Freeswitch voicexml asteriskJobs

I'm experiencing poor call quality on my Asterisk server, which is deployed on a Wide Area Network (WAN). The specific issue seems to be linked to the voice transmission codec, currently set to G.711. Key aspects of the project include: - Investigating and identifying the root cause of the poor call quality - Suggesting and implementing solutions to enhance the call quality - Potentially changing the codec from G.711 to a more suitable one, if necessary Ideal skills and experience for this project include: - Proficiency in Asterisk servers and VoIP technologies - Strong understanding of network configurations and troubleshooting - Knowledge of different voice codecs and their impact on call quality - Experience in improving call quality on WANs Your expertise and expe...

I'm experiencing poor call quality on my Asterisk server, which is deployed on a Wide Area Network (WAN). The specific issue seems to be linked to the voice transmission codec, currently set to G.711. Key aspects of the project include: - Investigating and identifying the root cause of the poor call quality - Suggesting and implementing solutions to enhance the call quality - Potentially changing the codec from G.711 to a more suitable one, if necessary Ideal skills and experience for this project include: - Proficiency in Asterisk servers and VoIP technologies - Strong understanding of network configurations and troubleshooting - Knowledge of different voice codecs and their impact on call quality - Experience in improving call quality on WANs Your expertise and expe...

I'm in need of a skilled professional who can set up SIP trunking for our system. Key Requirements: - Your experience with SIP Trunking, Asterisk, FreePBX, and PBX systems is essential. - Setup a system that enables us to make calls with our assigned caller ID. - Implement an IVR system with the primary goal to enhance customer service. The ideal candidate should have experience with IVR systems and should be able to suggest a simple solution and advice on how to store data which will be integrated with the IVR System, help me create prompts and scripts based on my ideas. Key requirements: - Your experience with SIP Trunking, Asterisk, FreePBX, and PBX systems is essential. - Setup a system that enables us to make calls with our assigned caller ID. - Experience wit...

Hi, We are looking for a reliable software development shop to help us maintain our B2B SaaS VoIP Telecom software system and add on new features. Key Requirements, programming languages in these languages is a must: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line MySQL Mongo DB React Electron GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS Purpose: Our software is a B2B SaaS and will be focused on Telecom, CRM, Data, AI using Open AI API and Claude Sonnet API and Marketing. So, if you have experience in these fields, it would be a plus. Features: Also will need to handle customer data management, call tracking and analytics, marketing automation, reporting, and analytics. This will be a long-term partnership, so I'm keen to find a team that...

De preferat vorbitor de limba romana I'm looking for an expert in Asterisk server configuration to set up call recording and basic transcription. using ,whisper,word2number,websockets,agi script,and other script Key Requirements: - Implement call recording for both incoming and outgoing calls on the server directly. - Ensure the recorded calls are available for playback and download. basic transcription. using whisper Transcription: - Basic transcription of these calls is needed. No specialized terminologies or technical fields are involved. - The transcriptions need to be clear and accurate to ensure easy follow-up. Ideal Skills: - Proven experience with configuring and managing Asterisk servers. - Proficient in setting up server-side call recording. - Familiarity...

I need a skilled Asterisk PBX expert to help me configure my system to connect our Avaya J179 IP Phones and Zopier SIP Soft Phones. Key Requirements: - Connecting and ensuring compatibility of Avaya J179 IP phones and Zopier SIP Soft Phones. - Enabling call forwarding functionality on both phone types. - Setting up the voicemail feature on both phone types. I need this project completed as soon as possible, so speed and efficiency are key. Ideal Skills: - In-depth Asterisk PBX expertise - Experience with Avaya J179 IP Phones and Zopier SIP Soft Phones - Strong understanding of SIP protocols and VoIP networking.

Tengo varios servidores con Asterisk y tengo el inconveniente de que el archivo principal de la base de datos de mysql se vuelve muy pesado. Requiero analizar la posibilidad de crear algun script de mantenimiento que permita limpiar dicho archivo cada cierto tiempo de manera que se eliminen los datos mas viejos y el archivo no crezca desproporcionalmente, para evitar que llegue a colapsar el sistema. El archivo es el que se encuentra en la ruta: /var/lib/mysql/ibdata1 y la idea es que este proceso se pueda correr manual o automaticamente en ventanas de mantenimiento sin que afecte el correcto funcionamiento de la base de datos. El proceso debe ser capaz de eliminar registros viejos de las tablas que utiliza Asterisk, limpiar dicho archivo y verificar que todo quede funcion...

I'm looking for a professional to help with a Speech-to-Text integration project. The details of the platform, language, and primary spoken language are yet to be determined, which will be discussed during the project. Asterisk & SIM7600G-H Speech-to-Text Integration Ideal candidates should have experience in: - Developing speech recognition features - Integrating speech-to-text technology - Being knowledgeable about various programming languages Kindly place bid if you think you are expert with this requirement as we get started

I'm looking for a professional to help with a Speech-to-Text integration project. The details of the platform, language, and primary spoken language are yet to be determined, which will be discussed during the project. Asterisk & SIM7600G-H Speech-to-Text Integration Ideal candidates should have experience in: - Developing speech recognition features - Integrating speech-to-text technology - Being knowledgeable about various programming languages Kindly place bid if you think you are expert with this requirement as we get started

I'm looking for a professional to help with a Speech-to-Text integration project. The details of the platform, language, and primary spoken language are yet to be determined, which will be discussed during the project. Asterisk & SIM7600G-H Speech-to-Text Integration Ideal candidates should have experience in: - Developing speech recognition features - Integrating speech-to-text technology - Being knowledgeable about various programming languages Kindly place bid if you think you are expert with this requirement as we get started

I need a professional who has experience with VoIP, specifically with Asterisk, to carry out this project. Your responsibilities will involve: - Installing Asterisk on Debian 12. - Configuring Asterisk with a SIP provider, namely telnyx, callwithus, or skyetel. - Setting up Asterisk to run a pre-existing Python script. Please note that the specific script was not specified, so I will provide further details upon getting in touch. - Ensuring that this Python script is successfully making calls to landline numbers to check the validity of some cards. The ideal candidate for this job should have previous experience with Asterisk, Debian 12, Python, and SIP configuration. Please submit a bid if you meet these qualifications. Be prepared to provide examples ...

(unfortunately i only havebudget of $111 , due to last freelancer unable to complete and is already in balance) Freeswitch is already setup and running via whm root terminal, just need someone to install fusionpbx without interfering with currrent cpanel that has wordpress front end site on it. after completion will display via subdomain 1. freeswitch is running, no need help for this 2. just install fusionpbx for me on a sub domain, the sub domain is actually set to 7.4 3. cannot interfere with current installation of cpanel

(Dont overbid, should only take 30 min for experienced freelancer) Freeswitch is already setup and running via whm root terminal, just need someone to install fusionpbx without interfering with currrent cpanel that has wordpress front end site on it. after completion will display via subdomain 1. freeswitch is running, no need help for this 2. just install fusionpbx for me, pgsql database is already installed as well for fusionpbx i believe 3. cannot interfere with current installation of cpanel

I'm seeking a skilled VoIP engineer who can boost the concurrent call capacity of our existing Asterisk server using OpenSIPS. Specifications: - The primary goal is to enhance the concurrent call capacity of our Asterisk server, making it more scalable and efficient. Key Responsibilities: - Implementation of an OpenSIPS SIP proxy that can effectively distribute SIP traffic across multiple Asterisk servers. - Ensure the SIP proxy is configured to handle a high volume of calls simultaneously. Ideal Skills: - Strong experience with VoIP technologies, particularly Asterisk and OpenSIPS. - Proven track record of optimizing call capacity on telephony platforms. - Familiarity with architecture design for scalable and redundant systems. This project offers a gr...

I'm seeking a skilled VoIP engineer who can boost the concurrent call capacity of our existing Asterisk server using OpenSIPS. Specifications: - The primary goal is to enhance the concurrent call capacity of our Asterisk server, making it more scalable and efficient. Key Responsibilities: - Implementation of an OpenSIPS SIP proxy that can effectively distribute SIP traffic across multiple Asterisk servers. - Ensure the SIP proxy is configured to handle a high volume of calls simultaneously. Ideal Skills: - Strong experience with VoIP technologies, particularly Asterisk and OpenSIPS. - Proven track record of optimizing call capacity on telephony platforms. - Familiarity with architecture design for scalable and redundant systems. This project offers a gr...

Actualmente estoy en la fase de desarrollo de un sistema de Call Center personalizado y necesito un desarrollador de Asterisk para desarrolar nuevo sistema desde cero - El sistema se debe crear desde cero. - Las funcionalidades específicas que se implementarán o mejorarán incluyen la integración con API para sistema de cobros, automatización de llamadas e informes y análisis de datos en tiempo real. - En cuanto a la integración de APIREST de Redsys, se espera que el sistema sincronice bbdd con los pagos que se hagan de datos de clientes, registre compras automáticamente y proporcione capacidades de seguimiento. El candidato ideal debería tener: - Amplia experiencia con Asterisk y sistemas similares. - Trabajo pre...

I'm currently working on a project that requires expertise in FusionPBX configuration. I'm having trouble with the extension setup which needs immediate rectification. 1. SSL Certificate install 2. Configure Fail2Ban 3. Configure IPTables 4. Poly (Polycom) phones won't download config from server 5. Server will not issue config 6. Configure Shared Line Appearances/Multi...Configure Shared Line Appearances/Multiple Call Appearances 7. Configure SLA/MCA held calls to be able to be picked up on other phone 8. Configure SLA/MCA live calls to be barged in from other phones 9. Yealink phones do not connect 10. SIP Trunk disconnects from time-to-time 11. Create dialplan for phones so users don't have to press SEND or DIAL after typing in phone number You must be fluent in ...

Crear un VoiceBOT en asterisk para automatizar la atención con Amazon Lex , el flujo es una demo de muestra con máximo 4 opciones. Se debe entregar una Wiki del paso a paso, para replicarlo en un ambiente controlado y no trabajar en ambientes de terceros, una vez entregado esto y probado se finaliza el proyecto

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...

PLEASE READ THE POST BEFORE APPLYING. I'm looking for a Node.JS developer to implement a real-time messaging feature in my application. This feature will be used for live voice streaming between Google Dialogflow CX and Asterisk. Key Requirements: - Expertise in Node.JS: You should have strong experience in building real-time messaging features using Node.JS. - Integration with Google Dialogflow CX and Asterisk: You will be responsible for establishing a smooth live voice streaming between these two platforms. Ideal skills and experience: - Previous Work: Please provide samples of your previous work that demonstrate your experience in these areas. - Communication: Strong and clear communication will be key to ensure a successful integration of the live voice streaming...

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.

Preciso de um discador automático em freeswitch, segue algumas necessidades: Necessidades - Multitenant - várias empresas no mesmo servidor. - Franquias - Usuário com permissão de selecionar as suas franquias ou empresas cadastradas em seu nome, para que possa selecionar e visualizar todos os dados daquela empresa, todo o dashboard e relatórios, enfim, toda a interface da empresa. Possibilidade de colocar o logotipo de acordo com o domínio cadastrado. - Página administrativa para cadastrar um novo servidor freeswitch e setar qual a cps máxima desse servidor, para que possa aumentar ou reduzir a cps por servidor conforme necessário. - O sistema deve ser capaz de operar integralmente atrás do nginx, pa...

I am in need of an Asterisk telephony expert who can help configure and customize my system. To be more specific, I require assistance in setting up the IVR (Interactive Voice Response) system and defining call routing rules. The main tasks for this project are: - **IVR Setup**: Develop a user-friendly and efficient IVR system for seamless customer experience. The system should have clear and concise voice prompts, and be able to direct callers to the right department quickly. - **Call Routing Rules**: Define specific call routing rules based on different scenarios and requirements. The following sub-tasks are included in the project: - **Call queues and agents setup**: Ensure timely and fair allocation of calls to available agents. - **Call recording configuration**: Configure...

I am seeking an experienced Asterisk expert to set up a SIP server on my Ubuntu machine. The task needs to be completed within the day, so I am looking for freelancers who are ready to start immediately. Key requirements: - Set up a SIP server on an Ubuntu machine. - Ensure the server supports websocket connection. I am using JSSIP. - You can use pie socket websocket tester extension on chrome to make sure its working. Please bid if you have relevant experience and are confident in your ability to complete this task within the specified timeframe.

ONLY FOR ASTERISK AND FREEPBX DEVELOPERS! I'm looking for an Asterisk developer to assist me with configuring a sophisticated call routing system. The task is relatively small, but requires an advanced level of understanding in Asterisk technology. We have a Freepbx setup done and we want to setup an ARI bridge With this library Which is on the same Server So whenever any external call comes it should transfer the call to this bridge Key Requirements: - **Call Routing Configuration**: The task entails setting up a routing system that goes beyond basic and intermediate levels. This will include time-based routing and routing based on caller ID. - **ARI Bridge Integration**: I need the call routing to be integrated with the ARI

ONLY FOR ASTERISK AND FREEPBX DEVELOPERS! I'm looking for an Asterisk developer to assist me with configuring a sophisticated call routing system. The task is relatively small, but requires an advanced level of understanding in Asterisk technology. We have a Freepbx setup done and we want to setup an ARI bridge With this library Which is on the same Server So whenever any external call comes it should transfer the call to this bridge Key Requirements: - **Call Routing Configuration**: The task entails setting up a routing system that goes beyond basic and intermediate levels. This will include time-based routing and routing based on caller ID. - **ARI Bridge Integration**: I need the call routing to be integrated with the ARI

I'm seeking an experienced Asterisk developer to help me set up a VoIP PBX with Asterisk 16. The main goal of this setup is to implement robust call routing. We have setup twilio phone number with Asterisk. Currently calls are coming to extension Instead we want the calls to be transmitted to the ARI bridge that is running Key Responsibilities: - Understand and evaluate my requirements for a VoIP PBX system - Configure Asterisk 16 to support VoIP PBX functionality - Implement effective call routing mechanisms Ideal Skills & Experience: - Extensive experience with Asterisk, particularly version 16 - Proven ability to set up VoIP PBX systems - In-depth understanding of call routing mechanisms - Strong communication skills to understand and im...

I need a professional to help me set up my FreeSWITCH instance with FusionPBX on my Linux dedicated server. The job requires some troubleshooting as I'm facing issues with my current setup and need a reliable, long-term solution. Key Points: - I have already installed FreeSWITCH on my server. But it's not working as expected - the socket keeps closing during voice tests. - I've checked the server requirements for FusionPBX, but I'm not sure if all the necessary dependencies and packages are properly installed. - FusionPBX seems to be the main problem - likely due to dependencies or configuration issues. Ideal Skills and Experience: - In-depth knowledge of FreeSWITCH and FusionPBX. - Proven experience in setting up and configuring VoIP solutions on...

Hi I'm looking for a senior consultant to port a PHP script to C. This script uses the 'Asterisk AGI' library, but you don't need to have experience with it as long as you know C very well you will manage to use it. It also logs to syslog, performs various queries with mysql, includes an external file from some localization labels. Those are the 'building blocks'. The script is 969 lines. Let me know if you are interested. Max bid 500 euros. Thank you.

I am looking for a talented system integrator who can implement Vicidial (with a user friendly interface) or Goautodial v4 into our operations. Key Requirements: - This project demands to set up functionalities such as Call Recording, Predictive Dialing with webRTC, and Interactive Voice Response (IVR) for the V...for connecting ip phones / softphones for internal extension to extension calls (~50 extensions) Ideal Skills: - Previous experience with Vicidial and VoIP. - Capability to implement various functionalities within the Vicidial system. - Ability to plan and execute for larger scale (10-50 agents). In your proposal, please share a brief summary of your experience with Vicidial or goautodial or asterisk and VoIP, and provide an overview of how you'll approach tacklin...

I urgently require a skilled Node.js developer to integrate a specific API into my asterisk setup. The API in question is: Specific Tasks: - Integration of the provided API into my existing project of asterisk Requirements: - Proficient in Node.js - Experience with API integration - Ability to work quickly and efficiently This is a time-sensitive project, so a fast turnaround is essential. If you can start immediately and deliver high-quality work promptly, I want to hear from you.

I'm in need of an Asterisk developer to help with an API integration. The primary goal of this integration is to add new features to an existing system. More specifically, here are the details: - The integration will involve the existing GitHub repository: It's vital for the developer to be well-versed in this repository and be able to make the necessary modifications and customizations. The ideal freelancer for this project will have: - Strong experience in Asterisk development and APIs. - Familiarity with the provided GitHub repository. - Past experience in enhancing existing systems with new features. If you're skilled in these areas and ready to take on this project, please reach out.

I need help setting up a dial plan for my FusionPBX/FreeSWITCH system for emergency priority paging. Key Requirements: - Configuration: Experience in configuring FusionPBX and FreeSWITCH is essential. - Dial Plan: Specifically, you'll need to assist with setting up a dial plan that prioritizes emergency announcements. The focus of the system will be to deliver urgent messages in time-sensitive situations.

I am looking for Freepbx/Asterisk Expert to help me remotely migrate from AsteriskNOW to the newest stable FreePBX Distro on the same server. We also need VPN setup so our Yealink phones can connect to this system both with or without a VPN. You can help me configure one or two Yealink phones, and I can do the rest. AsteriskNow PBX Version: PBX Distro:10.13.66-22 Asterisk Version:14.7.5 System Admin Pro Commercial version is installed, configured, and activated. We will provide Anydesk access to the Windows system, which you will need to use to connect to our PBX. The system has putty with SSH access and web access to the PBX. We have less than 20 phones. Only apply for it if you are an expert at doing this and have a lot of experience. Serious inquiries ONLY,

Asterisk - Incluir el primer campo, del primer formulario, asignado a una campaña de predictivo, en el nombre del archivo de la grabacion de la llamada.

I'm in need of an Asterisk expert to assist with the installation and implementation of a basic setup with default features, specifically focusing on voicemail setup. Key responsibilities include: - Executing the installation and configuration of Asterisk for a basic setup - Ensuring that default features are correctly implemented and operational - Setting up and troubleshooting any issues related to the voicemail system Ideal candidates for this position should have: - Proven experience working with Asterisk installations - Strong understanding of default Asterisk features - Prior experience in setting up and configuring voicemail systems - Excellent troubleshooting skills Please include in your proposal any relevant experience and examples of previous...

I'm in need of an Asterisk developer who can assist me in setting up and configuring my Asterisk server, integrating it seamlessly with FreePBX and developing a custom dialplan to meet our call functionality needs. - **Configuration of Asterisk Server**: I require the selected freelancer to set up my Asterisk server to function efficiently and effectively. - **Integration with FreePBX**: The developer needs to ensure that Asterisk and FreePBX are fully integrated, allowing for seamless communication between the two systems. - **Custom Dialplan Development**: I require a custom dialplan to be developed. This dialplan is crucial for the advanced level of call functionality I require, which includes IVR, call queues, and advanced routing. The ideal ...

Installation and Implementation of Asterisk Key requirements: - Design and implement an IVR system that will support automatic answering and response based on the caller's input. - This system needs to provide relevant information to the callers interactively. Skills & Experience: - Proficiency in Asterisk installation and IVR setup - Experience with automatic response systems - Strong understanding of telecommunications and call center software - Problem-solving abilities and attention to detail are fundamental for this project.

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.

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I'm in need of an Asterisk developer experienced with Asterisk 13. The primary task is to configure the system as a VoIP server for internal communication. Key requirements: - Understand Asterisk 13 well - Strong experience in setting up and configuring VoIP servers The ideal candidate should have: - Proven experience in customizing Asterisk for similar purposes - Good understanding of VoIP technologies and protocols - Ability to work independently and deliver high-quality work on time Please note that the project involves setting up the system, configuring call flows and ensuring it's ready for internal use.

I require an experienced and proficient developer who is familiar with both WebRTC and Asterisk. The objective is to expand the functionality of my current Asterisk setup by integrating it with WebRTC. The specific features to be integrated are: -Voice Calling -Video Conferencing This project seeks to enhance the communication capabilities of my Asterisk setup, thus the ideal candidate would have considerable experience managing such integrations in the past. Sound knowledge of VoIP, SIP, and related technologies will be greatly appreciated. An understanding of the Asterisk framework and its API will be critical.

I am in need of an expert who can effectively integrate WebRTC and Asterisk to achieve real-time audio and video communication. This functionality is intended primarily for internal team members. Therefore, it should ensure: - Seamless and effective call routing and handling - Successful integration with our existing telephony systems - Provision of standard-definition quality for both audio and video communication The ideal candidate for this project has substantial experience with WebRTC and Asterisk, and has successfully executed similar integration projects in the past. This professional should also be knowledgeable about telephony systems and their integration for real-time communication capabilities.

I'm in need of an experienced ASTPP/Freeswitch developer to create an Interactive Voice Response (IVR) system for my project. The primary goal of this IVR system is to Play a pre-recorded message when a customer calls a DID, then route the call according to any routing rules configured on the dialplan. This means that the system needs to be designed to handle customer queries and direct calls to the appropriate department. Ideal Skills: - Proficient in ASTPP/Freeswitch - Experience in IVR system development - Strong understanding of customer support processes - Ability to design system for call routing The project is expected to be completed within a reasonable timeframe. A successful candidate will be able to understand the nuanced requirements of the system and d...

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

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

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

I'm in 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