1. SIP Phones
    1. How to set paging function on the phone side
    2. SIP softphone we recommend
    3. Hard Phones
      1. ClearOne MAX IP Conferencing Phone
      2. Grandstream Networks Budge Tone - 100 Phone
      3. inter-tel
      4. ipDialog SIP Tone III Phone
      5. ipDialog SipTone™ V Phone
      6. Linksys IP Phone
      7. Nokia Phone
      8. Panasonic KX-TGP550 T04 DECT
      9. Panasonic KX-UT1xx
        1. Configure shared line in Panasonic KX-UT1xx
        2. Set up Flexible Button on Panasonic KX-UT1xx
      10. Polycom Kirk DECT
      11. Polycom SoundPoint IP 650/330 Phone
      12. Polycom SpectraLink 8020 Wireless Telephone
      13. Polycom VVX600 Business Media Phone
      14. Polycom VVX Intercom feature
        1. Intercom with provisioning
      15. SpectraLink 8400 Wireless Telephone
      16. SAXA IP Netphone SX Phone
      17. Snom m3 VoIP wireless DECT phone
      18. Snom Phones
      19. Swissvoice IP10S Phone
      20. Ubiquiti UniFi VoIP Phones
      21. Vtech Business Phones
      22. Yealink SIP Phones
      23. ZyXEL Prestige 200w
    4. Soft Phones
      1. Age Phone
      2. Bria
      3. Kapanga
      4. X-Lite
  2. VoIP SIP Gateways
    1. AddPac AP1200
    2. AudioCodes Mediant 1000(FXO/FXS)
    3. AudioCodes MP-104 (FXO)
    4. AudioCodes MP-114 (FXO/FXS)
    5. AudioCodecs TP-260
    6. Cisco Systems ATA186
    7. Grandstream Networks HondyTone 486
    8. Mediatrix 1104(FXS)
    9. Linksys SPA-3000
    10. Mediatrix 1204 (FXO)
    11. Medicatrix 2102 (FXS)
    12. Quintum Tenor ASM200
    13. Quintum Tenor DX Gateway
    14. Sangoma NetBorder Express Gateway
    15. TAINET - Venus 2908
    16. Yamaha RTV 700
    17. Yamaha RTX 1000
    18. ZyXEL Prestige 2602 HW
  3. ITSP
    1. BroadVoice
    2. DIDx
    3. iConnectHere
    4. InPhonex
    5. Megapath
    6. sipgate
    7. Skype
    8. Triad Telecom
    9. Virtual GlobalPhone
    11. Voxbone
    12. VoIPUSER
    13. Vonage
  4. WebRTC client
    1. JsSIP
  5. Misc
    1. Application-Level Gateway(ALG)
      1. InGate SIParator 4.6.1
        1. Standalone SIParator, with SIP server on the outside
        2. DMZ SIParator, with Brekeke SIP Server on the outside
    2. Ascom AA60
    3. Avaya PBX (CM 5.2)
    4. CyberData VoIP Ceiling Speaker
    5. CyberData VoIP Intercom
    6. DTH VoIP Billing
    7. Microsoft Lync
    8. Microsoft Lync or OCS
      1. Brekeke SIP Server and Lync or OCS
      2. Brekeke PBX and Microsoft Lync or OCS
        1. Configure Brekeke PBX for Lync or OCS
        2. Configure Lync or OCS for Brekeke PBX
    9. Microsoft Speech Server
    10. Mobotix T24M-Sec-D11 Hemispheric IP Video Door Station
    11. Nortel CS1000
    12. Radius Server
      1. Jerasoft Development - JeraSoft VCS
      2. Jerasoft Development's VCS Dynamic Routing
      3. Clearbox Radius
      4. FreeRADIUS
    13. Jeron's Nurse Call System - Provider 790
    14. Rauland's Nurse Call System - Responder 5
    15. Sangoma NetBorder Call Analyzer



This manual describes the quick setup of FreeRadius Server 2.1.10 to work with Brekeke SIP Server (Advanced edition) Authentication and Accounting. If you have more questions about detailed configurations about FreeRadius, please contact FreeRadius website:


Set up and Configure FreeRadius

1. Build and install FreeRadius

Download FreeRadius from

$ tar zxvf freeradius-[version].tar.gz 
$ ./configure 
$ make 
$ su - root 
$ make install  

2. Configure FreeRadius

Define radius client – Brekeke SIP Server at FreeRadius server

$vi /usr/local/etc/raddb/clients.conf

add the following lines at the beginning of the file

secret = testing123
shortname =

The above setting sample defined radius clients (Brekeke SIP Server) is at IP
and set “secret” as “testing123”
The same secret string should be set at Brekeke SIP Server side, check below.


Define authentication type

$more /usr/local/etc/raddb/sites-available/default

check if the line include “digest” in authorize is uncommended.
if not, edit file to uncomment the line with “digest”

Add User Authentication Accounts

$vi /usr/local/etc/raddb/users

add the following line of text at the top, before anything else.

100 Cleartext-Password := "test"
Reply-Message = "Authenticated REGISTER request for Brekeke SIP Server"
200 Cleartext-Password := "test"
Reply-Message = "Authenticated REGISTER request for Brekeke SIP Server"

With the above sample, user 100 and 200 has been added and their authenticaiton password is “test”


3. Run FreeRadius server as debug

$ radiusd -X
Brekeke SIP Server Setup

In this example, we use Brekeke SIP Server advanced edition, which already includes the radius setup in the admintool

1. At Brekeke SIP Server admintool, [Configuration] > [SIP] >[Authentication]
Set ON at authentication for both [REGISTER] and [INVITE] 
Save the setting

2. At Brekeke SIP Server admintool, [Configuration] > [Database/Radius] > [Radius],
set up Radius as below, and save settings

[On/Off (Authentication)]: on
[Port Number(Authentication)]: 1812
[Port Number(Accounting)]: 1813
[Server IP Address]: (set in step2 "Define radius client"at freeRadius setup)
[Shared Secret]: testing123 (set in step2 "Define radius client" at freeRadius setup)

3. Add the following dial plan to use Radius account plugin for each call

[Matching Patterns]
$request = ^INVITE

[Deploy Patterns]
$session = RadiusAcct
$continue = true

For Brekeke SIP Server v2, set Deploy Patterns as following:

[Deploy Patterns]
$session = plugin.radius.RadiusAcct
$continue = true

Click [Apply Rules] button after adding the dial plan

4. If realm is not necessary for the Radius Attribute User-Name,
please add the following parameter at Brekeke SIP Server admintool > [Configuration] > [Advanced]

radius.addrealmtouser = false

5. Restart Brekeke SIP Server from admintool when finishing above setup

User Agent Setup

At phone side, set Brekeke SIP Server IP as registar/proxy server
Set each phone userID, Authentication ID and password the same value as what are set when configuring FreeRadius step2 “Add User Authentication Accounts”

Yes No
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