Freeswitch voicexml asteriskemplois

I'm looking for a professional with specific experience in bulk calling systems, RTP optimization, and Asterisk configurations. The primary goal of this project is to effectively handle a high volume of calls. Key features of the system include customizable call routing. The main purpose of this routing system is to direct calls based on time zones. Essential Skills: - Expertise in bulk calling systems - Proficiency in RTP optimization - Asterisk configuration experience Desirable Skills: - Experience with time zone based call routing - Knowledge of customizable call routing systems - Skills in designing systems for high call volume handling

I'm looking for a developer to create a custom Android softphone for me. This softphone will primarily be used for voice calls and needs to be compatible with an Asterisk PBX system. Key Requirements: - The softphone should support high-quality voice call capabilities. - It should be fully integrated with the Asterisk PBX system. - Experience with Android development and VoIP technology is a must. - Understanding of Asterisk PBX systems will be a significant advantage. Please note that the softphone does not need to support video calls or messaging at this stage. The focus will be solely on voice call functionality.

This project is a text data analysis project. The goal of the project is to convert a file containing 36,000 book authors into a table of unique “wildcard” search terms ranked by their frequency. A “wildcard” search term is a character string ending in an asterisk. For example, the characters “ins*” would be a wildcard search term. When submitted into a search engine that accepts wildcards, “ins*” would return any word starting with the characters “ins”. For example, “insurance”, “insulation”, “instance”, etc. The project will involve the following main steps: * Parse the set of 36,000 book authors into separate words so that each word is a separate record. *&n...

I'm looking for a seasoned professional who can set up an Asterisk server for me. This server should be capable of recording calls, integrating with ChatGPT for customer service automation, and transcoding call recordings to email them to specific users. Key Requirements: - Set up Asterisk server with call recording capabilities. - Integrate ChatGPT for enhanced customer service automation. - Transcode call recordings and email them to designated users with the recordings attached. Ideal skills for this project include: - Extensive experience with Asterisk server. - Proficiency in integrating AI tools like ChatGPT. - Strong understanding of audio transcoding and email systems. Your expertise will help improve our customer service efficiency. PLEASE NOTE: I W...

...handle the signaling for voice calls. It's crucial to use SSL certificates to enable secure WSS connections. - Access Server via SSH: You will remotely access our server via SSH to carry out the entire setup. The entire process should be recorded as a screen recording, including accessing, setting up, configuring, and testing the server. - SIP Server Integration: The project involves configuring an Asterisk SIP server for call signaling. Ensuring integration between the WebSocket server and SIP to enable communication with Odoo's softphone is essential. - Nginx Reverse Proxy Setup: You will be setting up Nginx as a reverse proxy to forward secure WebSocket traffic (WSS) from clients to the backend server on port 8443. Ideal Skills and Experience: - Extensive experience...

I'm seeking a skilled developer to create an Asterisk AMI-based API for my SIP VoIP system. Key Requirements: - The API will need to interface with multiple third-party systems, including CRM software, billing systems, and softphones. - Proficiency in SIP is a must, as the system must strictly adhere to this VoIP protocol. Ideal Skills: - Extensive experience in Asterisk AMI and API development. - Prior work with SIP and its integration in VoIP systems. - Familiarity with interfacing APIs with CRM software, billing systems, and softphones. - Strong understanding of API design and software development principles. I look forward to receiving your bids.

I'm looking for an expert to configure ClearIP's inline proxy on my Asterisk VPS. The primary purpose of this proxy is to facilitate testing of my hosted VoIP setup. The selected professional will be responsible for: - Setting up the inline proxy on the VPS - Conducting tests on network performance and system integration of the VoIP - Ensuring the proxy is configured for optimal content filtering, performance improvement and security enhancement - Providing a comprehensive report on the tests results and any issues found Ideal candidates should have: - Extensive experience with configuring inline proxies - In-depth understanding of VoIP systems, particularly hosted VoIP - Proven track record in testing network performance and system integrations - Excellent skills in pr...

I am looking for a C programmer who is also familiar with Freeswitch to patch up a bug that freeswitch themselves have not given any attention to. I found what I needed within the source code, it is just a matter of now making it work correctly. This is not anything complicated, should not take more than 2-3 hours at the most from my experience. Key Responsibilities: - Debugging existing C code within the Freeswitch framework - Identifying and resolving functionality bugs impacting call routing Ideal Skills: - Proficient in C programming - Familiar with Freeswitch - Strong debugging skills - Excellent problem-solving abilities

For Asterisk, in php, checking AGI/AMI scripts

Preciso de um app VoIP em react-native para conectar no meu servidor asterisk para comunicação entre os ramais por audio e vídeo

I am experiencing configuration errors with my Asterisk and VoIP setup, specifically with the dial plan. The main issue is incorrect call routing. I need an expert who can troubleshoot this issue, identify the root cause, and implement the necessary fixes. Ideal skills and experience for the job: - Extensive experience with Asterisk and VoIP - Strong understanding of dial plan configuration - Proven track record of resolving call routing issues - Excellent problem-solving skills - Ability to work efficiently under pressure

I'm looking for an expert who can seamlessly integrate Asterisk 18 with Salesforce via a CTI Connector. Key Features: - Enable click-to-dial functionality from within Salesforce - Ensure automatic logging of calls - Configure screen pop-ups on incoming calls - Setup an auto-dialer and voice broadcasting - Enable call transfer functionality to forward calls to other representatives - Integrate voicemail system for missed calls - Implement automatic call recording for quality assurance - Provide detailed call analytics and reporting within Salesforce - Set up an Interactive Voice Response (IVR) system - Enable custom caller ID for different departments or agents - Send real-time notifications for missed calls and voicemails - Create a system for call escalation to higher-level ...

I need assistance with my Asterisk setup. The system is not integrating correctly with my custom billing solution, causing multiple issues. The ideal candidate for this project should have extensive experience with: - Troubleshooting Asterisk - Integration of Asterisk with billing systems - Working with custom billing solutions - Understanding and resolving call quality issues Please only apply if you are capable of diagnosing and fixing the integration issues promptly.

I'm looking for an expert for Vicidial and configure it with GoIP for VoIP call routing. Key Requirements: - GoIP setup for VoIP call routing - Experience with Asterisk and Vicidial Additionally, I need security measures for the SMS Gateway. Specifically, implementing IP whitelisting to ensure only approved IP addresses can send and receive messages. Ideal Skills: - Asterisk PBX configuration - Vicidial setup and management - GoIP configuration - SMS Gateway security measures implementation

Necesito buscar a alguien que pueda instalar webrtc en un hosting y este se conecte con un asterisk para finalmente la web pueda hacer llamdas telfonicas mediante una llamada de telefonia IP. hay documentacion pero se necesita alguien que tenga experiencia.

Hello: I need your help to - Install and configure Freeswitch on my Linux server (Debian) - Ensure the system is set up for optimal performance by using open source libraries Please only apply if you have a proven track record with similar projects.

Configure Asterisk dialplan, so it can be integrated with Python scripts the client already has

I need an expert to integrate my Asterisk IP PBX with WhatsApp for receiving customer support calls via SIP. This project involves both setting up new WhatsApp accounts and integrating with existing ones.

I'm dealing with intermittent connection drops on my Asterisk system, which is running on a cloud-based service. Ideal Skills and Experience: - Proven expertise in diagnosing and resolving Asterisk connectivity issues. - Experience working with cloud-based Asterisk services. - Strong understanding of SIP protocols and VoIP networks. - Excellent problem-solving skills and attention to detail.

I’m working on a project that aims to automate calls, includi...PBX Account: I have a Skyline account with a phone number, username, and password. - Issabel: Using Issabel to manage calls, connect Skyline PBX, and handle SIP configurations. - Python for Call Handling: I need Python to interact with Asterisk, answer calls, and handle STT/TTS and GPT processing. Current Issues 1. Asterisk ARI Compatibility: - I found ari-py, but it’s outdated and works only with Python 2. - Tried asyncari, but faced numerous compatibility issues with anyio and ARI event handling. 2. Goal: I need to connect Python with Asterisk to: - Answer incoming calls and handle events. - Process voice input (STT) and generate responses (GPT)....

I'm in need of an expert who can carry out advanced, custom configurations on my Asterisk SIP server. The job will also involve integrating the server with various other communication platforms. Experience with Asterisk and SIP server configuration is a must, as is proficiency in system integration.

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.

Hola buenas toy buscando un persona a que me enseñe como manejar magnusbilling para manejar el precio de tarifas de mi llamadas Asterisk de preferencia habla español

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...