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Sip Jobs

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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: More than 6 months, 30+ hrs/week - Posted
The TCP/IP VOIP Engineer will be providing functional and practical analysis related to the installation of the communication networks. The TCP/IP VOIP Engineer will be given the responsibility of Designing, maintaining and improving voice and telephony infrastructures and ensures that all are secure, available (99+% minimum uptime) and functioning efficiently. You will be able to ultilize your knowledge of commonly used communication concepts, practices and procedures to diagnose and resolve moderately complex communication issues. Coordinates strategies for defining, deploying and maintaining the companies voice and video communication architecture and its associated network connections and component hardware. The TCP/IP VOIP Engineer also manages all engineering projects for voice and video initiatives, planning technology roadmaps and configuring and optimizing all PBx telephone system and services.
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...
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...
Hourly - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I have a SIP trunk from my ISP. They have provisioned a second port on their handoff switch for the SIP traffic. I only have one LAN card on the server. I have a sonicwall firewall. I want some help with setting up the Class A private IP on X3. I also need some help with my NAT policies so that the Class C private IP that the Server uses knows to route calls passing the Class A address as the source and passing all replies from Class A to Class C.
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...