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Sip Jobs

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Fixed-Price - Est. Budget: $ 60 Posted
I wish to hire someone with good communication skills who can show me how to do a few things on my Grandstream UCM6102. I am quite knowledgable in IT in general - I just don’t have time to play around with the UCM6102 and discover how to set the device up myself. I expect a well prepared tutoring session over Skype and Teamviewer. I foresee the initial part of this contract to involve up to 2 hours tutorial. However, I imagine requesting the support of the VoIP expert sporadically in the future when troubleshooting issues or wishing to add new features. As the UCM6102 can only be accessed on our local network, I am happy to let the tutor teamviewer into my computer and look around the UCM6102 configuration before delivering to me the tutorial. What I would like to set up: Path 1 - office hours (Mon-Fri, 10am-10:45pm; Sat, closed; Sunday 8:30am-6pm) When someone calls, a welcome message played to everyone (OHGreeting). Then all phones ring (Hold). If not picked up, goes to...
Fixed-Price - Est. Budget: $ 2,000 Posted
We require a custom PTT mobile client for Iphone and Android. This will use our existing SIP server using SIP and an API we will provide. It will consist of the following features: 1. Run in background 2. User to login and remember the login details. 3. Group Selection (This will basically be an extension that will be called and setup for PTT) 4. PTT button 5. Volume Control 6. Mute 7. Will always use Speaker Mode 8. Ability to allocate a certain hardware button for PTT regardless of if the app is running 9. App will require branding and design. Given that we are targeting both android and Iphone it would be preferred if we can use xamarin.
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...
Fixed-Price - Est. Budget: $ 1,000 Posted
Configure a windows server WS2012R2 server for using Dialogic D/4PCIU cards, PCI version. Note that this is 4 lines analog card. We have a legacy system that uses the D/4PCIU via TAPI on windows server WS2003 32 bits Dialogic ended TAPI support with SR511SP1, which only runs “out of the box” up to windows server WS2003 32 bits We want to convert our legacy system to WS2012R2 The goal of this project is: PREFERRED: Enable the D/4PCIU to work with windows server 2012R2 TAPI This could be done configuring dialogic software, if available, or by coding an interface. Any interface must be compatible with visual studio 2013 (C#) ALTERNATE: Enable the D/4PCIU to work with windows server 2012R2 SIP ** (note difference with previous one) This could be done configuring dialogic software, if available, or by coding an interface. Any interface must be compatible with visual studio 2013 (C#) NOTE: We already have SIP to TAPI, that's why this may work for us, but we prefers the...
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...
Fixed-Price - Est. Budget: $ 150 Posted
We are looking for someone who can do the below mentioned tasks for us. 1- Build a website using our given template, and integrate it to Mysql database of our voip server. 1- Website must have signup and member/reseller portals, you can use the pre-designed portals provided by VoiP Vendor and modify them according to website design as well. 2- Integrate user and reseller to add payment using paypal,skrill,ecoPayz,and credit card. and in case of bank wire user can add transaction ID about his payment for verification. 3- Vendor API for DID Numbers must be integrated, User can Buy DID numbers from Inventory as well as from vendor push/pull queries. 4- Email must be configured to sent on different events (Signup/Email Confirmation, Balance Low, Balance Added, DID Purchase, DID Renew, DID Expire, DID Billing Coming Soon, Maintenance email to all users etc.) 5- Ability to order VPS from the portal, and can choose from predefined vps hardware, VPS price must increase/decrease...
Fixed-Price - Est. Budget: $ 400 Posted
Evaluating the performance of Voip in congested LTE network, by using parameters like Packet loss and End to end delay. the LTE network should be congested with services like FTP, HTTP and email services while the voip service is also being run. Also the performance of voip in non congested network will be evaulated, so as to be able to compare and contrast the reliabilty of voip when there is congestion and where there is not. Also codecs that can be used to improve voip whilst the lte network is being congested. the simulator tool i would like to use is OPNET. thank you
Fixed-Price - Est. Budget: $ 150 Posted
We are looking for someone who can do the below mentioned tasks for us. 1- Build a website using our given template, and integrate it to Mysql database of our voip server. 1- Website must have signup and member/reseller portals, you can use the pre-designed portals provided by VoiP Vendor and modify them according to website design as well. 2- Integrate user and reseller to add payment using paypal,skrill,ecoPayz,and credit card. and in case of bank wire user can add transaction ID about his payment for verification. 3- Vendor API for DID Numbers must be integrated, User can Buy DID numbers from Inventory as well as from vendor push/pull queries. 4- Email must be configured to sent on different events (Signup/Email Confirmation, Balance Low, Balance Added, DID Purchase, DID Renew, DID Expire, DID Billing Coming Soon, Maintenance email to all users etc.) 5- Ability to order VPS from the portal, and can choose from predefined vps hardware, VPS price must increase/decrease...