Asterisk perl jobber
As a sysadmin developer, I'm in need of an asterisk specialist to build a Docker Compose script or a bash script for an interactive vocal server. This project is multifaceted, carrying out outbound calls and saving responses in a database. Key responsibilities are: * Creation of an Interactive voice response (IVR) system. * Outbound calling function connected with my API for automated scheduling of phone calls. * MySQL database integration to securely store the recorded responses. For this assignment, it would be ideal if you have proficiency in using Asterisk, Docker Compose, API integration along with comprehensive database management skills It would be a cherry on top if you have prior experience constructing interactive vocal servers. Let's connect to disc...
I'm looking for a professional who can install Asterisk PBX to facilitate a call routing system. Key Requirements: - Asterisk PBX will function primarily as a call router, modifying incoming caller ID's to the outgoing trunk DIDs. - The system should handle both incoming and outgoing calls efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice tran...
Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe
I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..
...am looking for a PERL, HTMX, MYSQL programmer who can help me with an intermediate level project. The ideal candidate should have experience in these areas and be comfortable starting a project from scratch. I will give you a dump from my MySQL table. From which I need an interactive HTML interface, based on Perl, HTML, HTMX to this table. I need to be able to search, add, edit and delete entries as well as combine two entries into one, all via one HTML::Template file with a single perl script as interface to the MySQL table. I do want an active submit button for all changes, as well as a single interactive upload button for each field of the MySQL table. also each field needs to be validated via a given specification. All code needs to be well documented and commen...
...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 have a Perl script that searches a CVS file and generates HTML. When this script has zero results, it currently displays a page stating "SORRY there are no RESULTS". I need this behaviour to change: instead of generating a custom error message, I want the script to send a 404 call to the Apache server. Key Requirements: - Modify the Perl script to send a 404 call to the server when no results are found. - Implement a custom error message to be displayed on the 404 error page. Additional Details: - No logging of details of 404 errors is required for monitoring or debugging purposes. - The script should still function as normal when results are found, only altering its behaviour when no matches are found. - The ideal candidate for this project should have experien...
I'm looking for a skilled Perl expert to help me remove duplicates in an Excel dataset. The removal process should be based on multiple columns, ensuring a thorough cleanup. Ideal Skills & Experience: - Strong proficiency in Perl programming - Proficiency in data analysis - Expertise in Excel and handling large datasets - Prior experience in removing duplicates from Excel based on multiple columns Communication: I prefer to have project updates through video calls, so please be comfortable with this form of communication. Your ability to walk me through the process will be highly appreciated.
Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.
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...
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'm having issues with a specific user function in my Pgsql ( postgresql 8.4) database that needs fixing. After transferring server and DB from real PC to the VM, ... 1365 | 1365 32645239 | ADMIN | _util_update_stat | 27 | 37511 | 37511 calls And new doesn't have any calls of _util_update_stat at all which makes Stats table empty You need to track why is this happening & fix it. Server is a bunch of scripts, but seems it doesn't matter cause all scripts just READ this stats table (i guess) Pgsql expirience is a must Perl expirience is a plus. Plz no "manager" bids, don't waste your time, will talk with developers only. (if you are a "manager" it will be seen after the 2nd tech question Plz don't wa...
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.
Estamos à procura de um especialista em Issabel e Asterisk para implementar um sistema de Resposta Vocal Interativa (IVR) na nossa plataforma de servidor Issabel 4.0.0-6. O objetivo deste sistema é realizar entrevistas telefónicas automatizadas onde os respondentes possam responder a perguntas pré-gravadas utilizando o teclado do seu telefone. O sistema deve ser capaz de carregar uma base de contactos para realizar automaticamente as chamadas telefónicas e capturar estas respostas numa base de dados para análise subsequente. Além disso, o sistema IVR deve incluir a funcionalidade de conversão de texto em voz (TTS) para facilitar a criação e atualização de prompts de voz. No entanto, também d...
...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, 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.
II. the task #2 (less important but fast to fix) email notifications do not work for some reason (on server with webUI @159.65...) The email sending works with cron there contains 1 task here /var/spool/cron/tabs/root and 4 here as i understand /var/spool/cron/tabs/serg guess user serg is not configured properly(but it is a full disk copy, so not sure) or cron tasks are not active or lunching.. also i see 3 cron procs on old sever (ps-A|grep cron) and only 1 on new server... you can change email in scripts and test (the email sending is working ok tested it with small /root/) p.s. you can check working details on old server @213.168.... I. the most important but could take more time The statistics section do not work at all on the New server here is active clients on OLD server(s)...
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.
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...
I'm in need of a skilled freelancer who has expertise in Perl scripting, primarily for file manipulation tasks. Key Requirements: --Gather date from a table in oracle. Based on that date find the records available in a home directory and store it in a .txt file. Skills and Experience: - Proficiency in Perl scripting is a must. - Experience in handling various file formats, specifically .txt and .csv. - Strong understanding of file operations including renaming, formatting, and sorting (although the specific task was skipped). I need the code done within 2 days and willing to pay 3000 which would be double the cost of what is given .
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.
I'm currently using PERL 5.10 system and requires a proficient expert to aid in modifying how passwords are hashed in my system. Initially, we employed SHA-256 but we're looking to switch to PBKDF2. Ideal Skills and Experience: - Proficient in PERL 5.10 - Familiarity with SHA-256 and PBKDF2 hashing methods - Prior experience in password re-hashing Task Objectives: - Identify current SHA-256 hashing method in the system - Modify hashing method to use PBKDF2 instead - Validate successful implementation of the PBKDF2 hashing method The project necessitates a strict focus on security and confidentiality given its sensitive nature.
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: ...
Script Feature: - Login to a list of SSH servers by domain or IP (from text file) - Use a set of usernames + password (can be multiple passwords) to login to the servers. - Run Bash Script on the connected server (which comes from a txt file on the server running the script, not on the connected to server) - Timeout if SSH server did not answer. - Display results at the end of a run (show which servers the script did not run because of errors). Login simultaneously to all available ssh servers. Not server by server.
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 to convert the following Perl code to PHP: The expected output is a PHP class that can convert a docx file into text. The attached zip contains a mockup of such class (with demo code) and a test docx file. The task is straight forward. If you place a bid, expect it to be accepted. If you fail to deliver, you'll be subjected to a bad review. This is to avoid wasting my time with autoposters and people who lowball the bid and then try to ask for more money. You have been warned. Read the brief, check the code and place a bid accordingly.
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.
...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...
I'm seeking an expert to help cr...an expert to help create a Dockerfile that runs Perl and DBD::mysql so that I can run a legacy website. This should be a simple task if you have the knowledge. Specific requirements include: - A Dockerfile is the only output required, roughly corresponding to mine - It should build and run in wsl, i.e. docker run -p 8080:80 image_name - It must use Apache, Perl and the CPAN module DBD::MySQL - It should have been tested on a MySql database, the file is provided to run on localhost:8080/ - A screenshot of the above is required to prove that it is working (you just need a MySql Table with fields Id and Name) Candidates should ideally have: - Extensive experience in Docker and a basic knowledge of Perl - Understanding of Per...
...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...
...working. MochaHost tech support got the Import buttons on the forms fixed but suggests they may not be able to fix the XML & XML+XSL submission buttons at the bottom of the forms. Here's what they say about that: When the server is upgraded, the Perl modules in the new server might not be fully imported. That's why your website doesn't work because some Perl module dependencies might be deprecated. MochaHost support says: We highly suggest you talk about the error with your Web Developer to ensure they can help you add the missing path on the Perl module. Based on our check, your website form processing is missing pointing to the `File::Spec` module which has been installed on the server. It seems like that might be a fairly easy fix and I have...
...performance tuning, sizing and capacity planning, database backup and recovery in a Unix O/S using RMAN and Oracle export/import utilities, administration, and maintenance to provide backup and disaster recovery. • Knowledge of installation, configuration, and maintenance of multi-node Oracle RAC enabled databases, including Oracle ASM and Oracle CRS management. Knowledge of PL/SQL packages, shell, and Perl scripts for task automation, and proactive database monitoring and notification using Oracle OEM Grid Control. • Demonstrated written and oral communication skills reviewing and editing established database policies and procedures and developing new SOPs for database software installation and configuration. • Experience with Cloud(AWS/Azure/OCI) migrations (DB a...
I'm looking for seasoned coders that excel in Perl, to reconfigure my current Perl email script. The primary aim is to make it efficient in sending out emails using Google's SMTP. Key Deliverables: - Modify Perl script to use Google's SMTP via the Net::SMTP::SSL or Email::Send::SMTP::Gmail modules - Integrate secure connection to Gmail - Integrate Email authentication No requirement for handling attachments is necessary. Need to understand how the forms work. Proficiency in Perl, understanding of SMTP protocols, specifically Google's SMTP, and experience with Net::SMTP::SSL and Email::Send::SMTP::Gmail modules are crucial for success in this task.
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 skilled Perl coders to modify my email script. This script currently sends out emails well after a web form submission using the Net::SMTP::SSL module. Success in this project will mean updating the script to instead utilize the Email::Send::SMTP::Gmail module. Ideal Skills: - Proficient in Perl coding - Familiarity with Net::SMTP::SSL and Email::Send::SMTP::Gmail modules - Understanding of Google SMTP Key Responsibilities: - Reconfigure Perl script to work with Email::Send::SMTP::Gmail module - Ensure the script remains efficient for sending out web form submission emails - Test script after modifications to verify seamless integration with Gmail SMTP
