Huawei asterisk freepbx punët

Pershendetje Ardit, keni eksperience me Suite crm apo Vtiger crm? kam nevoje pernje konfigurim ne lidhje me crm dhe asterisk

...each agent individually: - Other agents cannot see each other's campaigns. 4. Ability for the main agent to assign one campaign to all of their agents. 5. Exporting statistics by agents: - Export a text (or any other) file with answered/unanswered numbers specifically for that agent. - Export statistics for a specific campaign or overall for the day. 6. Integration with Magnus Billing based on Asterisk 13: - Login credentials for the auto-dialing module correspond to Magnus Billing users. 7. It will be web-based. 8. Easy connection to another Magnus Billing server and transferring the module to another server. 9. Ability to manage AutoDialer users Create a new user, delete an old one, disable a user Please provide a brief overview of your relevant experience and how you...

...As we fire up our grills and make vegetable salads for our summer barbecues, we know that we can safely consume cured meats and veggies as long as we maintain moderation. The preservatives, both natural and synthetic, help keep our foods safe from harmful pathogens so we can enjoy our meals without worry. The food label will state that there are “no nitrates or nitrites added,” but an asterisk will often lead to a fine-print addendum with the clarification, “except those naturally occurring in celery juice powder,” sea salt or a vegetable juice. As a result some “natural” or “organic” roast beef and turkey breast, or other products cured with sea salt, evaporated cane juice, potato starch, or natural flavorings or seasoning...

I am in need of a simple, yet powerful tool that will enable me to capture and analyze the IVR menu options of my clients' toll free numbers. This tool will serve the ...Currently one of my team member is dialing the client's phone number, listening to the IVR options and manually capturing the details as follows *Level 1* Press 1 for English Press 2 for Tamil Press 3 for Hindi *Level 2* Press 1 if you are an existing client Press 2 if you are calling us for the first time *Level 3* Press 1 for sales Press 2 for Accounts Press 3 for service We already have a asterisk dialler in place and have the capability to dial phone numbers and also record the conversation but we don't have the ability to listen to the IVR options, dial each option to drill down to each ...

I'm in search of an experienced Radius Server Developer who can take on a comprehensive project. create a radius server that can communicate with OLT, ROUTERS AND VM on HUAWEI, CISCO, FIBERHOME, VSOLUTION, MIKROTIK, UBIQUIT and others through PPPOE, VLAN, IPOE, HOTSPOT ETC

I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

I'm in need of a SIP and Asterisk expert to help troubleshoot issues with a SIP interconnect between two companies, focusing purely on voice communication. Key responsibilities include: - Identifying issues within the existing setup - Repairing and configuring the SIP and Asterisk system Ideal skills and experience include: - Proven experience with SIP and Asterisk systems - Strong troubleshooting abilities - Understanding of VoIP technologies - Excellent communication skills

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

I'm in need of a knowledgeable and experienced Asterisk expert for the implementation in our company We need an Asterisk specialist to help us set up a carousel of different numbers and provider accounts. The essence of the task, when calling, automatically should be substituted random number from our list. Details and scheme will be discussed individually

I'm in need of an experienced PBX / VOIP / SBC engineer who can address several issues with our customers phone systems system. The primary goal of this project is to troubleshoot and maintain current systems. Key Responsibilities: - Troubleshoot ongoing call quality issues - Investigate and resolve call drops - Address any connectivity problems Skills Required: - Extensive experience with Asterisk and other PBX systems - In-depth knowledge of VOIP network architecture - Proven track record in troubleshooting and maintenance Please only apply if you have a strong background in VOIP technologies and are confident in your ability to resolve issues.

I'm seeking a proficient Asterisk developer who can build a custom PBX system with Issabela or something comparable. The PBX system requirements are as follows: - Less than 10 extensions. - Features such as call recording, auto-attendant, and call routing. - Compatible with VoIP phone lines. Ideal candidate should possess a deep expertise in Asterisk and a good understanding of VoIP technology. Knowledge of Issabela or similar software is an added advantage. Let's connect if you can guarantee a seamless and efficient communication system. For more information, sent the requerimients - The calls received would be answered by several receptionists who would log in/unlog in with their landline to receive the call. If there are more calls than receptionists, a message w...

As a sysadmin developer, I'm in need of an asterisk specialist to build a Docker Compose script or a bash script for an interactive vocal server. This project is multifaceted, carrying out outbound calls and saving responses in a database. Key responsibilities are: * Creation of an Interactive voice response (IVR) system. * Outbound calling function connected with my API for automated scheduling of phone calls. * MySQL database integration to securely store the recorded responses. For this assignment, it would be ideal if you have proficiency in using Asterisk, Docker Compose, API integration along with comprehensive database management skills It would be a cherry on top if you have prior experience constructing interactive vocal servers. Let's connect to disc...

I'm looking for a professional who can install Asterisk PBX to facilitate a call routing system. Key Requirements: - Asterisk PBX will function primarily as a call router, modifying incoming caller ID's to the outgoing trunk DIDs. - The system should handle both incoming and outgoing calls efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice tran...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..

I'm looking for someone to help me set up a SIP trunk on my FreePBX phone system. Key Requirements: - Configuration of SIP trunking: I need you to assist in the setup of a SIP trunk on my FreePBX system. Your expertise in this area is crucial. - Connecting to a specific SIP trunk provider: I have a specific SIP trunk provider in mind, and I need you to ensure that the FreePBX system is correctly connected to this provider. Ideal Skills and Experience: - Proven experience with FreePBX phone systems and SIP trunking: I'm looking for someone who has successfully configured SIP trunking on FreePBX systems before. Please provide examples of similar work. - Familiarity with various SIP trunking providers: Your knowledge of different SIP trunking ...

...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

I need a talented developer to create a game app similar to Codenames It should be accessible via iOS, Android, Huawei, and Desktop platforms. The key feature is the support for online multi-player functionality, making the gaming experience interactive and competitive. The design must be professional. Please have experience in: - Multi-platform app development. - Online multi-player functionality. - Creating professional designs. The application must be used Reactjs Nodejs (expressjs) mongodb Additional chat features with text and voice for each room. Project duration is 50 days . Your bid should include your strategy for implementing these requirements.

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.

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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 am looking for a competent professional with experience in Freepbx and Avaya IP integration. The primary objectives of the project are call routing, call forwarding, and establishing a unified voicemail. What I am looking for: - Experience with Freepbx and Avaya IP office 500 V2 systems - Proven track record with telecommunication technology - Capability to integrate functionalities even with gaps in specific call routing or forwarding options Please note that while some specifics on the call routing and forwarding options are not outlined, I expect freelancers to provide suggestions based on their experience and expertise. I look forward to working with an individual who can make these systems work smoothly together.

...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...

I'm in need of a highly qualified and certified network and security engineer. Ideally, you should have certifications from Cisco, F5, Symantec, HP, HPE, Dell, Huawei, Avaya, Fortinet, and Juniper. Key Responsibilities and Tasks: - Network design and configuration - Firewall management - Network troubleshooting - Telecommunication system setup - Network security audits - Hardware installation and maintenance - System integration - Vendor selection and management - Providing ongoing support Your background should demonstrate extensive experience with all the above functions. I'm particularly interested in freelancers who have successfully handled similar tasks in the past. Your success will hinge on your ability to improve our overall network and security infrastructure. ...

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...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

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.

...developed for delivering OTP messages through a web interface. Short codes will need to be delivered through web panel to customers. Text message delivery for multiple services used in accounts creation like facebook, instagram, gmail, whatsapp and other services.. This means we will be using physical sim cards, NOT virtual numbers. Web backend will need to communicate with physical hardware: (Moden huawei e303, 3131 FOR EXAMPLE) The system has to be the same as sms-activate, sms hub, sms pva, please if you have this experience I am searching for coders experienced is USSD/AT commands. Heavy experience with GSM/CDMA cellular networks and sms gateway integration. Look at the websites I've posted above, put together a compatible hardware list for the web frontend&backen...

Dados complementares para gravação das 130 frases: -> duração de gravação em torno de 20 minutos -> Assine apenas seu nome e ultimo sobrenome -> Gravação em telefones celulares de última geração, com no maximo 3 anos de idade: Apple, Samsung e Huawei por exemplo. -> Sobre o Ambiente de gravação: muito silencioso, sem nenhum eco (o som pode ser absorvido com a adição de edredons e travesseiros) -> É necessário que haja uma pausa (um segundo mais ou menos) entre a Alexa e a palavra de comando -> A velocidade geral de fala é um pouco mais rápida -> A distância entre o microfone a sua boca deve ser cerca de 20 cm -> Volume de...

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 a FreePBX expert to help configure the Sticky Agent feature on my system. Key Requirements: - Configure Sticky Agent - Route call to the same agent who answered previously. - If that agent are unavailable, then route to his VM. Call Flow: - Inbound Route - Queue - Static Agent Ideal Skills and Experience: - Expertise in FreePBX, specifically version - Only apply if you have done this previously. My server is live, hence no experimenting. Budget $50

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 need an expert who can modify my wife's Huawei Chinese phone to be fully compatible with American apps and servers. This is crucial as the phone was an expensive purchase and we can't afford for it to be useless in America. Key Requirements: - Experience in programming Huawei phones - Strong knowledge of the Android operating system - Ability to modify apps and servers to be compatible with the Huawei platform. The successful freelancer should have a track record of similar projects and provide a portfolio or examples of past work. Please do not hesitate to reach out for further information.

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...

Task: Need to send a GSM command string ( USSD or SIM Tool Kit commands ) to request a pre-paid mobile credit send, credit recall, and balance check. These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Us...These commands will be triggered by a web server, and communicate with the web server and log all events. Options: - Create stand alone software that will communicate with the web server. - Use the Diafaan software or similar, and customize with it’s C++ / .Net scripting connector - Use other pre existing software Current Hardware: Windows Machine with remote access Port modem , port Huawei modem 3 SIM cards from L...

I require an experienced Huawei BSC 6900 expert to help with the configuration and optimization of our BSC 6900 system. Key tasks will include: - Focusing on efficient and optimal base station control. - Overseeing resource management to ensure proper utilization and efficiency. - Specifically dealing with ongoing connectivity issues. This project is a fit for you if: - You have prior experience with Huawei BSC 6900 system running R09 - You have resolved connectivity issues in the past successfully. - Your expertise lies in network configuration and optimization. Your goal will be to optimize the BSC 6900 system to ensure enhanced connectivity, resource management and base station control.

...lower in crowded cities. Therefore, we need to change the backhaul connection of our gateways to 4G mobile Internet. There are two approaches to this: 1) using an external 4G modem that has an Ethernet connection (), 2) connect a 4G USB stick to the USB connector of the Kerlink Wirnet Station, like the Huawei K5150 (), or the Huawei K5160 (). As we also have some of the aforementioned 4G USB modems, what we are looking for is a solution for the 2) option, that is the integration of the K5150 or the K5160 to the Kerlink Wirnet OS. The solution should be able to handle both public and private APN-s

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.

Description: We are seeking a skilled developer to create a Visual Plan application for FreePBX 15 or FreePBX17 using Node.js or PHP. The application will be based on the Open Source project of FreePBX 13 available on GitHub (), but the candidate is free to decide whether to start from scratch or use an alternative library such as Draw2D (). The main purpose of this Visual Plan is to visualize and modify the flow of the phone system. Requirements: Proficiency in Node.js or PHP, depending on the chosen application Experience with FreePBX and its configuration Ability to work with graphic libraries like Draw2D Understanding and implementation of functional requirements for the Visual Plan Ability to

I'm facing a critical issue with my FreePBX server. To be specific, I'm unable to make outgoing calls despite not having made any recent changes or updates to the server. Furthermore, while attempting these calls, I encounter a 'No response from server' error message. This problem needs an experienced and knowledgeable freelancer to identify and rectify the issues. Key requirements for this task: • Expertise in FreePBX server maintenance and troubleshooting • Ability to work promptly, as this issue is significantly affecting my operations • Solid understanding of host networking and telecommunication systems • Experience in resolving 'No response from server' errors or related issues. Resolving this issue will restore the com...

Integrated Platform for Website, Control Panel, Content Management, and Mobile Apps (Android, iOS, Huawei) Overview: The project aims to develop a comprehensive platform that includes a website, a control panel, a content management system (CMS), and mobile applications for Android, iOS, and Huawei devices. This integrated platform will provide seamless access to content, management tools, and user interfaces across different devices, ensuring a cohesive user experience. Components: Website: Develop a responsive website with user-friendly interfaces. Implement modern design principles for aesthetic appeal and usability. Include essential features such as user registration, login, search functionality, and navigation menus. Ensure compatibility across various web browsers ...

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