Voip Software Jobs

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Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
We need to dev our mobile app for android using webrtc integrate with our freeswitch-kamailio server.
Skills: VOIP Software Freeswitch
Fixed-Price - Expert ($$$) - Est. Budget: $200 - Posted
I need a software (similar to a Skype Dialer) that automatically calls phone numbers given from a list of numbers in CSV format. The software should record the call statuses like, Call Answered, Call Disconnected, Call Un-Answered, Call forwarded to Answering Machine , number not in service etc.
  • Number of freelancers needed: 3
Skills: VOIP Software Interactive Voice Response SIP Skype
Fixed Price Budget - Expert ($$$) - $100 to $500 - Posted
Purpose/Project Setup and configure Asterisk for the embedded platform. Asterisk setup needs to be configured for the embedded environment for the low flash memory footprint. Only required modules shall be installed onto the target. Utilizing Asterisk, the project needs to achieve functionalities as described below. Target flash memory allocation for Asterisk: <7mb (excluding kernel and other packages) Target platform: Beaglebone black with Kernel 4.1.6 Functionality • Ability to support two call groups o Call group A (SIP client connection)  Can make calls using SIP trunk  Can receive calls from SIP trunk o Call group B (SIP client connection)  Can make calls using SIP trunk to only defined number (number shall be defined in the config file)  Cannot receive calls from SIP trunk • SIP notify. Must support following custom SIP notify (Subscribe/notify) o Message-summary (message format example: message waiting:3/5) o Provide example on how to send updates to SIP client • Ability for asterisk to connect to multiple SIP client o To register, SIP client shall provide one or more following information  Username  Password  IP address / port • Set the maximum number simultaneously connected SIP clients. • Set maximum number simultaneously connected calls • On an incoming sip call from SIP trunk, asterisks must ring all connected SIP clients o Ring must stop when one user answers the call. • Call logs o Asterisk shall provide call logs o Set the maximum number of call entries to save   Project deliverables Asterisk • Select the stable version of Asterisk and provide justification for the selection • Provide source of the selected version to install Asterisk • Build instruction for settings up asterisk (on a Linux platform) o Enable only required components • Memory footprint Asterisk configuration files • Configuration files to achieve the requirements Sample SIP trunk (NOT a part of this project) • A simple c++ SIP server that can be used to connect to Asterisk • Requirements for the application: {​To be defined}​
Skills: VOIP Software Asterisk Embedded Linux
Fixed Price Budget - Intermediate ($$) - $300 to $500 - Posted
I have installed BigBlueButton on my server but need a meeting manager that allows the following: Account creation Sign in/Sign-up for meeting organizer and meeting participant User profile User Account Password reset Schedule a meeting Send invitation by email Creating Meeting Agenda Meeting password sent by organizer Schedule instant meetings File Manager and Agenda Manager Integrate to VOIP [Dial In Feature] I need this to be integrated to my instance of OpenEdx LMS. you need to write in python as Django App for it. Very similar to this: http://www.talentlms.com/blog/how-to-use-bigbluebutton-in-talentlms/ Note: Experience with BigBlueButton required. Our Engineering team will support you on this.
Skills: VOIP Software
Hourly - Expert ($$$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We would like to have a corporate phone system. We would like to use Asterisk, specifically Incredible PBX due to the fact that it is relatively easy to administer. I have a Dell R730 server running Windows Server 2012R2 on which to host the VM that will run it. This is VM we are interested in running: http://nerdvittles.com/?p=14073 Please let me know you have downloaded image of file in the article. The only problem is that the OVA image won't work for us because we have found that virtualbox is unstable on our system. All our VMs are hyper-v which are more capable and offer better performance in Windows. There are utilities which will convert a VirtualBox image to hyper-v image of a VM. I'll let pick which one to use. Also we need a SIP trunk provider. I am thinking about Sigmavoip but I am open to suggestions. Then all I ask is that you get it up and running and teach me how to use it. We want support for 3 types of phones. I would be surprised if this takes you more than a day. We are almost there as it is and I am pretty sophisticated technically. I just haven't done this before. 1. SIP-T48G Ultra-elegant Gigabit IP Phone http://yealink.com/product_info.aspx?ProductsCateID=1206&CateId=147 We definitely want HD voice turned on. That's a huge feature for us. We definitely want AES/SRTP or stronger encryption enabled. We want to get the most out of these really nice phones 2. VOIP Apps running on mobile devices 3. "Soft phones" which are virtual phones running in HTML5 browsers The need is kind of urgent so the sooner the better. I will kneed to know how to administer it.
Skills: VOIP Software FreePBX SIP VOIP Administration
Fixed Price Budget - Intermediate ($$) - $1,000 to $2,000 - Posted
I need WebRTC and Websockets Developer to develop high levelAPI for us that we can use in our application easily without much effort. The API should provide the voice chat functionality using WebRTC (where browser supports it), and falls back to Websocket (or Socket.io). Since it will be used in a Web Application based in Meteor framework, we can utilize AtmosphereJS stable packages. The end product needs a simple demo UI where we should be able to call from browser to browser and following browser support is need Browser Support Needed: Internet Explorer 9+ Chrome Firefox Safari * The API should be able to scale, and voice quality should be like Skype for Web. But it will usually be used for 1-1 calls for now. * No Plugin (need for client to install), it should work without it. * It should not take too much bandwidth, ideally should work on 4G connection without consuming too much data like Skype, Facebook Voice Chat. * Should be able to use some compression in order to achieve less data usage, in case of fallback to websockets.
Skills: VOIP Software JavaScript Socket Programming WebRTC
Fixed-Price - Expert ($$$) - Est. Budget: $3,000 - Posted
Hi. We need to launch a complete Mobile Dialer solution. which Comprises on. 1. Android Mobile application for calling (This will deliver the basic calling features needed for a voip service.and able to work with all SIP Standard softswitches, and support high quality Codecs Such as G.729,AMR, IAX2 Or ILBC which have high compression ratio and ability to offer Crystal clear Voice Quality). 2. Linux (Asterisk) base server (which will be integrate with the Android mobile app and ((3)) Vpn/tunnel server. This linux server will have all management of Voip Needs. It will communicate with third party softswitches(Voip Servers). 3. Tunneling Server. ( It can be a vpn server/ByteSaver server/ or some Tunneling Server that will be utilized for bypassing Voice ports blockage issue of Voip By the Isps. For better performance can use IAX2 protocol because issues such as network congestion, firewalls and low bandwidth no longer affect voip Traffic). add our skype: gplusitsolution
Skills: VOIP Software Android App Development Asterisk Java
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
we are looking for VOIP Engineer/Administrator who can work with Kazoo, Kamailio, FreeSwitch, CGRates. complete configuration and modification to make full class 5 switch and also experience in network and app development for cloud base VOIP system o Voice and Video Telephony o Instant Messaging (IM) and Presence o Voice/IM conferencing o Interactive Voicebox and Voice-to-mail o Fax-to-mail, Webfax and Fax Clients o Serial and Parallel Call Forking o Periodic, time-based Call Forwarding (CFU, CFB, CFNA, CFT) o Inbound and outbound Call Blocking and anonymous Call Rejection o CLIP/CLIR, DDI, DID and Extension Dialing o Signaling over UDP, TCP, TLS, WS, WSS o DTLS-SRTP transcoding for WebRTC bridging
Skills: VOIP Software Asterisk Freeswitch SIP