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Difference between revisions of "Asterisk"
(New page: ==What is Asterisk?== [http://www.asterisk.org/ Asterisk] is an open source, free soft switch/soft PBX system for managing all manner of telephony needs. The Eclipse Foundation is current...) |
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==What is Asterisk?== | ==What is Asterisk?== | ||
[http://www.asterisk.org/ Asterisk] is an open source, free soft switch/soft PBX system for managing all manner of telephony needs. The Eclipse Foundation is currently using it for a voice-conferencing solution. | [http://www.asterisk.org/ Asterisk] is an open source, free soft switch/soft PBX system for managing all manner of telephony needs. The Eclipse Foundation is currently using it for a voice-conferencing solution. | ||
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===Telephone=== | ===Telephone=== | ||
| − | We have numbers in Canada, North America (toll-free), Germany, France, and the UK. Calling these numbers from any phone or VoIP phone will enable you to join a conference call. | + | We have numbers in Canada, North America (toll-free), Germany, France, and the UK. Calling these numbers from any phone or VoIP phone will enable you to join a conference call. Your project or IWG leader will provide the number to call and passcode as part of the invitation to a meeting. |
===SIP=== | ===SIP=== | ||
| − | SIP is a popular VoIP protocol. We allow | + | SIP is a popular VoIP protocol. We allow incoming SIP calls from around the world. These calls are free although they do require Internet access. To call an extension, you would use the following syntax in your SIP client: |
<i>extension</i>@asterisk.eclipse.org | <i>extension</i>@asterisk.eclipse.org | ||
| − | + | (replace extension with the extension you wish to reach) | |
==Conference commands== | ==Conference commands== | ||
Revision as of 12:33, 6 March 2012
What is Asterisk?
Asterisk is an open source, free soft switch/soft PBX system for managing all manner of telephony needs. The Eclipse Foundation is currently using it for a voice-conferencing solution.
The system was implemented in 2012 to reduce costs and offer greater flexibility.
How do I access it?
Telephone
We have numbers in Canada, North America (toll-free), Germany, France, and the UK. Calling these numbers from any phone or VoIP phone will enable you to join a conference call. Your project or IWG leader will provide the number to call and passcode as part of the invitation to a meeting.
SIP
SIP is a popular VoIP protocol. We allow incoming SIP calls from around the world. These calls are free although they do require Internet access. To call an extension, you would use the following syntax in your SIP client:
extension@asterisk.eclipse.org (replace extension with the extension you wish to reach)
Conference commands
The following commands are available when connected to a conference call.
*1 to mute/unmute yourself *2 lock/unluck the conference (moderator only) *3 eject the last person to join (moderator only) *4 decrease conference volume *6 increase conference volume *7 decrease your volume voume *9 increase your volume voume *8 exit the conference