VoIP, otherwise known as Voice over Internet Protocol, is a digital solution for telecommunication that utilizes the transmission of voice through the use of data over the internet. This technology is typically adopted for both commercial and privately used applications. VoIP developers specialize in programming telecommunication systems using data transmission to provide efficient services for audio and video functions.
VoIP developers are highly sought after professionals as they can create a versatile and advanced system for both desktop and mobile devices. This can include creating a more interactive experience for customers with multi-way calls, conference calls, video conferencing and voicemail systems. VoIP developers can set up your business’s phone system such as forwarding calls remotely to other devices with extensions or even communicating with customers utilizing software like Callcentric, Flowroute or Voxbone.
Here's some projects that our expert VoIP Developer made real:
- Secured and configured private networks by implementing security measures like IP SLA, NAT and ACLs
- Integrating SIP trunks between billing platforms, PBXs and VoIP applications
- Setting up Postfix mail server on Linux systems with TLS authentication and encryption
- Installing and configuring sip telephony software like ASTPP into businesses
- Developing web applications that can send and receive SMS messages utilizing VoIP APIs
Our expert VoIP Developer have created numerous projects to help business owners manage their telecommunications in a professional bet. Our developer have an array of experience in creating a secure and reliable system customized to each client demands. If you're looking to improve or create your business’ communication system, then post your project today on Freelancer.com to hire one of our skilled professional VoIP Developers today!De 26,398 opiniones, los clientes califican nuestro VoIP Developers 4.87 de un total de 5 estrellas.
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Need enterprise level vicidial customized installation
I am looking for an experienced developer to help me design and implement a full Cloud Pbx system with mobile application (example: 3CX). I have decided to use a multiple-server architecture, and I do require specific hardware and software solutions. I also have a deadline in mind, so this project needs to be completed quickly and efficiently. If you believe you are a fit for the job, and can help complete my project in a timely manner, please contact me. Thank you!
Require voip expert for installing fusionpbx and configuring it with our termination provider ip address.
Hi I am looking for voip and programmers to help to train me and get me started on VoIp business
Soy de Brasil, como vai? Representamos a empresa VOIP, ramais, sip video conferencia e coloquei dentro do CRM Bitrix um Sip Trunk mas ainda falta alguma autorização da Goto (antiga LogMein)......por enquanto nao fará parte do escopo. Preciso Integrar CRM Bitrix via API com meu sistema ominichanel simples, jogando as conversas chat para dentro do CRM. Projeto perfeito seria : VOIP que ja ta no marketplace Bitrix + ominichanel jogando los protocolos, pesquisas de satifacion e auditoria que ja estão desenvolvidos (frame flask phyton) .... acredito que será tudo por API. Projeto mais que perfeito :VOIP + Gravações integrados no CRM,call center e gestão de projetos Bitrix + chat GPT para acompanhar historicos tarefas cronogramas e...
Setting up a Wan ethernet connection and aggregate 2 WANs Setting up lte 4g as a failover alternative Set up rdp with port forwarding Security on users entry Vpn between rdp users Vpn to othersites Setting up a vlan Prioritise Voip traffic with
Tasks: 1.- Define hardware specifications for a VM, linux distro and version and we'll provide corresponding ssh credentials. 2.- Setup GenieACS 3.- Create initial config of GenieACS 4- Create provisioning templates for AVM Fritz!Box (2 models, 7530 and 7590) 5.- Specific to routers: Being able to monitor these parameters from the devices: ping, manage WAN settings (PPPoE dialin or DHCP), manage PPPoE credentials, manage vlans, manage multiple VoIP accounts and VoIP PPPoE login Provisioning data will come from an external RESTful API with JSON response. Query parameter is the provided serialnumber of the specific device. It will return all credentials (PPPoE, VoIP, etc.) at once. A sample will be provided. We need to have the project completed within 2-3 weeks. This project is...
I am looking to set up a Twilio Ringless Voicemail System with VoxDesk for 1-3 users. This system should be able to drop voicemails without ringing the line.. I realize you've probably never used VoxDesk so we're looking for someone who can test it out. It is a no-code platform but you need to be familiar with Twilio. We will also want our service to forward to a different telephone system. We have a $150 budget for this. Thanks
I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.
voice broadcasting, automations call , IVR set up software with help of SIM card IVR device
we need an expert in flutter to develop flutter app that will combine 2 plugins #1 flutter gsm dialer # =sip client the app will rout the flutter gsm dialer mic audio to the flutter sip client speaker and rout the flutter gsm dialer speaker audio to the flutter sip client mic To route the microphone audio in a Flutter GSM dialer with a Flutter SIP client speaker, you can use the Flutter AudioRecorder and AudioPlayer plugins to capture and play audio, respectively. You will also need to use a SIP client library to establish a SIP session and communicate with the SIP server. Here are the general steps you can follow: Add the flutter_sound and flutter_sip plugins to your Flutter project. The flutter_sound plugin provides audio recording and playba...
I need an expert in installing FreePBX system on our server. I have specific requirements and I want these requirements to be fulfilled before setup. If you think you are the right fit, please let me know – I am looking for someone to start this project as soon as possible. Thank you for your time!
Hi, I have goautodial 4 setup on server with ssl/tls everything is working great except for the carrier dial plan that does not make calls out. All i need is assistance with the dialplan to make sure that the calls can be made from the webrtc phone built into goautodial 4. The software is setup already the solutions looks like it works fine. I assume the problem is with the dial plan. I need someone that can go through the system to make sure calls can be made out using the web phone built in.
the project have 2 api First api --- route audio from whatsapp mic to sip client speaker and from whatsapp speaker to sip client mic second api this api will reside in android and will be connected a to remote server that will be able to instruct the api in the android to initiate call in whatsapp and to monitor the call stat like ring dial . call connected and end call and to be able to terminate the call both api will work on android devices with android version 7 and up
we are looking for a exceptional good expert about SIP protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome t...
Hello, We have server on which is installed FreePBX. We want to buy Yeastar TG100 GSM and Cisco SPA502G. We need to configure this 2 devices so when call is made to GSM SIM cart that is in Yeastar the call to be trough VOIP to Cisco
Dear Freelancers, We have a video bridge jitsi installed. We need to achieve the following: Need to be able to login from a raspberry pi with a calling of url. The url call will happen if someone dial an extension on an asterisk pbx. Once the extension dialled from a SIP phone asterisk call a url which makes the raspberry online in the jitsi meeting room. From this time the raspberry is opened the meeting roon and participants can log in to the room. The raspberry pi need to work with a Konftel conference system . So the Konftel will be connected with USB to the rasperry and then the raspberry need to use its camera and microphone in the meeting.
An Android tablet will be placed at the door. The tablet will log in to a local SIP server, set in the settings page. During idle state, it will show a static image or video of the user's choice, set in the setting's page. When the user taps on the screen, he will be presented with 2 choices, "Guest" or "Delivery" Tapping on 1 of the choice will initiate a SIP video call to the preset extension number. The extension number will be set in the settings page. Video of the conversation will be recorded. Video will only be from tablet to server, there will not be any video from server to the tablet. There will only be audio from server to tablet. After the call ends, it will return to idle screen. 1 setting field to define where the videos will be saved, opt...
Hello i need someone to install and configure vicidialer or any other webhosted dialer webrtc need to know the cheap hosting/vps/server so i can buy and let him setup ... Regards
I need some DID experts with a long experience who can set up the whole process which is needed to create DID numbers together with the Telecom company. Please do not respond if you do not have that experience.