The IPFire Project has its own VoIP telephony service which is use for direct (and encrypted) communication between developers.

It is possible to call people directly via their extension. For example or from the POTS under +49 2363 6035 XXX where XXX is the extension of each developer.

PBX Features

900 Conference Service
901-909 Shortcuts to conference rooms 1-9
000600... Prefix to dial for Verizon WebEx
000601... Prefix to dial for AT&T WebEx
990 or voicemail Voicemail
998 or park or press *1 during a call Parking Extension
#NNN Agent Login/Logout with NNN being the number of the hotline
Test Services
991 or echo Echo Test Service
992 or music Music

Conference Service

The conference service is available at extension 900. When you call this extension, you will be asked for a conference room number which is either 1-9 or a six-digit number. Please enter the desired conference room number and press #.

The conference service is also available from PSTN as well:

  • Germany: +49 2363 6035 900
  • US: +1 (650) 272-3300

Blind Transfer

You can transfer a call to an other extension by dialing # and the extension.

Parking Calls

You can park calls for up to five minutes by dialing 1 in an ongoing call or by transferring the call to extension 998*. The system will then tell you under which extension the call has been parked and hang up. You can now call that extension from any other phone to continue the call.


The primary protocol we use for telephony is SIP.

Please use the following credentials to register:

Caller name: your real name
Username: your SIP ID
Password: your password
Protocol: TLS or TCP

Your client should automatically find the server to which to connect to by using DNS SRV and NAPTR records. Make sure that this feature is enabled. When you have trouble with your client, you may optionally use as a SIP proxy.

You may register multiple clients at a time, which will all ring when you are being called.

Secure communication

For secure communication with the server, you can use SIP over TLS (TCP/5061) to encrypt the SIP messages and SRTP for encrypting the RTP streams. Only very few clients support these features at this time.

Supported Codecs

The SIP server transcodes calls when necessary and supports the following codecs in this order of preference:

  • Opus
  • G.722
  • G.711 (alaw)
  • G.711 (ulaw)
Edit Page ‐ Yes, you can edit!

Older Revisions • August 18 at 11:26 pm • Jon