Asterisk PBX Jobs

I am experiencing issues with my PBX and Voip systems. Specifically, I'm having connectivity issues. I am using multiple services/platforms for my Voip but I'm not sure which solutions or troubleshooting methods I should use in order to fix my current issues. I'm in need ofa freelancer to fix the issues remotely.

I am looking for a freelancer who can help me build an encrypted VoIP system using military-grade encryption. I have no preference regarding VoIP protocols, however I anticipate the need for up to five simultaneous calls. If you have the necessary skills and expertise to build this system, I would love to hear from you. Something like;

My project entails the installation of playsms on a Linux server. I require full setup and configuration assistance, but no additional features or customizations beyond the basic installation. This project will be completed rapidly and with ease by a qualified professional, so that I can take full advantage of this powerful text messaging software.

I am looking for a freelancer who can help me create a Twilio workflow for my sales team. The main purpose of the workflow is to streamline our sales process and improve our customer experience. Please see the attache dfile so that you can supply a fixed price quotation. Ideal Skills and Experience: - Experience with Twilio API - Experience creating Twilio workflows - Familiarity with sales processes and customer experience Specific Requirements: - Call forwarding feature is a must-have - Customization is required for the call forwarding feature - The call forwarding feature should allow for forwarding to different departments or agents Overall, I am looking for someone who can help me create a customized Twilio workflow that will improve our sales process and customer...

Hi I am a 68 year old singer guitar player not professional . I have hearing loss and use a pair of mix fi blue headphones and a headphone amp to get the sound and gain I need to play on stage . Just invested in IEM system form Sensaphonics . 3 dme touring pro .I have been IEM route before an always wind up back with headphones . Need an engineer that specializes in IEM monitoring . I can never get the fidelity or gain I need from iems . Going through Presonus live digital board . Everybody says that iems are better this is a 3000 dollar system I can’t even get it to sound decent

I need a webrtc to sip gateway to be implemented so I can connected some webrtc softphones (asterisk webrtc softphones on our Odoo CRM) to twilio sip domain

I am looking for a developer to create a SIP Softphone that is compatible with Android operating systems. The main requirement of the Softphone is to make phone calls. I should be able receive inbound calls and read any dtmf tones Ideal skills and experience for this job include: - Experience with SIP Softphone development - Strong knowledge of Android operating system - Experience with VoIP and telecommunications - Familiarity with SIP protocol - Strong attention to detail and ability to deliver high-quality work - Excellent communication skills to ensure project success.

We are upgrading machines from Centos 7 kernel 3.10.0-1160.71.1.el7.x86_64 to the mainline version 6.3.5-1.el7.x86_64, and as such we need to be able to compile and install the drivers. Dahdi-linux Dahdi-tools Using the Kernel And Kernel Headers And Devel package Deliverables will be a step by step process on what steps are required to get the package to install under this version of the kernel and kernel-headers. We have the drivers function 100% (Compile and Install) under the latest stock kernel 3.10.0-1160.71.1.el7.x86_64, however, there appears to be an issue with compiling under the newest kernel indicated above. We can not provide a development machine. You should download the lates version of Centos 7 Then upgrade the kernel and Kernel Source as indicated above. ...

I am looking for a freelancer who can help me with my FreePBX Dynamic Routing project. My current setup is a Single Server. I need a developer who can set up a mySQL database to capture caller inputs in order to enable dynamic routing. I require 2-4 different routing options. The contractor will only be able to access the system through me I don't have remote access. Also, the person will need to walk me through each step of development and implementation. Ideal skills and experience for this job include: - Strong experience with FreePBX and mySQL - Knowledge of dynamic routing - Ability to create customizations and integrations when necessary - Excellent communication skills and ability to work independently

I am looking for a skilled developer to create a program that integrates my Fritzbox VoIP calls with a conversational AI chatbot. The goal is to enable voice calls to be converted to text, send the text to a ChatGPT model trained with knowledge about my company, generate a response, convert the response text to audio using Google Text-to-Speech API, and stream the audio file as a reply through my Fritzbox. The workflow of the program should be as follows: Receive incoming VoIP call on the Fritzbox. Convert the voice call to text using a suitable speech-to-text engine like Google. Send the text to a ChatGPT model that has been trained with information specific to my company. Generate a response from the ChatGPT model based on the input text. Convert the response text to audio using Google ...

I am looking for an experienced Asterisk API developer to create an Asterisk Webhook API. The developer should have extensive expertise with Asterisk and be able to develop the API using an open source programming language. The API must include functionality for call control, call recording, and voicemail management. We are open to suggestions for the programming language and frameworks used to create the API, however, it must be versatile and reliable. We expect comprehensive documentation for the API as well as ongoing support to ensure it functions properly. Thank you for your time and we look forward to your proposal.

I need an asterisk device on my server with binding to sip number. all on my own crm laravel. I want to be able to create a web client so that you can make calls from anywhere.

I just installed Vici Dial on a Server, I have also Magnus Billing on an another Server which means I want now to make auto dailer so I fixed Vici Dial but I have never heard about this before so now need someone who can help me just to open users or what ever is needed to configure it.

This would be to develop and install a small call flow that runs on FreeSWITCH to screen calls. YOU Must know FreeSWITCH and have used it. You will need to install the FreeSWITCH stack and then write custom call flow for it. Integrates with Google Contacts API and Pushover notifications. you should have experience building call flows, databases, webforms/pages etc... this call flow will answer the call if the caller is in the subscribers google contacts # list, you will send the call to a second # if they are NOT in the google contacts # list, you will ask the caller to say their name and company name you will record that and transcribe it you will send the transcription via pushover to the subscriber you will the call the subscriber at a different number and play the info recorded if ...

I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode