OpenSIPS Jobs

34 were found based on your criteria

  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Hello, Below is our environment: Asterisk version: 11.10 Environment: Amazon EC2 Operating System: Ubuntu 14.04 (64-Bit) Servers (Individual): Asterisk1 (Amazon EC2), Asterisk2 (Amazon EC2 - Backup Server), MySQL1 (Amazon RDS), MySQL2 (Amazon RDS - Backup Server), OpenSIPS (Amazon EC2) Note: All servers are deployed in Amazon VPC. Here, OpenSIPS server will be acted as SIP proxy server and Asterisk will be acted as Media server. Scenario: SIP users table, extension (call rules) table, voice mail tables will be created in ...
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    We are looking for an EXPERT Linux / VoIP Systems Architect who is able to perform an integration for us using either Kamailio or OpenSIPS. The objective of this installation is to gain high CPS for enterprise / call center VoIP Traffic. We were recently notified that the easiest integration would be using OpenSIPS and Freeswitch as our billing/routing engine has existing frameworks to integrate easily with FreeSwitch. More information about this integration is available here: http://docs.jerasoft.net/display ...
  • Hourly – More than 6 months – 10-30 hrs/week – Posted
    We are a company that just had our first switch installed We are looking for someone that has great knowledge of administering and maintaing FreeSwitch, LCR and ASTPP billing. We are wanting to add new modules as we grow and expand. You must be available during US hours and be able to work 4hr days 5 days a week. You must be willing to teach the system as well, as we are eager to learn. We are looking for someone ...
  • Fixed-Price – Est. Budget: $750.00 Posted
    Hello, We require the building of a VoIP Switch which focuses on inbound DIDs. It must handle 1000 concurrent calls. We have our own number range consisting of over 1000 000 DiDs It will have multiple trunks for over 10 providers who will send calls to this switch. The switch must accept the call and RECORD the IP/TRUNK/CHANNEL where the call is originating from. It must also route the call to the relevant SIP Account the DID is ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Bid ONLY if you are extremely skilled with OpenSIPs, Asterisk, RTPProxy, MySQL. Also bid only if you are available to work all day Sunday (your timezone). I had a speed testing application created for me which was housed on my server where users would download a softphone and have it running in their system tray. The user softphone registers to opensips. Every 15 minutes, Asterisk will call this softphone and measure jitter, latency, upload/download speed. The results will be ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I am looking for someone that can help enable our system to write the carrier that terminated the call to the DB in order to pull reports. We are using opensips with radius as well as media-proxy
  • Fixed-Price – Est. Budget: $15.00 Posted
    I would like to clear the following medium vulnerabilities 1. HTTP Server Prone To Slow / Denial Of Service Attack. 2. Apache mod_info /server-info / Information Disclosure Vulnerability 3. Web Server Supports Weak / SSL Encryption Certificates 4. Web Server Supports / Outdated SSLv2 Protocol Need to correct and confirm the vulnerabilities are cleared.
  • Fixed-Price – Est. Budget: $50.00 Posted
    I am looking for a Kamailio Server Expert to resolve the following. Our clients is build with linphone source code. Android and IOS. 1. Not able to do voice and video in 3G/4G 2. Client receiving calls from server after testing. 3.Check on the ports opened and advise what to be opened.
  • Fixed-Price – Est. Budget: $25.00 Posted
    We are developing a mobile apps with chat feature. We are using Kamailio SIP Server and Linphone for Client. Presently, the chat works only in home wifi. It does not work in 3G/4G network and public wifi. We searched the net and found a solution. Need to enable "STUN" in Kamailio Server. I need a Kamailio SIP Server Expert to enable and resolve the problem for us. I can provide VPN details for you to work on.
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I am interested in finding someone to consult with on the configuration of an OpenSIPs installation. We already have it setup and able to make local calls, what i want now is to be able to do dynamic routing to SIP external trunks (some requiring authentication).
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