Asterisk voicemail web jobs

...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for a skilled Python developer who specializes in VoIP and PJSIP development. The successful candidate would ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up S...Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP trunking and DID. - Familiarity with analog to VoIP co...

Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.

...**Requirements:** Create GoHighLevel account. Complete Gohighlevel Client Onboarding Sales funnel setup Landing page creation Automation and workflows SMS and email template design and automation Pipeline configuration ChatGPT integration Custom domain setup Payment integration Membership area setup Media buying for lead generation website in GHL Automate lead nurturing through follow-up campaigns via voicemail, calls, SMS, emails, and Facebook Messenger. Implement appointment scheduling features. Integration of calling software Configure key performance indicators (KPIs) for tracking our sales and marketing performance. Set up custom reports to analyze data and gain insights into our campaigns. training and troubleshooting GoHighLevel platform to our team. Migrating contacts ...

...aspects of this project include: - Setting up a voicemail system: Implementing a voicemail system that captures and organizes customer messages is crucial. Prior experience in voicemail management is desirable. - Call flow optimization: I need a system that can efficiently route calls and ensure they reach the right person. Any experience with call routing systems would be very beneficial. - Prioritizing urgent calls: Sometimes, urgent calls may come through. Being able to set up a system that can recognize and prioritize these calls is a huge plus. - Scalability: My business is growing, and I need a solution that can grow with it. A solution that is easily scalable would be ideal. I'm looking for someone who has: - Experience in setting up voicemail s...

...automatic voicemail drop using Zapier's webhook functionality. I use a CRM (Spark Membership Software) that can automatically trigger a webhook. I want Zapier to capture the webhook and execute an automatic phone call dropping a voicemail in the process. Key Responsibilities: - It's critical to successfully connect and properly configure the Zapier webhook service. - Once the webhook has been activated, the task would be to automate dropping a voicemail. Specific Requirements: - There should be two phone calls triggered from the webhook at the same time. One is a call from my personal phone number, the other is a call from a private number. This is so the receiver cannot answer the call and instead receives the voicemail drop and a missed call fr...

Someone who has developed some prerecorded audio call sending system. The idea. is to send hundreds of thousands of calls daily to users, the user receives a voice call that must be listened to as a voicemail with pre-recorded audio. Has anyone created a project like this?

...Audio/video conference calls and scheduled meetings Call transfer and multi-call management (pause and resume) One-to-one and multi-participant conversations Intuitive message delivery status (IMDN) SMS/chat with Message Syncing for all devices (Meaning sms/chats will be synced, so users can see the conversations). Recordings on server (We have APIs to do that and check if user is allowed). BLF Voicemail Plus a few additional features. The ideal developer for this project should have experience in: - Mobile app development, specifically on iOS and Android platforms - Proficiency with Linphone or similar VoIP applications - UI/UX design to make the necessary interface changes - Integrating applications with third-party services - Meeting tight deadlines As I r...

I am looking for a competent professional with experience in Freepbx and Avaya IP integration. The primary objectives of the project are call routing, call forwarding, and establishing a unified voicemail. What I am looking for: - Experience with Freepbx and Avaya IP office 500 V2 systems - Proven track record with telecommunication technology - Capability to integrate functionalities even with gaps in specific call routing or forwarding options Please note that while some specifics on the call routing and forwarding options are not outlined, I expect freelancers to provide suggestions based on their experience and expertise. I look forward to working with an individual who can make these systems work smoothly together.

...solution, I'm seeking guidance from a skilled freelancer experienced with both Raspberry Pi and Asterisk server setup. Key Project Details: - I do not require a full-fledged PBX setup, just a basic VoIP service facilitated through my Raspberry Pi. - The primary feature I'm interested in implementing is voice calling. Required Skills: - Proficiency in configuring an Asterisk server on Raspberry Pi. - Strong understanding of VoIP and related protocols. - Ability to guide and explain the setup process clearly. Your Role: I've tried to set it up but no voice can be heard on the other end. So this task is mainly a trouble shooting job. Your primary role will be to walk me through the setup of Asterisk on my Raspberry Pi, ensuring proper configuration f...

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

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

I need a virtual Indian mobile number service that comes with key features like call forwarding, SMS forwarding, and voicemail. The critical functionality this service should have is call-based OTP verification. This OTP verification should be facilitated by an automated voice message that will provide users with the OTP. While there hasn't been a need for additional features such as an OTP recall option, multilingual support or a customizable message template, I would definitely prioritize a service that delivers on prompt and seamless OTP verification. Freelancers with prior experience developing or integrating similar services will have a distinct advantage. Crucial skills for this project include experience with VoIP services, OTP systems and automation. Understanding o...

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...

We're in need of an IT consultant who has a solid background in working with Verizon tech support. Our 15-year-old business mobile number has been experiencing accessibility issues and we are looking to "re-secure" it so that we can po... Your Responsibilities: - Assess the account related to our business mobile number. - Implement necessary porting measures to "re-secure" our number. - Facilitate the porting process of our number to a VOIP service provider of your choice, ensuring that all essential features are seamlessly incorporated. The key VOIP features that are crucial for our business include call forwarding, voicemail to email, and an auto-attendant setup. If you feel you have the right blend of skills and experience necessary for this p...

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

i am looking for some one with expertise in podio CRM . i need a drip automation installed. where the system automatically follows up with our non answer clients. either every day, every 2 weeks or every 30 days. with s text sms and voicemail. More details: every 2 weeks, some every 30 days it depends on client. Both Text SMS and Voicemail Fully personalized with client name and specific details

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...

looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protoc...FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related ap...

I'm in need of a proficient specialist to establish my SIP trunk and an IPPBX server. Here are the functionalities I require: • Making both inbound and outbound VoIP calls. • Call recording, call forwarding, and voicemail to email features for my IPPBX server. You'll also need to set up three or more SIP trunks. Experience with VoIP technologies and SIP systems is crucial for the success of this project. Knowledge of REST APIs will be advantageous. Please ensure your proposal demonstrates your related past projects and expertise.

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

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

I need an expert in 3CX Pro systems to fully optimize my call flow setup. Key tasks will include configuration of: - IVR - QUEE - QUEE WAIT 2 Minits after - If Closed Play Message and redirect to Voicemail - Or All Users Bussy - Play Message and redirect Voicemail I have atached one Pic.

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 an expert in 3CX Pro systems to fully optimize my call flow setup. Key tasks will include configuration of: - Closed Play Message and redirect to Voicemail - All Users Bussy - Play Message and redirect Voicemail I have atached one Pic. i need like this !

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.

...Ringba dashboard, where we manage voicemail tracking and recordings, and our WordPress system, ensuring seamless data flow and synchronization. • Utilize your expertise in call technology to parse and transcribe voicemails into text-based entries within WordPress, creating a new User Type labeled as "Voicemails." This process should replicate the structure and fields used for current website lead users. • Develop a system to analyze zip codes entered during calls, utilizing our extensive list of all U.S. zip codes to automatically populate the corresponding city and state in the voicemail user's profile in WordPress. • Configure WordPress to include specific fields for user-entered zip codes, text transcriptions of voicemails, direct links to ...

I need an expert in 3CX Pro systems to fully optimize my call flow setup. Key tasks will include configuration of: - Closed Play Message and redirect to Voicemail - All Users Bussy - Play Message and redirect Voicemail I have atached one Pic.

I have a particular project requiring expertise in Freeswitch, preferably with exposure to a Debian Linux environment. The key objective is to use Freeswitch for answering machine detection. The main requirement is to capture and analyze voicemails. We want to build a custom module or ready solution which can detect human or machine on the outbound dialer. The solution should work on any freeswitch version and without any licensing restriction. Please note, we have already used AVMD module and it just detects beep. We want to detect the machine or human from the initial screening of voice only. Skills and Experience Needed: - Solid experience with Freeswitch answering machine detection - Familiarity with working on Linux systems - Ability to effectively capture and analyze audio data.

I'm in need of an expert to configure FusionPBX for 1-5 tenants with Twilio. This includes: - Call Routing functionality - Recorded welcome - Voicemail capabilities An important aspect of this project is customization for each tenant, to include: - Individual Dial Plans - Custom Voicemail setups - Tenant-specific features and settings (something to start with so we can grow packages) - Potential ongoing support as we grow Prior experience with FusionPBX is essential for this job. This project requires a deep understanding of client-focused VoIP solutions. Sound technical skills in FusionPBX platform setup with Twilio, VoIP configurations, and tenant-centric customization are key in this role. The deliverables should not just meet the standard FusionPBX platform requ...

...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...

...experiencing an issue with the FreePBX phone system that I use, specifically, I'm unable to consistently receive calls on all my soft virtual extensions or to my welcome greeting. Voicemail functionality is also part of this setup. Here's what I need: - Evaluation and Diagnosis: Investigate why call reception works only some of the time across all extensions. - Troubleshooting and Repair: Identify the problem and implement a fix to ensure consistent and reliable call reception. Ideal Skills Required: - Extensive experience with FreePBX Phone Systems, particularly concerning soft virtual extensions and voicemail functionalities. - Strong understanding of VoIP systems, and hands-on experience in troubleshooting call reception issues. It is crucial for th...

Hi Andrew M., We'd like to hire you again to record another voicemail message for us. Here's the script: "Thank you for calling Rhinohide. We are at the 2024 Perth Caravan and Camping Show from 21st to 24th March, Thursday to Sunday. All orders received during this period will be fulfilled on Monday, 25th March." Looking forward to hearing from you. Cheers!

I'm in need of a cost-effective ringless voicemail drop software for our customer outreach initiative in the United States. Key Features: - Automated Messaging: The tool should effectively and efficiently send voicemails to our extensive customer base without disturbing them with a ring. - Pre-recorded Ringless Voicemail Drop: We must be able to send our pre-recorded messages successfully and seamlessly to our contacts. - Capable of Handling Large Volumes: The software must have the capacity to reach more than 10,000 contacts. Ideal Applicant: - Candidates should possess strong experience in creating ringless voicemail drop software. - Applicants should demonstrate sound understanding of implementing automated messaging and pre-recorded voicemail drop ...

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.

I'm in need of someone who can assist in creating two friendly and casual, female-voiced voicemail greetings for my business. The messages will be custom - I will provide specific scripts for each one. IDEAL CANDIDATE: - Experience in voice-over work, especially in voicemail greetings - Ability to deliver a friendly and casual tone - Must have a female voice Please include samples of your work for consideration. I'm excited to hear your interpretation of my scripts!

...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...

I'm looking for a skilled freelancer to establish and configure a FreePBX telephone system with the following features: - Call Recording: Implement reliable call recording for quality assurance and training purposes. - Voicemail: Set up voicemail boxes that are easy to access and manage. - Call Forwarding: Enable seamless call forwarding to ensure no call goes unanswered. I am anticipating fewer than 10 extensions for my organization at this moment. Key Skills: - Extensive experience with FreePBX systems and VOIP technology. - Capability to deliver a secure and user-friendly setup. - Strong understanding of potential integrations (CRM or ERP) is a plus though not immediately required. Please highlight your experience with FreePBX in your proposal. I am looking for so...

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...

I'm looking for a soothing, warm female voice talent to record my phone answering service message. The script is short and should be delivered in under 30 seconds. Ideal skills and experiences: - Professional experience in voice acting or narration - Understanding of the "warm and soothing" voice tone - Proficiency in delivering concise and clear voice instructions for a phone answering service. The following are the exact two audio files we need recorded: 1: "Thank you for choosing Skyline Missions Massage, where you can embrace the escape. Please hold while we connect you with one of our staff to schedule your getaway." 2: "We're unable to get to the phone right now, but please visit us at to schedule your next escape. We offer our signature ZenD...

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, poten...

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 ...