Freeswitch voicexml asteriskprojekty
I need to stream live call audio from Asterisk to a WebSocket and vice versa. The purpose of this is to facilitate AI-based interactions during the calls, specifically for conversation analytics. Key Requirements: - Set up a system to stream live call audio to a WebSocket and back to Asterisk. - Enable AI-based interaction for conversation analytics. - Capture and analyze various data points, including sentiment trends, keyword frequency, and call duration patterns. Ideal Skills: - Experience with Asterisk and WebSocket. - Familiarity with AI and conversation analytics. - Proficient in setting up live audio streaming systems. The system must fully integrate with the AI to provide real-time analysis and responses.
De acuerdo a la información suministrada por el cliente, el trabajo consiste en diagnosticar el origen del problema al momento de marcar desde el terminal Bitrix24 usando el siptrunk integrado con asterisk. Requerimientos: - Acceso a terminal Bitrix24 vía anydesk, con posibilidad de instalar software Wireshark para realizar captura de paquete y emulación del problema. - Acceso a FreePbx. Procedimiento: - Acceso a Freepbx para validar operatividad/configuración de troncal SIP. - Conexión a terminal Bitrix24, se procederá a instalar software Wireshark para captura de paquetes. - Emulación del problema y captura de paquetes. - Análisis de captura y tráfico de paquetes SIP entre el endpoint, PBX y CRM. Costos: - El costo ...
I'm looking for an Asterisk expert to assist with my Ubuntu 24.04 system which has Asterisk 18.25.0 and Freepbx 16 installed. The job involves: - Installing and configuring the chan_dongle driver - Installing PlaySMS and fully integrating it with chan_dongle for basic SMS sending and receiving Access Remotely Via Anydesk to my localpc If you have experience in Asterisk, chan_dongle, and PlaySMS, and can handle this setup efficiently, please get in touch. Ideal candidates will have proven experience with Asterisk and Ubuntu, with references to similar projects. A hands-on approach, attention to detail, and the ability to explain technical concepts in a comprehensible manner will be highly valued.
I need a skilled professional to integrate my own TTS and STT APIs with Asterisk/FreePBX along with ChatGPT API for a complete telephony solution. Key Requirements: - Configure SIP trunk for seamless inbound and outbound calls. - Implement TTS for bot responses using my own APIs. - Use my STT APIs to convert user speech to text in real-time. - Integrate ChatGPT API for generating conversational responses. - Design a call flow where the bot handles calls via API. - Enable real-time, two-way conversation between the bot and the user. Additional Tasks: - Set up call recording and database storage of all conversations. - Ensure transcripts, metadata (caller, timestamps, duration), and audio file paths are saved in a database. Deliverables: - A fully operational system with comprehens...
I'm looking for an expert in Odoo and FreePBX(Asterisk) to automate various processes through integration. With this project, I aim to streamline the following: - Call logging and tracking - Customer support ticketing - Sales follow-ups The integration will specifically involve the following Odoo modules: - CRM - Helpdesk - Sales Ideal candidates should have extensive experience with Odoo, FreePBX(Asterisk), and the aforementioned Odoo modules. They must be able to understand the intricacies of these systems and create a seamless integration that enhances efficiency and productivity. Knowledge of call tracking software and customer support systems would be beneficial.
I'm experiencing a frustrating issue with my JIO SIP trunking setup via FreePBX using Asterisk. While I can successfully make and receive calls, incoming calls consistently drop after 32 seconds. I suspect this may be a networking issue, as it seems like the ACK is not getting received from the provider. My current setup includes: - ER605 router - OC200 controller - OMADA Switches Despite my efforts to troubleshoot by disabling SIP ALG and reconfiguring FreePBX settings, the issue persists. The ideal freelancer for this job should have: - Advanced expertise in networking and VoIP - Previous experience with similar issues - Ability to troubleshoot remotely Please, let me know if you can help fix this issue. Thanks!
...Hugging Face Transformers. Dialogflow (Google), Microsoft Azure AI, or Amazon Lex: These platforms provide pre-built conversational AI frameworks that can handle voice interactions. JavaScript (Node.js): If the AI is expected to interact with other services or APIs for real-time updates (like restaurant menus), Node.js can handle the backend development. Telephony Integration (Twilio, Voximplant, or Asterisk): For managing voice calls and hotlines. Speech-to-Text and Text-to-Speech APIs: For converting spoken Arabic into text and vice versa, APIs like Google Speech-to-Text, Microsoft Azure Speech Services, or Amazon Polly. Arabic NLP Tools: Experience working with Arabic language models, specifically Egyptian dialect, will be critical. Database Management (SQL/NoSQL): The AI may ...
Sticky routing function Our retirement village Asterisk PBX system needs a sticky-routing function. This will direct first-time callers to enter a resident house number, while ensuring every subsequent call from that caller is auto-routed to the same extension. The system should use both the caller's previous CLI call history or the resident's outgoing call history for this routing. Key Requirements: - If a caller has previously interacted with two agents, they should be prompted to enter a house-number on every call. - Blocked CLI number callers should be prompted to enter a house-number on every call. - A reset number sequence will be needed to clear all history for that house number. - The empty data storage will need to be locally initialised with startup historical ...
I'm looking for a VoIP engineer who can implement a two-call method: the first call should ring for no more than half a ring, and the second call should go directly to voicemail. I've already set up this two-call method on FreeSWITCH, but I can't get the second call to not ring at the destination phone. I’d love to discuss how we can fix this. Key Responsibilities: - Implementing Ringless Voicemail on our existing system - Enhancing our customer feedback collection process - Ensuring optimal performance of the VoIP system Ideal Candidate: The ideal freelancer for this project would have extensive experience in VoIP, specifically with Ringless Voicemail. They should also have a deep understanding of customer service enhancement techniques and how to implement...
...Hugging Face Transformers. Dialogflow (Google), Microsoft Azure AI, or Amazon Lex: These platforms provide pre-built conversational AI frameworks that can handle voice interactions. JavaScript (Node.js): If the AI is expected to interact with other services or APIs for real-time updates (like restaurant menus), Node.js can handle the backend development. Telephony Integration (Twilio, Voximplant, or Asterisk): For managing voice calls and hotlines. Speech-to-Text and Text-to-Speech APIs: For converting spoken Arabic into text and vice versa, APIs like Google Speech-to-Text, Microsoft Azure Speech Services, or Amazon Polly. Arabic NLP Tools: Experience working with Arabic language models, specifically Egyptian dialect, will be critical. Database Management (SQL/NoSQL): The AI may ...
I need an Asterisk expert to help me with an issue I'm experiencing. The Accountcode is not being assigned within the Dialplan. Despite having not made any recent changes to the configuration, this problem has arisen. Ideal skills and experience for this job: - Deep understanding of Asterisk and its Dialplan - Previous experience with resolving Accountcode issues - Ability to troubleshoot and diagnose issues effectively - Excellent communication skills for explaining technical issues in simple terms
I'm in urgent need of an Asterisk expert with substantial Kolmisoft MOR experience. This project is for personal use and requires immediate attention. Please include in your application: - Your past experience with Asterisk and Kolmisoft MOR. - Specific examples of troubleshooting tasks you've successfully completed in the past. - Your availability to start and complete the project as soon as possible. Your expertise and prompt response will be greatly appreciated.
I have a GSM Gateway, and I want to connect it to an Asterisk server to dial using MDFT (Multi-Digit Feature Tones). However, the gateway cannot connect to Asterisk and is returning a 401 Unauthorized error. I need assistance to troubleshoot and fully configure the Asterisk server and GSM Gateway for successful connectivity and operation. Tasks: Diagnose and troubleshoot the 401 Unauthorized error between the GSM Gateway and the Asterisk server. Set up and configure the Asterisk server to properly authenticate and connect with the GSM Gateway. Configure MDFT settings for dialing through the GSM Gateway using Asterisk. Test the setup to ensure everything is functioning correctly and calls can be made without errors. Provide a d...
I need to relocate my existing FreeSWITCH setup to OpenSIPS + FreeSWITCH primarily for performance enhancements. Skills and Experience Needed: - Extensive experience with FreeSWITCH and OpenSIPS. - Strong understanding of voice over IP (VoIP) systems. - Proven track record in system migration. - Excellent problem-solving skills to ensure seamless transition with minimal downtime. - Ability to optimize the new setup for enhanced performance. 1 ) install opensips, opensips-cli, opensips-cp 1 h
I'm looking for an expert who can assist with installing and integrating Chan-dongle with Playsms and Asterisk on my Linux system. Currently, I have Playsms and Chan-dongle installed, but need help with Asterisk and the integration process. The end goal is to enable seamless sending and receiving of messages via Chan-dongle through Playsms. Key Requirements: - Expertise in Linux operating systems - Extensive experience with SMS management systems, specifically Playsms - Proficient in VOIP technologies, particularly Asterisk - Prior experience with Chan-dongle - Ability to provide a seamless integration I do not require any customizations in Playsms beyond its standard features. If you believe you have the necessary skills and experience, I would love to hear fro...
I need a freelancer who can install Asterisk and Vicidial on my local Ubuntu Linux system. - The operating system is Ubuntu, so you should be familiar with this Linux distribution. - I need the latest stable versions of Asterisk and Vicidial installed, so you should be able to identify and install these versions. Skills and experience that would be beneficial for this job include: - Proficiency in Linux, particularly Ubuntu. - Previous experience with installing Asterisk and Vicidial. - Ability to identify and install the latest stable versions of software.
...destination's WhatsApp number. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to the SIP backend like CALL SUCCESS(200 OK), BUSY(486 Busy Here), UNAVAILABLE(503 Service Unavailable), etc.... to try other rounds on the other SIP Switch. Functional flow 1) Calls from PBX/sip gateway will be sent to WhatsApp gateway 2) WhatsApp gateway converts the SIP to ...
I'm looking for an expert to install FusionPBX on my Debian server. I have an existing PostgreSQL database server, so no need to set up a new one. However, I do need assistance with ...looking for an expert to install FusionPBX on my Debian server. I have an existing PostgreSQL database server, so no need to set up a new one. However, I do need assistance with the installation and initial setup. Key Requirements: - Installation of FusionPBX on a Debian server - Configuration of custom dial plans - Assistance with installation and initial setup - Set FusionPBX to use our fork of Freeswitch, not it's install instance Ideal Skills: - Extensive experience with FusionPBX - Proficiency in Debian - Experience with PostgreSQL - Ability to configure custom dial plans - Strong tr...
I'm looking for an expert in VoIP PBX setup with Asterisk. The primary purpose of this system is for business communications, and we're using a SIP trunk from Voxtelesys. Key Requirements: - Setting up the system to handle up to 10 concurrent calls. - Configuring Voicemail to email feature. - Provide detailed instructions on how to set up on a vps hosted on Holsinger. Ideal Skills: - Extensive experience with Asterisk and VoIP PBX systems. - Strong understanding of business communication requirements. - Previous work with Voxtelesys SIP trunk is a plus.
I'm looking for an experienced developer to create a simple and minimalistic web softphone using React, that connects to latest LTS version of Asterisk(20.9.3). The softphone should allow for voice calling. Key Features: - Voice Calling: The primary function of this softphone will be to facilitate voice calls. UI/UX: - The design should be simple and minimalistic, focusing on usability and clarity. -We will also be needing to be explained Asterisk configuration how its been setup.
Integrate Asterisk Freepbx system with vtiger. Currently utilizing Twilio for service. Scope: currently need terminal settings configured properly with Twilio for calling. 1. Adjust Twilio settings for proper calling functions in Freepbx with my existing softphone. 2. Integrate FreePBX / Asterisk to integrate with vtiger. Contact me with a reasonable quote. After acceptance I will provide login details, and add extensions to practice prior to production mode. Thank you
I need an expert in softswitch MORX14 and Asterisk to troubleshoot an issue I'm facing. Incoming calls are not going to the IVR, and I need them to play a recorded message. Ideal Skills and Experience: - Extensive experience with softswitch MORX14. - Proficient in troubleshooting Asterisk. - Able to configure Asterisk to redirect calls to a pre-recorded message. - Experience with IVR systems. Please note that the recorded message is a calling card IVR.
I'm looking to set up a robust Call Center using Asterisk PBX. The PBX will be hosted on AWS and fully integrated with a user-friendly, web-based front-end dashboard. Key Requirements: - A comprehensive call center set up using Asterisk PBX - Hosting on AWS using services like Amazon EC2, Amazon S3, and Amazon RDS - Development of a web-based dashboard with features like: * Call monitoring and reporting * User management * Real-time analytics Ideal Skills: - Extensive experience with Asterisk PBX - Proficiency in AWS services, especially EC2, S3, and RDS - Expertise in developing web-based applications - Knowledge in implementing call center solutions - Strong understanding of real-time analytics and reporting systems.
...update the freelancer will have to install, and test all the labs to make sure everything works. It is not about writing, it is about testing Asterisk 22 on Ubuntu 24 and update the labs. The labs are available at github. Key Tasks: - Revise the current user training materials to reflect updates in our training to use Asterisk version 22 and Ubuntu 24. - Create clear, concise PDF guides Content to be Included: - Step-by-step instructions for using the VoIP system - Troubleshooting tips to help users solve common problems independently Ideal Skills and Experience: - Previous experience with Asterisk - Strong instructional design skills - Excellent technical writing abilities - Experience in creating user training materials for school environments
...research surveys and possibly other projects in the future. If possible have the AI bot integrated into our existing servers and work with Asterisk and VICI dial as the interface. The developer can also create another interface if needed. Key Requirements: - The AI bot should support both English and Spanish. - It should be cost-effective, utilizing budget-friendly options from OpenAI. - The bot needs to perform various types of data analytics: predictive analytics, sentiment analysis, and trend identification. - Documentation and Training Ideal Skills: - Expertise in AI development, particularly with voice bots. - Experience with Asterisk and VICI dial. - Proficiency in predictive analytics and sentiment analysis. This project is perfect for you if you ...
I'm looking for an experienced Asterisk and VICIDIAL developer to customize the API for user interface integration, specifically for the dashboard and the call control panel. Key requirements: - Integrate the call control panel with the API to enable smooth call management. - Modify the dashboard for real-time call monitoring via the API. Ideal skills and experience: - Proficiency in Asterisk and VICIDIAL with a strong background in API customization. - Experience with user interface integration. - Capability to enhance the dashboard with real-time call monitoring features.
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Queremos integrar Free BPX a través del SIP Connector que tiene Bitrix24. Hemos obtenido mensajes de error y queremos que alguien nos guíe en el proceso.
I'm seeking a developer to create a patch to be applied to the Asterisk 20 source code that's running PJSIP. This patch will log instances where a phone connecting to the system presents a certificate that doesn't match our self-signed certificate, indicating a failure in mutual TLS (mTLS). The primary purpose of this logging is for asterisk logging. The logs should be in plain text format and should only be recorded into the asterisk Logs. This will help me keep track of mTLS failures for security auditing purposes, without the need to enable asterisk debug. Ideal skills and experience for this job include: - Extensive knowledge of Asterisk and PJSIP - Proficiency in patch development - Experience with mTLS Asterisk PJSIP shou...
Integration of Asterisk PBX into Perfex CRM Hello, We are seeking a skilled freelancer to assist us in integrating Asterisk PBX into our Perfex CRM system. Our goal is to streamline communication processes and enhance our customer relationship management capabilities. Please note that we have some additional requirements to add to the initial project description. Here is the complete list of requirements: 1. Expertise in Asterisk PBX: You should have a strong understanding of Asterisk PBX and its features, such as call routing, voicemail, IVR, call recording, and reporting. 2. Proficiency in Perfex CRM: Familiarity with Perfex CRM or similar CRM systems is essential. You should have experience working with APIs and integrating third-party tools. 3. Integrat...
I'm looking to set up an Asterisk and FreePBX system to enhance our business communication. The system needs to include an Interactive Voice Response (IVR) feature. Key Requirements: - Design and implement a robust Asterisk and FreePBX setup tailored for a business communication system. - Integrate an IVR within the system. - Seamless integration of the IVR with our existing Customer Relationship Management (CRM) software. Ideal Skills: - Extensive experience with Asterisk and FreePBX setup. - Proficiency in IVR system design and implementation. - Strong knowledge in CRM software integration. - Excellent understanding of business communication systems.
I'm looking to set up an Asterisk and FreePBX system to enhance our business communication. The system needs to include an Interactive Voice Response (IVR) feature. Key Requirements: - Design and implement a robust Asterisk and FreePBX setup tailored for a business communication system. - Integrate an IVR within the system. - Seamless integration of the IVR with our existing Customer Relationship Management (CRM) software. Ideal Skills: - Extensive experience with Asterisk and FreePBX setup. - Proficiency in IVR system design and implementation. - Strong knowledge in CRM software integration. - Excellent understanding of business communication systems.
... -- Router and Asterisk Security: Analyze routers that interact with Asterisk systems and ensure that no unauthorized access or data retrieval (such as configurations, user credentials, secure phone numbers lists, or IP addresses) is possible. The possession of such data would indicate a critical security flaw in our system. -- Security Recommendations: Based on the findings, provide a detailed report that includes actionable security recommendations for mitigating risks and improving overall system security. QUALIFICATIONS:: -- Expertise in Kali Linux or similar penetration testing tools (e.g., Metasploit, Burp Suite, Wireshark). -- Proven track record of conducting successful penetration tests and delivering detailed security assessments. -- Experience with Aster...
I'm seeking an expert to set up a FreeSwitch PBX-VOIP system on a Linux VPS for customer support purposes. Key Responsibilities: - Install and configure the FreeSwitch platform on a Linux VPS. - Prepare the server for optimal performance and security. - Implement the feature of voicemail to email. - Inbound & Outbound Calling Rules -Trunks setup Ideal Skills and Experience: - Extensive experience with FreeSwitch and Linux VPS. - Prior work with setting up PBX-VOIP systems. - Knowledgeable in server preparation and configuration. - Understanding of customer support communication needs. - Ability to implement and configure specific VOIP features.
...Overview We are integrating an open-source CRM (currently SuiteCRM) with Asterisk and FreePBX, which is already set up on our local host to manage incoming and outgoing calls effectively. CRM users appear in FreePBX, confirming a successful integration. ### Requirements 1. **Click-to-Dial:** - **Outgoing Call Form:** - Develop a custom script to connect the CRM user interface with Asterisk/FreePBX. - When a phone number is clicked in SuiteCRM, it should trigger a call through Asterisk/FreePBX using a registered IP phone extension. - A form must pop up for agents to add and update call details when making a call. 2. **Screen Pop:** - **Grandstream Phone Integration:** - Configure Asterisk/FreePBX to send call data to the CRM upo...
...features, and modules, ensuring multi-company compliance and security. 5. User-Based Email Integrations Email Management: Integrate with popular email providers to allow users to manage emails directly from the CRM. Enable personalized email templates and automatic logging of communication history linked to relevant leads or clients. 6. VoIP Call Management via Asterisk PBX Integration VoIP Integration: Integrate with Asterisk PBX for inbound and outbound VoIP calls, with logic to route calls based on predefined rules. Dubai Location: Configure SIP for inbound calls, requiring a VPN connection for softphone integration. Route calls to the PBX front desk or transfer to extensions, supporting conference calls while adhering to local regulations. UAE Numbers: Set up a S...
I'm looking for an experienced server administrator to help with my Asterisk Vicidial server. The project involves the installation and setup of a multi-server cluster on CentOS, as well as ongoing maintenance, updates, troubleshooting and support. Key Responsibilities: - Install and set up a multi-server cluster for Vicidial on CentOS - Perform regular maintenance and updates to ensure the server is running optimally - Troubleshoot any issues that arise and provide timely support Ideal Skills and Experience: - Proven experience with Vicidial server setup and administration - Proficiency in CentOS - Experience with multi-server cluster setups - Excellent troubleshooting and problem-solving skills
I'm in need of a full-fledged Freeswitch Multi-Tenant UCaaS solution primarily tailored for business communication. This system should encompass a variety of features including VoIP calling, instant messaging, video conferencing, and more. Key Features: - VoIP calling - Instant messaging - Video Conferencing - API integration - Customer front-end GUI - Mobile and Desktop Apps -CRM itegration* -CCaaS -Call Reporting module *CRM Integration: Most softwares now allow API's or Webhooks, so I want the ability to utilize (write to those APIs) if that makes sense. I current have about 4000 extensions on my current system and some clients are linked to Salesforce, Hubspot, Zoho, SugarCRM. The solution needs to be robust enough to handle multi-tenancy. The ideal candidate shoul...
I am seeking a skilled professional to help integrate Asterisk into my network system for an OTP (One-Time Password) system primarily used for password resets. The OTP should be delivered via SMS. Key Requirements: - Expert knowledge in Asterisk and network systems - Experience with setting up OTP systems - Proficiency in configuring SMS delivery for OTP Ideal Skills: - VoIP technology - Automated systems - Call recording and monitoring systems If you are capable of fulfilling these requirements and possess the necessary skills, I look forward to your proposal.
Integration of Asterisk Vicidial into Perfex CRM Hello, We are seeking a skilled freelancer to assist us in integrating Asterisk Vicidial into our Perfex CRM system. Our goal is to streamline communication processes and enhance our customer relationship management capabilities. Please note that we have some additional requirements to add to the initial project description. Here is the complete list of requirements: 1. Expertise in Asterisk Vicidial: You should have a strong understanding of Asterisk Vicidial and its features, such as call routing, voicemail, IVR, call recording, and reporting. 2. Proficiency in Perfex CRM: Familiarity with Perfex CRM or similar CRM systems is essential. You should have experience working with APIs and integrating third-party ...
I'm looking to implement a system on my Asterisk server that checks a caller's credit based on the number dialed. Key Requirements: When user tries to make a call, send the dialled number to the API, the API will return how many minutes of credit they have for that number. If they have > 1 minute of credit available then allow the call to go ahead, otherwise send caller to a pre-recorded message advising them they have no credit (we will supply the message). During the call if credit falls below 2 minutes play "beep" warning to the caller every 15 seconds, when credit runs out disconnect called party and send caller to pre-recorded message advising they ran out of credit (we will supply the message). The API is a standard RESTful API that works with JSON tha...
I need a developer to create a web-based softphone for me using Asterisk, WebRTC, and React. This softphone should be able to support call recording as a key feature. We regard this as quite an easy project for a developed with experience working with Asterisk, WebRTC. The softphone will be deployed at our website.
I'm looking for an experienced developer to configure a SIP soft on my ViciDial/Asterisk setup. The autodialer should utilize Python and PJSIP, and potentially incorporate existing autodialer code. Key project specifics include: - PJSIP as the preferred autodialer software - Implementing functionalities for outbound calling and playing pre-recorded files - Incorporating Google Text-to-Speech for call content - Setting up an auto dialing mode that pulls data from a database Ideal candidates should have a strong background in Python, PJSIP, and Asterisk, with prior experience in setting up autodialers. Knowledge of Google Text-to-Speech and database integration will be highly advantageous.
Asterisk & WebRTC Installation on Linux
I'm looking for a skilled developer with experience in FreeSWITCH to create a custom phone system for me. The primary functionality needed is the development of a comprehensive phone system. Key features of this phone system will include an automated attendant. Ideal skills for this project include: - Proficiency in FreeSWITCH - Experience in developing VoIP systems - Knowledge of automated attendant systems - Call routing programming skills - Voicemail system development experience Please include examples of similar projects you've worked on in your proposal.
I'm looking for a skilled Freeswitch developer to create a robust conference calling platform. Key Features: - The platform must support more than 100 participants simultaneously - Essential VoIP feature: Call recording Ideal Skills and Experience: - Proven experience with Freeswitch - Expertise in developing VoIP services, particularly conference calling platforms - Knowledge and experience in implementing call recording features - Ability to create scalable and reliable systems Please provide examples of similar projects you've completed in your proposal.
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Busco experto para configurar un server de asterisk y diferentes modulos con mms sms entre otras
Job Title: Core Java Developer Brief: Kirusa, Inc a leading company that provides many Value Added Services and AI based solutions to Telcom Companies in more than 18 countries world wide is looking fo...communication skills. Educational Qualifications: Bachelor’s degree in Computer Science, Information Technology, or a related field. Preferred Qualifications: Experience in using cloud based deployment solutions Experience with RESTful APIs and web services. Knowledge of front-end technologies is a plus. Familiarity with Agile development methodologies. Experience with VoiceXML based application development. Benefits: Competitive Compensation Professional development opportunities Flexible working hours Friendly and supportive wo...
I'm looking for a skilled developer to create a softphone for me using Asterisk, WebRTC, and React. The softphone should support the following: - Functionality: The softphone must be capable of handling voice calls. - User Authentication: The softphone does not require any form of user authentication. - Platform Compatibility: The softphone should be designed to work exclusively on desktop platforms. Ideal skills for this project include: - Extensive knowledge and experience with Asterisk, WebRTC, and React. - Proven track record of developing desktop softphones. - Ability to design a user-friendly, functional, and efficient softphone. - Experience in creating software with no user authentication requirements. Please submit your proposal if you meet these criteri...