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< Infrastructure
Revision as of 18:15, 22 July 2013 by Russellb (talk | contribs) (Calling)

Conferencing System

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:

  1. sip:s@pbx.openstack.org
  2. PSTN: +1 512-808-5750

Conference Numbers

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:6000@pbx.openstack.org

In-call menu

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

SIP clients

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.