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Telephony

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 sip:XXX@ipfire.org 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
980 Parking Extension
990 Voicemail
991 Echo Test Service
992 Music
993 Time
999 Emergency Services
891/890 Agent Login/Logout
000600... Prefix to dial for Verizon WebEx
000601... Prefix to dial for AT&T WebEx

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: +49 2363 6035 900

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 # in an ongoing call and then transfer the call to extension 980. 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.

SIP

The primary protocol we use for telephony is SIP.

Please use the following credentials to register:

Registrar: ipfire.org
Caller name: your real name
Username: your UID - 900
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 sip.ipfire.org 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:

Audio

  • Opus
  • Silk
  • Speex
  • G.722/G722.1
  • G.711 (alaw)
  • G.711 (ulaw)
  • GSM
  • Codec2

Video

  • H.264
  • VP9
  • VP8
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Older Revisions • December 10 at 1:28 pm • Michael Tremer