This page is obsolete. For complete usage instructions for configuring client softphones and IP phones with Fedora Talk, refer to this page.

Using Fedora Talk

After access has been setup, all that is left is to setup your client

Supported clients

There are many clients that support SIP based communication. The clients supported by the infrastructure team are ekiga (Gnome), twinkle (KDE), and linphone. Other clients will work but troubleshooting will be difficult as the team may be unfamiliar with them.

Settings In Brief

The proxy server, outbound proxy, and registration server are all set to The port on all three is 5060. There is no STUN server so far though should work. Your login is the FAS login.

Setting Up A Soft Phone


There is a useful video tutorial for configuring Twinkle with Fedora Talk. Useful content may be added to the Fedora Project wiki, possibly even by you.


  1. Open Ekiga. Cancel the druid (if any). You will get a screen like this.
    Ekiga main screen
  2. Goto Menu → Edit → Accounts. You have this.
    Ekiga accounts window
  3. Edit and fill the details
    Ekiga edit accounts screen
    • Name: Fill in whatever name this account should have in the list.
    • Registrar:
    • User: Your FAS username — this is what you use to log into the wiki or the Fedora Accounts System.
    • Authentication User: leave this empty
    • Password: The password you set in the VoIP section in FAS. (Note that this not necessarily the same as your FAS password)
    • Timeout: Set to whatever feels right to you. Otherwise just leave the default.
    • Enable account: Be sure this is checked!
  4. When selecting "OK", you will get back the account screen. You can check if everything worked out well by checking the status in the "Status" column.
    Ekiga completed accounts window
  5. You are done. You can enter sip:extension or to call. You can find your extension in the VoIP section of FAS.

Testing the setup

Perhaps the most difficult aspect of VOIP for new users is troubleshooting issues. Please see troubleshooting for more help. The best way to test is to call one of the actual dial in numbers listed on the main page and dial your own extension. Users will be able to hear what they sound like on both ends and determine if there is an echo, static, or other issue.