Freeswitch asterisk callback jobs

...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 ...

...(The cost of the work is $300, divided into two milestones 150/150): 1. Advertising on Facebook 2. Advertising on Facebook messenger 3. Instagram Advertising 4. Advertising on Diarect Instagram 5. Advertising on Vatsam 6. Advertising for retargeting + pixel settings 7. Advertising for A/B test SEO website optimization + website conversion optimization (The cost of the work is $200): 1. On-site callback marketing plugin 2. Marketing plugin for pop-ups on the site 3. Marketing plugin for online hours with automatic responses (communicating bot). 4. Online chat marketing plugin with several channels for communication from 3 to 10 possible. 5. Marketing plugin that allows you to identify some contacts of some site visitors 6. No meta description was found for the page. 7. No keyword...

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 seeking a highly skilled freelancer with intensive experience in FusionPBX and Freeswitch. The main objective of this project is the creation of a precise dial plan for FusionPBX. The developed dial plan should effectively handle active calls - holding them temporarily during paging calls. Key tasks include: * Dial plan must allow higher priority calls to override held calls * Acceptance of priority levels for different callers as input parameters The ideal candidate should possess these skills and experience: * Proven experience in FusionPBX and Freeswitch * Knowledge of implementing dial plans * Understanding of call flows and priority handling. Your task will revolve around ensuring a smooth call flow process during the paging, which detailed understanding of ...

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 currently dealing with an issue in my iOS React Native application. The change mode callback isn't triggering properly. I am looking for a seasoned React Native developer to step in and add additional functionality to fix this issue. Main requirements: - Excellent knowledge in React Native specifically for iOS - Proven experience in debugging and enhancing callback functions - History of adding new features to existing code You can expect to: - Diagnose the issue where the change mode callback isn't triggering properly - Add additional functionality to the callback, ensuring its functionality - Enhance the overall performance of the change mode callback function. Your strong understanding of callbacks in React Native will be instrum...

...HTML to set up a callback URL for PhonePe integration in my Android application. The frontend integration has already been completed; however, I require assistance with configuring the backend to handle PhonePe transaction callbacks effectively. Requirements: Expertise in Laravel: The developer should have strong proficiency in Laravel framework and HTMl to seamlessly integrate the PhonePe callback URL into the existing backend infrastructure. PhonePe Integration Experience: Prior experience in integrating PhonePe payment gateway or similar third-party payment systems is highly preferred. Familiarity with handling transaction callbacks and response validation is essential. API Integration Skills: The developer should be well-versed in working with APIs and implementing ...

I am in need of an expert OpenSIPS VOIP Developer who can help create a robust system with high quality call routing. The main feature's to be implemented are NAT Handling in the Cloud, Microsoft Direct Routing, and Mid Registrar. The system should be designed to interact efficiently with PBX systems (FreeSWITCH) and OpenSIPS. Required Skills: - Extensive experience with OpenSIPS VOIP system development - Strong understanding of direct call routing - Proficiency in integrating VOIP systems with PBX and OpenSIPS - Familiarity with general telecoms concepts and technologies. The ideal freelancer should have worked on similar projects and be ready to showcase relevant examples in their proposal. If this project goes well, there could be more ongoing work. I look forward to disc...

I'm currently dealing with an issue in my iOS React Native application. The change mode callback isn't triggering properly. I am looking for a seasoned React Native developer to step in and add additional functionality to fix this issue. Main requirements: - Excellent knowledge in React Native specifically for iOS - Proven experience in debugging and enhancing callback functions - History of adding new features to existing code You can expect to: - Diagnose the issue where the change mode callback isn't triggering properly - Add additional functionality to the callback, ensuring its functionality - Enhance the overall performance of the change mode callback function. Your strong understanding of callbacks in React Native will be instrum...

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 ..

...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've a custom built application t... My payment gateway is Worldpay for which I've set a callback for the settlement process. However, the issue lies in the processing of the callback response - the current processing is incorrect and needs to be fixed. Key Tasks & Skills: - Debugging: Diagnosing the incorrect processing of the callback response. - API Integration: Expertise in integrating Worldpay's settlement callback correctly. - Node.js: Proficiency in node.js is essential to understand and troubleshoot the existing code. The ideal candidate for this project should have a strong background in node.js, previous experience in Worldpay integration, and a solid understanding of API callbacks. You should be able to quickly identify and resolv...

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 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...

In this project, our colour prediction game, constructed using Node.js, is functioning well aside from a small yet significant issue. We're looking for someone proficient in Node.js to help us debug a critical problem in our callback system. The problem is specifically as follows: - After a successful money recharge in the game, the recharged amount is not added to the user's wallet. - The expected behaviour is that upon recharging, the callback should trigger and the user's in-game wallet should reflect the correct total balance. - Unfortunately, the callback is not triggering as expected, creating a disconnection between the money recharge process and the user's wallet update. Key skills required for this project are: - A strong understanding ...

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.

I'm in need of a developer who can proficiently integrate a callback URL and API to send data in real time. Key Responsibilities: - Integration of a callback URL and API for real-time data transmission The ideal candidate should have: - Proven experience in API integration - Proficiency in real-time data handling - Strong understanding of callback URL implementation - Familiarity with the APIs commonly used for the integration of data The project does not require specific payment gateway integration, only the seamless connection of the API and callback URL to ensure the real-time data transmission. Please, only apply if you have the necessary skills and experience as indicated above.

My game is built using unity build to webgl to run on tele. I need to deploy a new token on jetton, create a smart contract so users can convert this coin token into gold used in the game and vice versa. To do it, you need to catch the callback when the user deposits and withdraws events are successful and shoot it to my game server api. I need it for a maximum of 2 days, you must be experienced in working with ton - jetton, and possibly unity .

I'm in need of a skilled PHP developer to help writing Phonepe payment gateway callback URL in codeigniter PHP environment. Key Tasks: - Implementing the Phonepe payment gateway callback URL in PHP to retrieve payment status Experience: - Previous experience with payment gateway integrations - Proficient in PHP programming - Strong understanding of web services and API integrations Please note that the Phonepe payment gateway account is already configured, so this project is primarily about implementing the callback URL.

I'm in need of a detail-oriented cold calling agent with expertise in real estate. Your focus will be to reach out to leads and successfully book appointments. Key responsibilities: - Call from a provided list of real estate leads - Confidently emphasize our company's background and reputation ...potential leads Ideal skills and experience: - Prior experience in cold calling and real estate is highly desired - Exceptional verbal communication skills - Knowledgeable about our company to accurately present our reputation - Proficient in maintaining call logs and tracking call outcomes. Your success will be judged on the number of appointments you can book, not just expressed interests or callback requests. Let's chat if you have a keen understanding of customer needs...

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

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

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

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...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'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...

I'm looking for an OpenSIPS expert with comprehensive knowledge in call routing. currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have iss... currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have issued when A call B, and B ring 180, after that B refer call to C, call not reach to C - I'm follow opensips Docs we have to config uac_redirect but i try with no luck 2. We use media server is Freeswitch, how can i pass a variable to Freeswitch, i try add header follow format: X-Variable, sip_h_X-Variable but in luascript of FS...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...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...

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.

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...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...

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...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 only w...

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

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

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

We are looking for consultants - Consultant preferred from India. 1) Consultant - SRP Experience in workin...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 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