Voip Software Jobs

60 were found based on your criteria {{ paging.total|number:0 }} were found based on your criteria

show all
  • Hourly ({{ jobTypeController.getFacetCount("0")|number:0}})
  • Fixed Price ({{ jobTypeController.getFacetCount("1")|number:0}})
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
Hourly - Entry Level ($) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We have a fresh FusionPBX box that is already working without any problem. Now we need to make some extra configs and we will need to complete next tasks. You must be very expert in FusionPBX / FreeSwitch. * Add native G729 protocol * Get a document to change / reset admin after moving / cloning VPS server to new IP * Implement mp3 natively to send * Install spanish and portuguese IVR / voices * Setup a couple of outgoing dialplans and configure default prefixes in 3 countries * Help configure tenants interconnection - we want to offer one single number accessible to all tenants * Be available for future configs / needs * We're interested in long term business relation You bid price per hour and the time you need to implement this (2h for an expert should more than enough) Thanks
Skills: VOIP Software Asterisk Freeswitch
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
I am looking for somebody experienced with Twilio TwiML/API/Etc. who can setup a TwiML Bin for one of my numbers. I want 2 code examples to do the following. 1) Use any Caller ID of my choosing for call forwarding, as well as outbound calls. 2) Record both sides of the calls. 3) Use Twilio backend system to track call recordings per phone number. I want to be able to view ALL recordings for inbound/outbound calls. I will want to verify everything is setup correctly. (If you have a better suggestion I am all ears). I doubt this will take somebody very long, so I have no problem paying for an hour or so to have you walk me through the Twilio Admin System. I want this done ASAP, since I plan on launching my latest campaign in a day or so. I got the Call forwarding to work perfectly, but I cannot figure out the call recording part. Ideally, I would prefer to hire somebody on an ongoing basis to help me setup multiple campaigns with Twilio. I have a bunch of ideas I want to bounce of an experienced coder.
Skills: VOIP Software Twilio API VOIP Administration
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