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.
|901-909||Shortcuts to conference rooms 1-9|
|991||Echo Test Service|
|000600...||Prefix to dial for Verizon WebEx|
|000601...||Prefix to dial for AT&T WebEx|
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
You can transfer a call to an other extension by dialing # and the extension.
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.
The primary protocol we use for telephony is SIP.
Please use the following credentials to register:
|Caller name:||your real name|
|Username:||your UID - 900|
|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.
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.
The SIP server transcodes calls when necessary and supports the following codecs in this order of preference: