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 27,245 opiniones, los clientes califican nuestro VoIP Developers 4.87 de un total de 5 estrellas.
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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.
Looking for someone who can configure auto dialer for freepbx We already have it working but now it dials the customer and when that is picked up, it dials the agents. It should start dialing the agent the moment it starts ringing on the other end
On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. this module had to be compatible with latest freepbx 16 version. - Freepbx Asterisk distribution : - superfecta module to work with : fixed price : 100 $ - ASAP within 1 day.
This is a test task to find the right partner for onging work on this topic. Expected solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to make a self executable app (in Java), which detects incoming SIP/VoIP calls. On incoming calls, pjsip (c++ lib) app opens a browser with a (caller-)URL (details see below) Examples: ) So the app has to work on a desktop (windows, linux, macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working sip account after award) Later (not scope here) ports to iOS/Android required too The caller-URL contains some query parameters, like the callers number. The app can run only one instance and is listening t...
We are setting up a FusionPBX with a peering trunk. But we are having registration failed with error 403. Would be great if you can help us with clearing the error to register the sip peering as well as setting up an IVR. Please do not bid if you are not a professional of VoIP.
I'm looking for a developer to create an AI based VOIP system. The main focus of this project is to implement natural language processing as a feature of the VOIP system. I'm looking for a full feature set, with all the bells and whistles that come with the AI feature. Additionally, there will be some existing systems that need to be integrated into the new VOIP system, we want to offer this service as an API, so experience in integrating existing systems is highly desired. The idea behind is people can automate their inbound call centers and our AI bot will take all calls Example system is