Difference between revisions of "Infrastructure/Conferencing"
(Created page with "= Conferencing System = The OpenStack Infrastructure team maintains a voice conferencing system based on [http://www.asterisk.org Asterisk]. This page documents how to acces...") |
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Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:6000@pbx.openstack.org | Alternatively, if you are calling in via SIP, you can call a bridge directly: sip:6000@pbx.openstack.org | ||
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| + | == In-call menu == | ||
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| + | 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 == | == SIP clients == | ||
Revision as of 20:10, 18 July 2013
Contents
Conferencing System
The OpenStack Infrastructure team maintains a voice conferencing system based on Asterisk. This page documents how to access the system.
Calling
The following methods are available for calling the conferencing system:
- sip:pbx.openstack.org
- PSTN access coming soon!
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
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:
Jitsi
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.
