Asterisk pbx ivrmunkák
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 in need of an expert who can help install and set up an Alcatel PBX system for a medium-sized enterprise. The project also involves integrating this PBX system with VoIP services. This will be remote project, our team will give you access from the site. Key Responsibilities: - Configuration of Alcatel PBX system OXO408 - Integration of the PBX system with VoIP services along with call recording. - Integration with existing Cisco IP Telephony Ideal skills for this job include: - Profound understanding of Alcatel PBX systems. - Experience with VoIP services integration. - Prior work on medium-sized enterprise deployments. Your expertise in these areas will help us to establish a reliable communication system for our business, ensuring seamless...
I'm looking for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom VoIP libraries to make IVR calls. Here's a brief on what I need: - IVR Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integration: You'll need to integrate the bot with an existing SIP server. Ideal Freelancer: - Proficient in Python, especially in developing Telegram bots. - Experience with VoIP libraries, particularly in the context of IVR calls. - Familiar with SIP server integration. - Strong understanding of DTMF technology. If you're confident in your ability to bring this project to life, let's connect.
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.
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
...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...
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...
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
...with Nice Incontact IVR system to improve my current setup. - Simplify Call Flows: We currently have 10-20 call flows which need to be simplified to enhance customer experience and boost efficiency. - Voice Recognition Improvement: The system's voice recognition capabilities need upgrading to ensure a seamless communication process for customers calling in. - Application Integration: The final task would involve the integration of the IVR system with other applications to optimize functionality. The ideal candidate should have proven experience with Nice Incontact, call flow design, voice recognition technology as well as system integration. Understanding of customer service operations will also be an added advantage. Your job will be to streamline and optimize ...
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.
Good day, everyone. We are seeking a VoIP system developer to create the following features from scratch for our system: We are VoIP ...sign-up option. 2. The portal should allow customers to operate on a pay-as-you-go system, enabling them to recharge or pay their bills directly from the portal using various payment methods such as cards, bank transfers, or cryptocurrencies. 3. We aim to integrate a dialer system within our customer portal, allowing clients to create webphones for their agents/users and utilize other dialer or PBX features. We are open to demonstrating our current working system to provide a better understanding of our requirements. We are seeking a scalable solution, and if anyone has innovative ideas, we are more than willing to explore and collaborate on thos...
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
Requiero ayuda con unas horas de soporte para aclarar algunas dudas sobre una integracion entre una central Grandstream y Opera PMS.
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
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...
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 urgently looking for a skilled professional to quickly handle an IVR-related task for me: - Make IVR calls to 120 different cell phone numbers delivering specific information. Experience in both script writing and call routing configuration is appreciated, although not mandatory. The swift initiation and completion of calls is paramount to this project. Therefore, the ideal freelancer will demonstrate adeptness in utilizing any form of IVR system, be it traditional PBX, cloud-based, or hybrid system. Please note that this project is time-sensitive and needs to be started and finished ASAP. Your adaptability and readiness to start immediately will be highly regarded.
looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
My objective is to significantly enhance customer service efficiency and personalize customer interactions through an AI-based IVR system. Key Tasks: - The IVR system should answer frequently asked questions autonomously. - It should intelligently route calls to the corresponding department based on customer input. - Gathering customer data for more efficient call routing and enhanced personalization should be a key capability. Integration requirements: - The IVR system must integrate effortlessly with live chat to ensure a cohesive customer service offering. Ideal Freelancer: - Proficient in AI and IVR systems - Experience in implementing live chat integration - Understanding of efficient call routing mechanisms - Experience in developing personalized cust...
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
...experienced freelancer to set up a Voice over Internet Protocol (VOIP) system for a yet to be determined number of lines. As no preference for current phone system was specified, I am open to recommendations and would ideally like someone who has experience with both traditional landlines and hosted VOIP systems as well as virtual PBX systems. Ideal Skills and Experience: - Experience in VOIP system installations - Knowledge of Traditional landlines, Virtual PBX, Hosted VOIP systems - Ability to recommend suitable solutions depending on project needs Please include in your application: - Your past work related to VOIP systems installations - Specific experience you have in relation to this project's needs - A detailed project proposal outlining how you will approac...
...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 an expert in 3CX Pro systems to fully optimize my call flow setup. Key tasks will include configuration of: - IVR - QUEE - QUEE WAIT 2 Minits after - If Closed Play Message and redirect to Voicemail - Or All Users Bussy - Play Message and redirect Voicemail I have atached one Pic.
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'm in need of a skilled voice-over artist to inject warmth and friendliness into my PBX system. Meeting the guidelines below is crucial: - Provide a voice-over in a warm, friendly, and conversational style - The recording should be in English preferable US Englisch and range between 1 to 5 minutes - it's 2 scripts but each for 4 different company names. I'm fine if you cut the names so you do not need to record the scripts each 4 times. - Ideally, the voice-over artist should have a middle-aged vocal range I'm looking forward to hearing your warm and engaging voices.
I'm in need of a talented freelancer for a voice recording project: - Unfortunately, I skipped the questions regarding the intended use for the voice recordings, what information successful freelancers should include in their application, and the preferred language for the voic...voice recordings, what information successful freelancers should include in their application, and the preferred language for the voice recordings. - Even without this information, I expect potential candidates to be adaptable and versatile with their voice talent abilities. - Regardless of the language and the purpose of the voice-over, having previous experience in voice-over, podcasting, or IVR will be advantageous. I look forward to hearing from diverse talents who can cater to multiple voice r...
...seamless call handling through two GSM SIM lines, connected through IP phones and a GSM gateway. Scope of Work: 1. Configure existing hardware, including: * 2 IP phones (astTECS Mini PBX UC100 & Yealink T48S) * 1 GSM gateway (Yeastar TG400) * 1 Computer system loaded with Bicom PBXWare software Version (link unavailable) * 1 Network switch * 1 Internet modem 2. Set up the software to work with the following: * Failover and receive calls through the two GSM SIM lines * Make and receive calls through the IP phones and GSM gateway 3. Set Call Transfer 4. To set and configure the IVR ( Interactive Voice Response (IVR) 5. To set and configure the MOH (Music on Hold) Deliverables: * Successful configuration of the phone system to meet the above requirements * T...
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.
...text base messages. Can you help me build this system and how much is going to be the total cost I donot have any number to use at the moment? Autodailer: I need to do unlimited calling to usa, Canada and UAE, on raw but verified phone numbers. (sort of cold but verified data of 10million contact numbers) Secondly I need an auto dialer which is going to be linked to a dashboard along with an IVR played in three to four accents which doesn't sound robotic. I need numbers atleast 10 for dialing and routes most importantly sometime the numbers are pin pointed as spam so for after every 10k calls the numbers needs to be changed or delisted. Now the challenge: How many numbers can you provide? How many routes do you have? How many ips we need do we need for mass quantit...
I'm seeking an expert...integrating Zoho CRM and 3CX for fewer than 10 users. The key goal of this integration is to set up 3CX telephony within Zoho CRM. I tried but failed and don't know what I'm doing wrong. Key responsibilities: - Calls are to be looked at via the Zoho CRM telephony function and lookup contacts to show who is calling. Looking for: 1. Jump on a call via Zoom 2. Help me read: 3. Via Zoom guide me or provide support on what I'm setting up wrong. 4. Test solution or help me contact support if needed. This project offers the opportunity to bring significant improvements to our CRM and telephony capabilities. I look forward to engaging with seasoned professionals who can efficiently and effectively execute this project.
...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...
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, H...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 questions about PBX or CRM
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 ...
I'm looking for a skilled developer to create an AI Intelligent Virtual Assistant (IVA). It will need to integrate Natural Language Processing and Machine Learning Algorithms, along with Digital Human capabilities. The main functions should include: • Understanding and processing user inputs • Developing responses using machine learning • Implementing computer vision for a digital human interface The AI has to be compatible with Windows and Mac desktop platforms, Mobile platforms like Android and iOS, and also web-based applications. Ideal freelancers should have significant experience in AI, machine learning, natural language processing, and virtual assistant development across different platforms.
...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...
Task details - For ISUP/SS7 (E1 card implementation) along with IVR knowledge in any open source SIP servers like Freeswitch/Yate/Mobicents etc. or working experience in CRBT server. Interested candidates please apply
I am searching for a skilled software developer with a strong background in Asterisk, Dialer, IVR and VOIP technologies. Although I haven't specified particular functionalities, general familiarity with call routing, call recording and interactive voice response (IVR) would be beneficial. The ideal candidate for this job should be proficient in: - Designing, implementing, and maintaining Asterisk software - Developing dialer functionalities, with emphasis on auto dialing, click-to-dial, and predictive dialing - Ensuring system is up-to-date and secure Freelancers who apply should provide any past work, detailing their experience and including project proposals, if any. If you believe that you have the expertise to effectively take on this projec...
We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experie...Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenarios. Intereste...
Necesito ayuda para revisar por que mi central telefonica Issabel no esta grabando las llamadas.
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. - Inc...
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...
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 ...
