Voip Jobs

69 were found based on your criteria

  • Hourly – More than 6 months – 30+ hrs/week – Posted
    I'm looking for a webrtc expert that has some knowledge of node.js to help debug our WebRTC implementation.
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Looking for some help to configure Asterisk server to support MGCP-NCS protocols for multiple eMTA devices.
  • Hourly – Less than 1 month – 10-30 hrs/week – Posted
    Feature Code script to enable end users to dial a feature code followed by an account code to be written to the Asterisk CDR's for billing. Ie. *11987545*Phone Number where *11 is the feature code, 987545 is the account code, followed by the phone number they wish to dial.
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    We have a phone system that was setup by a employee that is no longer with us and left little documentation. I need to add an emergency line and learn how to change our Voicemails. We are using Vitelity and PBX in a flash.
  • Hourly – 1 to 3 months – 10-30 hrs/week – Posted
    We need help integrating Plivo. We're having issues with calls not connecting properly. We can hear the other side of the line but the person can't hear us for 2-3 seconds. Please message on skype. Thanks! Rob
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    We are looking for freelancers to our team to do further develop of existing Kamailio setups with new features special for the needs we have like smsc ect. Please contact me and tell a little about yourself and your practical knowledge and experience with Kamailio. There will be ongoing tasks for the right person and the opportunity to become a permanent part of our international network.
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I am running elastix but would like to move over to FreePBX. I made an image of my elastix setup and upgraded the freepbx part as far as i could i saved that backup and put a clean version of FreePBX on and loaded the backup. I still need things looked at: 1. Fix sip_nat errors. 2. Clean out the outbound routes. Use the 1 route and trunk for all external calls. 3. Ensure that the contexts for Company-A and ...
  • Hourly – More than 6 months – Less than 10 hrs/week – Posted
    We are a french company specialized in e-learning by video conference. We use Adobe Connect. We want a freelance or a company to assure the technical support for our US teachers. Your first mission: -Resolve sound trouble with one of our teacher You will be contacted for other similar missions in the futur. Your profil: - experienced in video conference technology ......................................................; French translation: Nous sommes une entreprise française spécialisée dans l'enseignement à distance. Nous utilisons Adobe Connect. Nous recherchons un ...
  • 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