Difference between revisions of "Infrastructure/Conferencing"
(→Conferencing System) |
(→Calling) |
||
Line 7: | Line 7: | ||
The following methods are available for calling the conferencing system: | The following methods are available for calling the conferencing system: | ||
# sip:pbx.openstack.org | # sip:pbx.openstack.org | ||
− | # PSTN | + | # PSTN: +1 512-808-5750 |
== Conference Numbers == | == Conference Numbers == |
Revision as of 16:57, 22 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: +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
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.