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Voip Software Jobs

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Fixed-Price - Est. Budget: $ 150 Posted
1. Setup the security and call quality settings correctly. 2. isymphony complete setup 3. incoming call diversion to operator logging, (may have 2 operators) 4. correctly setup/verify approx. 10 extensions and 2 pstn connection 5. call recording schedule into dropbox or Justhost folder 6. make sure pabx run smoothly without having any security issues 7. change ivr menu options and hold on music options 8. setup on demand recording individual files to as user option to email or downloaded 9. help to setup headset’s as optimum use conditions & rectify current issues with existing trunks 10. Given full tanning including; Free PABX commands essentials Setup and use of isymphony Setup new extentions call recording Managing backup’s
Fixed-Price - Est. Budget: $ 100 Posted
This is a project for me as an individual, not a company. If the budget seems insufficient I'm ready to reconsider. Objective Limit to only some actions the usage of the phone for my mother without giving the impression that someone is limiting her freedom or doubting her mental abilities, while allowing full use of the phone for the helper present in the house. Configuration Device: Obihai OBI110 LINE: connected to a standard PSTN line in France. Ethernet: connected to a ADSL router inside the house Phone: connected to a standard DECT phone ObiTalk activated SIP Account: managed on CallCentric.com with a specific French SIP number (different from the PSTN line) The Obihai device is currently with me in the US (West Coast) for configuration and testing. As soon as it is fully configured, I will send it to Paris to be plugged in by the helper into the Line, the router and the DECT phone. It should work immediately out of the box. More Context The primary user (my...
Hourly - Est. Time: 3 to 6 months, 10-30 hrs/week - Posted
I have an app that I'd like to add VOIP functionality to. Basic functions would be obtaining phone numbers and forwarding them to other numbers. Infrastructure would be in the cloud - likely AWS. I'm a developer myself but new to VOIP so would like to discuss architecture plan first and then proceed from there in phase 2 if all looks good.
Hourly - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
My company does mostly PRESS-1-SURVEY We will also be doing some outbound campaigns. I need some one who has max experience with these dialer systems. I need you to help my team in configuring it and give ongoing support in my Vicidial/Goautodialonly experts are welcome to send their proposals. If you apply for this job, please put your favorite M&M color in your reply.
Fixed-Price - Est. Budget: $ 100 Posted
Tehnology we use: -Asterisk -FreePBX -WebRTC (Tried with Sip5ML, JsSip, Sip.js) - 4 different trunks - One stationary number (our ISP) - 4 GSM numbers Problem we have: When calling trough stationary trunk on stationary trunk we get sound delay when answering late (Delay is not always the same it depends on the time when user pick up the call) Explanation: Ok so we start off with what is working. If we make a phone call through GSM trunks on any number (GSM or Stationary) everything works fine. The important thing here is that when connection is established and call starts to progress we get response 183 (With early media) and so on... The 183 response is sent by our GSM VoIP terminal. The problems appear when calling through our ISP trunk, since provider can't fake 183 response if it is not sent to them from dialed number. When we call through ISP trunk on GSM numbers things work fine, when connection is established GSM providers send 183 response with early media...