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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...
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: $ 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: $ 100 Posted
This is a software specific project. I need someone who's an expert on Newfies dialer to do the configuration for me. I need this software to be setup with flowroute and make outbound press 1 campaigns calls. The software is installed on a Debian server the software is already installed, just need to be configured to run press 1 campaigns. May need to setup a basic cron job to delete stored data every month or 2. Please do not bid on this project if you have not worked with Newfies dialer before. Thanks
Fixed-Price - Est. Budget: $ 100 Posted
i am using asterisk 1.8. recently found a problem. i have 9 trunks. every trunk have 16 gsm port. my carrier send his call using "77761XXXXX" prefix. i need to distribute all calls equally using 1 (first) priority. So i need to distribute all calls like round-robin and equally. Thanks.
Fixed-Price - Est. Budget: $ 1,000 Posted
I am looking to develop a website in odoo with integrated voip and 2 way sms to confirm appointments that have been booked through the website and directly via CRM area of odoo. The site must list products for sale and calculate costs for postage to display final cost. I have created a demo site of what I am looking for and have attached a screen shot of the links on the site.