FreePBX Jobs

153 were found based on your criteria

  • Fixed-Price – Est. Budget: $50.00 Posted
    We need someone to configure the vTiger-Asterisk API to integrate both the systems. We are using vTiger v6.2 and Freepbx. Under organizations/contacts of vtiger, the call history and call recordings should appear. However this should only show as per user access level. For dial out, once the number is clicked, the hard-phone (cisco) should ring of the user and call should be connected.
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Some phones constantly going offline. I believe it may be due to Firewall issues or...?? I would like to make a script to email alert everytime a phone goes to an UNREACHABLE status. Name/username Host Dyn Forcerport ACL Port Status 100/100 D N A 56642 OK (80 ms) 101 (Unspecified) D N A 0 UNKNOWN 106 (Unspecified) D N A 0 UNKNOWN 201 (Unspecified) D N A 0 UNKNOWN 202 (Unspecified) D N A 0 UNKNOWN 203 (Unspecified ...
  • Hourly – More than 6 months – Less than 10 hrs/week – Posted
    Hi there, Im looking for someone to sort out a few issues with the phone server. At the moment some calls are dropping on pick up (mainly mobile calls) Looking for someone available asap and also some I can work with long turn
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    We have a project to get Mitel phones working with a hosted FreePBX distro. We have the provisioning working and we are able to do everything we need on the phones, except BLF. We can program the button and we even see the successful subscriptions both on the server and on the phones. For some reason however, the phones seem to be "ignoring" the status updates (InUse, Idle, Etc) so the button never lights up. We need someone who has ...
  • Fixed-Price – Est. Budget: $50.00 Posted
    CMC (Corneilius & Merril Corporation) currently uses Voipo for sip trunk and pbx hosting. However, we want to migrate just the pbx hosting from the voipo account to the hostpbx.us asterisk hosting account. This means the person would tell us what needs to login to the voipo pbx account, extract the necessary connectivity information and then login to our hostpbx.us asterisk account and setup the asterisk trunk to receive and send outgoing calls through the voipo sip trunk service ...
  • Hourly – Less than 1 week – 10-30 hrs/week – Posted
    Dear All, We are a Singapore based IP PBX service provider having years of experience in implementing IP PBX's around various regions in the world. We are looking for an expert who has more than 8 years of experience in implementing IP PBX's, who has a thourough knowledge troubleshooting and handling Freepbx End Point Manager/ Digium Phone module etc. You will also have a good knowledge in setting up HTTP/HTTPS/TFTP and FTP servers for provisioning and ...
  • Fixed-Price – Est. Budget: $50.00 Posted
    I've set a VPS server up with CentOS and installed FreePBX but require assistance with the configuration there of and the IP phones that will be connecting to it. Using Grandstream DP715/710 with 2 phones. The server will need to be secured against hacking. In the end the DP715 has to be able to phone in and out with 3 extensions one being a mobile phone. Recording will needs to be activated on all calls and Voicemail enabled ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    1.       Configure a new Polycom IP 6000 phone We have a spare IP6000 phone which we would like to configure. Extension should be 1801. Take note that this extension is already configured on two other phones (Digium phones) and that this should still remain as is. The phone is physically connected to the network. MAC address of the phone is: 0004.f2f6.75e4 2.       Ext 1840. Change name from Fawn Yong to Stephanie Lin (stephanie.lin@rohatyngroup.com). Also configure ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I have a virtual a2billing server setup on a digitalocean cloud server, I am trying to setup a bunch of IP phones coming from one netowrk to the same billing account i created multiple sip accounts all linking to one account code(pin) But when calls go out after a while there is allot of call failed and quality issues and i do not see any reason in the CLI, I am looking for someone to look over the settings ...
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