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[Michael Jackson](//users/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 |
| 000600... | Prefix to dial for Verizon WebEx |
| 000601... | Prefix to dial for AT&T WebEx |
| Services | |
| 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.
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 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 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:
- Opus
- G.722
- G.711 (alaw)
- G.711 (ulaw)