di lavorare in team.
 Target di Clienti: * Ristoranti e locali del settore della ristorazione.
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Siamo entusiasti di conoscerti e far crescere insieme il nostro team!

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.

Looking for aws expert, to launch running instance from ami from another account, that account is blocked from creating new instances. Need to set proper security groups proper login credentials and same elastic IP this instance apache ubuntu and had PHP running

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?