Voip Software Jobs

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Fixed-Price - Entry Level ($) - Est. Budget: $500 - Posted
I need a marketplace script that connects service based businesses to customers with the marketplace in the middle. I need the ability to have customers pay the marketplace for service jobs, and then the marketplace pay the service business when the customer advises the job is complete. I also want the ability to split the payment from the marketplace to the business into two payments if I choose to. So if the total amount charged for the job is $350 and I take $20 for the marketplace commission then the remaining $330 could be split into two payments. One when the job is paid for initially and one when the job is complete. I want the process to be similar to the following: User browses to site. User selects type of service they need. User is presented with the option to get a custom quote from multiple service businesses and/or is given the option to immediately pay a flat rate price set by the business for their service. Customer funds go into escrow and once customer and business have accepted the work half of the funds are deposited into businesses bank account. Once customers work is satisfactorily completed the customer requests that the remaining funds be given to the business. I want the payment platform to be Stripe. I need the ability to set either the custom set-price lead fee which is added to the customers payment and/or a percentage fee that is added to the total given to the customer. This software feature set could be similar to what is available here: http://www.mymarketplacebuilder.com/#!features/bte9j
Skills: VOIP Software CSS HTML HTML5
Fixed-Price - Intermediate ($$) - Est. Budget: $850 - Posted
We provide ITSP services and are in the process of re-deploying an ITSP service which has been offline since the beginning of 2016. Asterisk, FreePBX and A2billing are already installed on the server. We need to configure the server to provide all the features of Linphone (an open source softphone from Belletone Communications), to provide all the features promoted by Linphone on their website. Included in the project is a registration form and minor branding of the user area of A2Billing (insert our logo). Please see the attached file for details.
Skills: VOIP Software A2Billing PHP
Fixed-Price - Intermediate ($$) - Est. Budget: $3,000 - Posted
We are looking for a freelancer who has strong experience with WebRTC communication system. Here are requirements. - iOS and Android module which easily include to any project. - Javascript sample code which works with mobile app. - Signaling server in NodeJS - Voice calling - Video calling - Group Voice calling - Group Video calling We did these features in house, but it turned out that it is very difficult to make stable voice/video calling in the mobile app. So we are looking for WebRTC expert. Please, not apply to this job who will make the feature by googling or reading the tutorial.
Skills: VOIP Software Android iOS Development WebRTC
Fixed-Price - Intermediate ($$) - Est. Budget: $2,500 - Posted
Hello, We need a quote to develop a WebRTC communication suite with those features: - Talk between customers - Video call between customers - Instant Messaging between customers - Security for all the communications, from the SSL accessing to the page application to all the encryption systems available within the framework to secure and encrypt any communications made with the application. - Create an admin-panel for us to create new users, modify it, revoke with the possibility to group users under a group (useful for example for revoke multiple licenses from the same customer in one clic) - Test for all the mobile devices and desktops - Possibility to add a contact in a friends list (assigning a name) and searching for username (like Skype does) - Branding with logo, color and title OpenTok (is white label) - Adding a second language and language switch - Auto-deleting of chats history, calls history with pre-set timers: 10Minutes / 30Minutes / 1 Hour / When the application is closed / Manually with a button or When the page is closed - Show the status of a contact (Idle, Away, Available, Offline etc..) - Possibility to receive all the messages received when offline and back online (Like Skype does), this will also under the timer control of auto-deleting option (as discussed before) ########################### OPTIONAL FEATURES TO EVALUATE - Possibility to send files (encrypted) and receive files I've found https://opentokrtc.com this products created by https://tokbox.com that is a very fast and useful platform where build a Webrtc application without worries about to create or use external api to develop something of stable and compatible with almost all of the mobile and the desktop browsers. If you have an other framework to use, with same functionalities and fast as this we can evaluate it. Please post only the real price to develop this and ask me all the informations do you need to make a realistic quote, i'm here for you. Thank you Regards
Skills: VOIP Software HTML5 json
Fixed-Price - Intermediate ($$) - Est. Budget: $50 - Posted
For my 2 Voip phones I have an account with Grasshopper. I need someone to comb through them and ensure incoming calls are routed to the correct v-mail. My struggle is configuring Google voice, Grasshopper & AT&T (cell) to work together. You should know going into this that I don't typically use data or WiFi and that seems to complicate matters for me. You should also know that I'm guessing, unless you are a super braniac that this project might take several hours. I need help simply because I can't keep track of all the different variables. variables being 6 different phone numbers for 2 people. I am open to ideas about how to smooth things out. Below are notes I took from when I tried to do this myself: My numbers are: 406-660-1828 - owned by Grasshopper, Voip 406-660-3063 - owned by Grasshopper, Voip 406-660-1075 - owned by AT&T, cell 406-660-0989 - owned by AT&T, cell 406-200-8367 - Google Voice 406-272-6520 - Google Voice 1828 to 1075 w/0 predial at 10:24am = 406-200-8367 on cell caller ID & my new voicemail msg. Get v-mail notification on cell phone. w/predial at 10:28am = 406-660-3064 on cell caller ID & my new voicemail msg. Get v-mail notification on cell phone. 1075 to 1828 at 10:31 = goes straight to new v-mail. Got Grasshopper notification in gmail along with transcript. Shows as ext 0 in Grasshopper. House phone to cell w/o predial = 272-6520 on ID & my new v-mail w/predial = 660-0989 on ID & my new v-mail (should be Starr’s) 1075 to 3063 = recording “please leave a message”. V-mail goes to Grasshopper as ext2 with no transcript in email. 1075 to 272-6520 = Starr’s voicemail Needs to be Where are my text messages? Can I segregate phone activity for my 2 phones? Need to change caller ID thingy for office phone to 1828. I shouldn’t have to do the caller ID thingy because my number is 1828. Can I get TM transcripts of my voicemails? So I can make the house phone look like 1075 by calling what number? Calls from office 1828 but show 1075. Then 1075 gets routed to 1828 when I’m home. So if 1075 is my main number. Calls go to 1075 AT&T Calls go to 1075 routed to Grasshopper. Can I get a TM notification? Calls from house 3063 but show 0989. Then 0989 gets routed to 3063 when Starr’s home.
Skills: VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $100 - Posted
Please ignore our budget and place your best bid for the project. We search for a expert to implement a VoIP SIP validation of a list of phone numbers. As a result of the validation we need as much as possible details about the given number. The goal is to reduce the manual effort of a call center by knowing if the given phone number is a valid or invalid one. Also it would be nice to know what type of availability/unavailability was the result of the phone validation. To be clear a simple regEx validation with https://github.com/googlei18n/libphonenumber shall be done in front of the VoIP checks, but they are not enough, since they do not check the real phone status. As a input you will get a CSV list. Your implementation shall be in Apache POI 3.14 and the column containing the phone number shall be configurable by a the column header name. We do not need a database, since we have a custom implementation. BUT your model classes have to properly designed to be later able to use it easily with a custom persistence layer. To ensure you are not a simple poster please add the result of eight power two on top of your application. As a result we expect to get a updated CSV file with the new columns which you need to provide for the end user to understand the status of the phone number validation. Ensure that your implementation has a component for CSV reading, one for VoIP validation and one for CSV writing. Ensure the components are independently usable! We prefer for VoIP / SIP Java Standards. As a runtime environment we expect runable on: - in JavaEE 7 (wildfly 10) - Java 8 As the development environment we expect: - Eclipse Neon - maven With your application provide us following details: - delivery date for milestone 1 - brief class level design with public methods - a list in word or excel of statuses collected for the later VoIP validation (see also https://de.wikipedia.org/wiki/SIP-Status-Codes) - delivery date for milestone 2 - working VoIP calls, no extraction of validation results - delivery date for milestone 3 - working validation extraction and implemented CSV writing
Skills: VOIP Software Apache Jakarta POI SIP
Fixed-Price - Intermediate ($$) - Est. Budget: $3,700 - Posted
Looking for PHP developer who have excellent experience in developing requirement based web portal having monitoring, reporting and exceptional GUI. For this project we are looking for someone who can develop a web based software integrating the features of the following existing softwares: 1) ViciDial 2) Go Auto Dial 3) DialFire 4) Avatar 5) Twilio etc Interested people please contact asap as we have an urgent requirement and let us know a time frame and budget for the same.
Skills: VOIP Software Asterisk FreePBX MySQL Administration
Fixed-Price - Expert ($$$) - Est. Budget: $600 - Posted
We are looking for an experienced engineer/developer with knowledge in Siemens OpenScape/HiPath and VoIP. The work is to configure the Siemens OpenScape PBX (provided by freelancer, this can be a demo o temporary licence for the poc only) in a lab environment. Configure 3 extensions/softphones (3 SIP). no need to configure for external calls. Once this is up an running, we will need the freelancer to setup Orecx or similar packet sniffer to create voice and metadata files for the recording calls. We will provide port span configuration. Integration with CSTA will be a plus but must be optional. Then using Web Services (API will be provided) you will need to create a service running in the background that will send voice and metadata to a remote machine. Please do not apply if you have no experience on this. To apply start the covert with "I am a dreamer". You will need to provide the documentation and the code needed to accomplish the task.
Skills: VOIP Software Call Handling Freeswitch Network Engineering