Sugarcrm asterisk vicidialJobb

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

Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error. [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descr...

Creazione di un software che da un lato dialoghi tramite AMI (Asterisk management interface) con un centralino e dall'altro in base a delle condizioni impostabili, invii del messaggi tramite un gateway whatsappa già pronto chiamato che permette di inviare (pagando) messaggi illimitati tramite WhatsApp. inoltre questo software deve poter inviare dei messaggi anche da una lista di numerazioni che gli si fanno caricare.

I'm looking for someone with experience in React, who also understands Asterisk servers and JSSIP. I have an existing asterisk server that I connect to with a React phone client to make outbound and inbound calls. Everything was working fine, but I recently refactored my code to use Redux instead of Context. In doing so, now when I make outbound calls I get this error [ERROR] JSSIP UA not initialized And when making inbound calls it says the number is not available. I can provide the old, working version of the code, and the new version that doesn't work. I just need someone to look at it and tell me what I'm doing wrong after converting to Redux. If you can help, please include the word "briefcase" in your bid so I know you've read the descri...

...guide team members in their career development. IMPORTANT: Utilize expertise in Vicidial to understand call algorithm blocking, spam tagging, and implement effective circumvention strategies. ***If you don't have this experience, do not submit application. Let's save each other's time. As this is a medical call center, basic knowledge of medical terminology is required. Any prior experience in a healthcare call center will be considered a plus. Demonstrate a dedication to fostering a positive work environment and creating a successful team. ------ Position Overview: We are seeking a dynamic and experienced Call Center Manager to oversee our medical outbound account. The ideal candidate will be an expert in Vicidial and possess a deep understanding of ca...

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & fix this issue. Budget: $50

We are looking for an engineer proficient with Raspberry Pi, as we are in need of developing a VoIP PBX system on a Raspberry Pi 3 Model B+ . The end goal includes the integration of specific features into the system such as: - Call Recording - Voicemail to Email - Conference Bridging - Additional bespoke features Your expertise should include not only Raspberry Pi but also Asterisk and RASPBX/FreePBX. We are aiming for a robust, stable, and user-friendly system with custom features tailored primarily to business needs. The successful contractor will be required to develop the system on his own Raspberry Pi and submit an IMG file for loading onto other Raspberry Pis. The successful contractor will be working with our software developers so bids from individual contractors only w...

Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.

...providing top-notch customer service and sales solutions in the healthcare industry. We prioritize excellence in every interaction and strive to deliver optimal health solutions to our valued customers. Job Description: We are seeking a dynamic and experienced Call Center / Customer Service / Sales Expert to join our team. The ideal candidate will be proficient in utilizing VICIDial and possess extensive experience in voice-based customer service and sales roles. As part of our team, you will play a crucial role in delivering exceptional service to our clients and driving sales conversion initiatives forward. Responsibilities: Utilize VICI Dialer efficiently to manage and handle customer interactions effectively. Provide exemplary customer service and sales support to client...

I am urgently seeking an experienced telephony and data processing specialist to configure my Grandstream UCM6302A with Asterisk. The core functionality required includes receiving calls, playing a welcome message, meanwhile working with Caller ID and Web API to determine where to forward the call. When a call comes in, • first a welcome message is played () • in the meantime the caller ID will be sent to web API preferably POST, but get can be if POST is not possible () •The API will respond json array: - {forward_to: 33356853 } - Forward the call to 33356853 - {forward_to:0 } - Play message and terminate the call • If forwarded call is not answered by Agent in three rings, another call to API will

I am looking to incorporate AI features into my call center system, specifically Vicidial and Asterisk. As these platforms form the core of our operation, it is essential that any alterations enhance our outlay without disrupting the existing structure. Key Aspects of the Project: - AI Implementation: Even though I haven't specified the exact AI features to integrate, I'm interested in potential focus areas such as speech recognition and transcription, natural language processing, or sentiment analysis. Proposals that offer comprehensive strategies addressing these or other AI fields will be highly considered. - Dual Integration: The AI features must be incorporated into both Vicidial and Asterisk, aligning and harmonizing their performances. - Effi...

Hello, I'm seeking individuals to join a call center management team with two distinct roles to fill. Candidates should possess basic proficiency in Vicidial & Magnus Billing, alongside fundamental customer support skills. Role 1: Customer Support involves order processing and fulfillment using Vicidial & Magnus billing systems. Role 2: Involves sourcing a consistent supply of inbound DID Numbers for the United Kingdom, United States, Australia, New Zealand, and Finland. Weekly pay ranges from 7000 to 10500 INR based on availability. If interested, please reach out via direct message

i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much

I need a professional to assist with the installation and configuration of the latest version of Vicidial on a Linux operating system. The key tasks include: - Vicidial software installation: Ensure a smooth and efficient installation process, troubleshooting any complications that may arise. Familiarity with various versions of Vicidial is a plus, notably the latest. - Setting up configuration: Proactively set up and optimize Vicidial configurations to cater to our business needs, ensuring seamless functionality. Ideal candidates should have solid experience with Linux and Vicidial, including its installation and configuration. Your bid will be more favorable if you've previously completed similar projects successfully. I'm happy to provide...

I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.

I need a professional to assist with the installation and configuration of the latest version of Vicidial on a Linux operating system. The key tasks include: - Vicidial software installation: Ensure a smooth and efficient installation process, troubleshooting any complications that may arise. Familiarity with various versions of Vicidial is a plus, notably the latest. - Setting up configuration: Proactively set up and optimize Vicidial configurations to cater to our business needs, ensuring seamless functionality. Ideal candidates should have solid experience with Linux and Vicidial, including its installation and configuration. Your bid will be more favorable if you've previously completed similar projects successfully. I'm happy to provide...

...featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can provide the necessary Asterisk API documentation. Ideally, you are able to show me a s...

Hello, I operate a fax communication system leveraging Hylafax, integrated with an Asterisk server and iaxmodem, all running on Alpine Docker. While our outgoing fax functionality performs flawlessly, we are encountering persistent issues with incoming faxes. Specifically, incoming fax pages frequently get cut off midway, resulting in incomplete document reception. We are in search of a seasoned Hylafax professional who can diagnose and rectify this particular issue. Expertise in managing Alpine Docker environments and Asterisk/iaxmodem configurations will be highly regarded. Desired Expertise: Demonstrable experience with Hylafax, especially in fixing issues related to incoming faxes. Deep knowledge of Asterisk and iaxmodem. Proficiency with Docker containers, pre...

...seeking a VoIP consultant for improvement of my existing computer-based VoIP system. The purpose of the project is twofold - improved communication efficiency and enhanced call quality. Key Tasks: - Analyzing the current computer-based system setup - Implementing the connection of Physic SIP to asterisk on the cloud for enhanced call quality Ideal Skills and Experience: - Proven experience as a VoIP consultant - Excellent knowledge of IP PBX system - Experience with connecting Physic SIP to asterisk on the cloud - Ability to improve communication efficiency and call quality. Kindly submit your proposal outlining your plan to achieve these two goals along with your previous relevant work. Looking forward to finding a VoIP specialist who can provide a swift and efficie...

Using another dialer now and want to see if vici can improve our stats in terms of connection, answers, voicemail detection, autorotate to local area codes, spam detection etc.. Basically looking past the basic setup and someone who has used vici and knows how to add code into the system. Put "I can help" in the first line so I know you read the entire job post. Also please detail some things you could suggest in terms of adding codes. We are calling home owners in the US.

...developer experienced in WebSocket/AudioSocket technologies and Asterisk integration to develop a solution that enables real-time transcoding with OpenAI Whisper through gRPC. Requirements: 1) WebSocket/AudioSocket Integration: Develop WebSocket/AudioSocket functionality to facilitate real-time audio communication with OpenAI Whisper. 2) gRPC Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [a...

I am searching for a skilled software developer with a strong background in Asterisk, Dialer, IVR and VOIP technologies. Although I haven't specified particular functionalities, general familiarity with call routing, call recording and interactive voice response (IVR) would be beneficial. The ideal candidate for this job should be proficient in: - Designing, implementing, and maintaining Asterisk software - Developing dialer functionalities, with emphasis on auto dialing, click-to-dial, and predictive dialing - Ensuring system is up-to-date and secure Freelancers who apply should provide any past work, detailing their experience and including project proposals, if any. If you believe that you have the expertise to effectively take on this project, I encourage...

We need to create an Asterisk aplication (v18) for Service at workshop by appointment for vehicles. This aplication must have voice recognition in English /Spanish language and Text to speach language with Google technology. Functionality: i) Welcome. ii) Select Languague. iii) Request Data: * Type of vehicle * City * Car licence plate * Telephone number. * Date request. * Time request. d) System will confirm first date/time available and customer will confirm. At this time, application will not have conectivity with real system....only must confirm next day and time users told. But it will have errors control, confirmation recognized data, etc.....

...engineer to implement an Opus encoder and decoder in C# for seamless integration with Asterisk. The project involves handling voice audio from Asterisk, decoding it, incorporating Text-to-Speech (TTS) functionality, and encoding the synthesized speech before sending it back to Asterisk. The ideal candidate should possess the following skills and experiences: - Proficiency in C# programming language. - Extensive experience with audio processing and Opus codec. - Familiarity with Asterisk, SIP, and IVR systems. - Knowledge of Text-to-Speech (TTS) integration. - Ability to deliver high-quality code within specified timelines. Main Tasks: - Implement Opus encoder and decoder in C#. - Integrate the solution with Asterisk for audio processing. - Incorporat...

In our quest for an outstanding and interactive VoIP web app, we're in need of an experienced developer, ideally one who is well-versed in Asterisk. The main features we're after, though not limited to, are call logging and reporting, an IVR system, and call routing and forwarding capabilities. Key Skills and Experience: - Extensive experience in VoIP development. - Proficiency in Asterisk. - Ability to develop an IVR system. - Experience in call routing, call forwarding, and call logging mechanisms. Given the nature of our project, it’s crucial to have a level of experience in these areas. Understanding the intricacies of these features is what will drive the success of our project. Though we have not specified it in the initial questions, potential co...

I have install Asterisk (with freepbx), for gui to a Raspberry pi 4. I need a DEVELOPER TO WRITE a succesfull CODE and configurate the system to: Make a AUTO-VIDEOCALL from extention ''0'' to extention ''1'' or ''2'' or BOTH at the same time. The exctention ''1'' or ''2'' are log in to androip application, called ''PortSin'''or another app called ''Linphone''. Extention ''0'' is already loged into Raspberry with the programm ''Linphone''. When i press a button connected to GPIO 21 i want the audiocall to be started for 9 seconds. If the call is anwsered I also need to OPEN the DOOR by pushing ...

I'm seeking a skilled professional to set up a new VoIP asterisk system with primary features of call routing and call recording. Must have the ability to have custom caller ID for outgoing calls. This project entails configuration for a small scale operation with less than 10 users/phone lines. Key Job Requirements: - Proficiency in setting up and configuring VoIP systems - Outstanding knowledge of asterisk - Experience with call routing and call recording This task requires an efficient and effective approach, understanding the needs of a smaller user-base. Someone with a strong track record of setting up VoIP systems and the knowledge to troubleshoot potential issues would be perfect.

We used Asterisk as our phone system And we used Flutter's Sip-UA as a client The communication platform is WebRTC The problem we have is that when the internet suffers a few packet losses during a call, the client leaves the channel and then it is completely silent until the call is disconnected. We simulate the same scenario with a softphone, after the internet is disconnected and reconnected, the call continues and there is no problem. Are there any friends who can guide me?

I'm experiencing issues with the current function of Vicidial Survey's human detection. Specifically, we're facing: - False positives - False negatives - Inaccurate detection These issues are largely present when we call English speakers based in the United Kingdom. Consequently, I need enhancements specifically for the detection of UK English. Although perfection is not the aim, the updated system should be capable of detecting English words accurately, such as simple greetings like "Hello". Skills and Experience Required: - Proficient in Vicidial Survey's software - Excellent understanding of UK English, particularly accents - Prior experience with improving speech detection accuracy

Seeking a professional to integrate Asterisk VoIP into my existing Odoo system for my call center operations. Key tasks include: - Installing Asterisk VoIP - Setting up integration with Odoo - Implementing Call Recording functionality - Configuring Interactive Voice Response (IVR) - Implementing Automatic Call Distributor (ACD) The system should be able to handle 21-30 concurrent calls at peak times without any lag or quality degradation. Skills & Experience Required: - Proficiency in Odoo and Asterisk VoIP setup and integration - Proven track record in call center technology setting-up - Ability to work under strict deadlines and handle pressure - Understanding of call center operations and key processes - Excellent problem-solving abilities - Attention to d...

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I am in need of a specialist for customizing my Asterisk PBX system. The primary tasks will focus on two areas: call routing/forwarding and call recording/monitoring. Call Routing and Forwarding: I require an extension based routing system to be developed. Call recording and Monitoring: The second part of this project revolves around call recording and monitoring. Specifically, I require an on-demand call recording system, coupled with real-time call monitoring. I am expecting the freelancer engaging with this project to have a deep understanding of Asterisk PBX system customization, particularly in call routing, forwarding, recording, and monitoring. Prior experience in similar projects will be a great advantage. Strong communication and troubleshooting skills will be ke...

I am setting up asterisk with Amazon Chime for sip trunking, but am very new to this and am having difficulty where making a test call doesn't work. If you think you can help, please bid the amount you would charge, and use the word "alarm clock" somewhere in your bid so I know you read the description and the bid isn't automated. If things go well, I will likely need more help in the near future setting up other details of the Asterisk config. For now though I am just trying to get a test call to work.

ONLY APPLY IF YOU HAVE EXPERIENCE IN VoIP and SIP. I am in need of a skilled freelancer with intermediate experience in Asterisk to set up a VoIP and SIP trunk on AWS. The successful candidate should have solid background with AWS infrastructure and Asterisk application. Key Deliverables: - Set up VoIP on AWS using Asterisk - Implement a SIP trunk functionality Requirements: - Must have experience in setting up and managing Amazon Web Services (AWS). - Asterisk application experience is essential. - Previous work on similar projects will be advantageous. Please make sure to include your experience with Asterisk and AWS in the application. I look forward to receiving your proposals and working with you on this exciting project.

As a Part-Time Vicidial Developer, you will be integral to the continuous improvement and maintenance of our Vicidial system, ensuring optimal performance to meet our organization's unique requirements. This role is perfect for a committed developer with a passion for telephony solutions and a solid background in Vicidial development, particularly in GoHighLevel integration. Responsibilities: Customization and Configuration: Tailor, configure, and uphold the Vicidial system to align seamlessly with organizational needs. Collaboration: Work collaboratively with cross-functional teams to comprehend business requirements and implement Vicidial solutions accordingly. Troubleshooting: Identify and resolve issues related to Vicidial system perfo...

I am seeking a specialist to help integrate AWS with our Asterisk PBX system. Key responsibilities will include: -Diagnose and resolve organic AWS to Asterisk PBX connection issues -Assure AWS and Asterisk PBX platforms interface correctly Ideal skills: -Extensive experience with both AWS and Asterisk PBX is essential -Experience with system integration and troubleshooting -Ability to work independently and find creative solutions to complex problems Whilst the specific functionalities and preferences for this project have not yet been finalized, these will be communicated upon initiation of the project. Your experience and suggestions will be valued in these discussions.

I am seeking a skilled developer who can build an Asterisk solution on Amazon Web Services (AWS). The system should be proficient in handling a load capacity of 10-50 concurrent calls, ensuring efficient performance and high availability, thereby exceeding the traditional communications infrastructure. Key Features: - Providing VoIP services - Enabling Conference Calling - Incorporating Call Recording Preferred Skill Set: The ideal freelancer for this task should be experienced with not only telephony engineering, but also cloud environment configuration. Knowledge in Asterisk operation, VoIP, and AWS platform are absolutely essential. Proven experience in similar projects is highly appreciated. The goal is a robust and scalable Asterisk environment that leverages th...

As your client, I'm looking for an expert in both public cloud technologies like AWS, Google Cloud, or Azure, and Asterisk, to build a custom environment for my activities. The functionality I'm seeking includes: - Voice over IP - Conference calls - Interactive voice response I expect this environment to handle between 50 and 100 concurrent calls. The ideal freelancer for this project would have extensive experience with both Asterisk and public cloud deployments, and be able to provide examples of similar successful implementations they've undertaken in the past. They should be able to ensure high-quality voice communication, smooth run of conference calls and efficient functioning of interactive voice response systems. Moreover, the freelancer should ensur...

I am looking to create an Asterisk environment within the Google Cloud Platform (GCP). This environment will primarily serve the following critical functions: - Call center operations: Facilitate business communications and manage a large volume of calls efficiently. - Interactive Voice Response (IVR) functions: Integrate an efficient IVR system to assist callers and route calls automatically based on user inputs or chosen menus. The system must be designed to handle less than 100 calls simultaneously while ensuring seamless operation and minimal downtime. Ideal experience and skills needed: - Proficient with the Google Cloud Platform — this project requires deploying services in GCP. - Solid track record in building and managing Asterisk environments. - Understandi...

Looking for Asterisk implementation engineer who is good at configuring inbound and outbound setup.

As a Part-Time Vicidial Developer, you will be integral to the continuous improvement and maintenance of our Vicidial system, ensuring optimal performance to meet our organization's unique requirements. This role is perfect for a committed developer with a passion for telephony solutions and a solid background in Vicidial development, particularly in GoHighLevel integration. Responsibilities: Customization and Configuration: Tailor, configure, and uphold the Vicidial system to align seamlessly with organizational needs. Collaboration: Work collaboratively with cross-functional teams to comprehend business requirements and implement Vicidial solutions accordingly. Troubleshooting: Identify and resolve issues related to Vicidial system perfo...

Seeking an experienced developer to enhance our existing dongle voice SMS termination system with chan dongle and asterisk on Linux using custom firmware. The main focus will be on three key aspects: voice call termination, SMS termination as well as custom firmware development. While keeping the focus on the dongle voice SMS termination system, you will also need to utilize your Java skills to ensure seamless integration with our Linux server control system. Skills & Experience - Developing firmware - Working with chan dongle and asterisk on Linux - Java programming I invite applicants who can showcase both their past experiences relevant to this role along with any other skills and competencies that might be beneficial for the execution of this project.

I'm looking for a...someone with experience in Asterisk and SIP integrations. Key Project Aspects: - Asterisk Setup: Your task will be to set up Asterisk with multiple features including monitor calls + transfer + hang up calls, an IVR system, and call recording. I require a high level of accuracy and efficiency in the configuration of these features. - SIP Service Connection: Additionally, I need to connect Asterisk to a SIP service. We have an AI that speaks and give response to your recorded message. So the idea is to connect this AI to the Asterisk that is connected to the SIP so it can speak with people on the phone. Ideal Skills: - Experience setting up and configuring Asterisk, IVR systems, and call recording - Knowledgeable abou...

I am looking for a skilled Asterisk developer to customize and integrate my existing AI systems into Asterisk. Specific tasks include: - Integrating Asterisk with intelligent speech recognition and natural language processing systems. - Incorporating chatbot functionality for automated customer support. - Implementing AI-powered call routing and prioritization. The ideal candidate should have prior experience in Asterisk customization, AI integration, and development. Knowledge of speech recognition, natural language processing, chatbot technology, and AI-based routing is crucial for this project.

...PBX principles. In particular, your role will require the development of a Telegram bot, with integrated services improving the functionality of Asterisk. This will be a comprehensive project, necessitating precision and a thorough understanding of programming. Key features should include: - Both DTMF and Any Caller ID functionalities. - Implementations for these features should enable Call Recording. - The bot should also provide hold music capabilities, specifically using pre-recorded tracks. Ideal candidates will have a solid foundation in both Python and PBX systems. A background in telecommunications, specifically interacting with Asterisk systems, is also beneficial. Your expertise should encompass Bot development, particularly within the Telegram platform, as the...

In the search for an experienced developer, I require a web2call application based on Asterisk. This web2call should focus on enabling international call capabilities. Specifically, the created application should run adeptly on Linux platform systems. Additionally, the application needs to support one-on-one call functionality for its users. Hence, any freelancer bidding on this project should preferably have: - Proficiency in Asterisk software - Experience with building web2call applications - Knowledge of Linux platform compatibility issues - A background in creating systems for making international calls Your understanding of one-on-one call functionality support will be highly valued in this project. This project will be a significant contribution to connecting people ...

Hi I have a working ipsec tunnel (one host as a client and one host as a server) and I need to make sip calls from asterisk (as a client) to freeswitch (as a server) thru this tunnel. It works outside the tunnel but when I try to use the ipsec tunnel IP I can't hear audio and get call hangup after few seconds. I can provide ssh to both client and server and it's easy to 'switch' between using and not using the tunnel. Please write down the changes that you make. Also please only bid if you have experience with freeswitch. Max 100 euros. Thank you.

Requiero la instalación y configuración de Festival y Cepstral en varias voces en español, así como hacer la configuración necesaria para que las grabaciones que se están grabando en WAV se graben en wav, todo lo anterior en asterisk 11.25 en Issabel.