The OpenStack Infrastructure team maintains a voice conferencing system based on Asterisk. This page documents how to access the system.
The following methods are available for calling the conferencing system:
- PSTN access coming soon!
Once you call into the system, it will ask you for a conference number. We have currently reserved 6000 through 6999 as conference bridges. This can be easily changed as needed.
Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:firstname.lastname@example.org
By default, conference users have the following menu available to them during a conference call:
- * - Play a prompt that describes the menu (then press a digit as described in the prompt)
- *1 - Toggle mute
- *4 - Decrease listening volume
- *6 - Increase listening volume
- *7 - Decrease talking volume
- *8 - Leave conference
- *9 - Increase talking volume
Any SIP soft or hard phone should work fine. The following clients have been tested with the system:
You can find Jitsi here. Download and install it.
Once you have Jitsi running, you must first create an account. Choose the "SIP" account type. Enter a name, but no password. Do not put "@anything" in the name field. This type of account will let you make SIP calls from your computer without registering to any SIP server.