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Difference between revisions of "Asterisk"
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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. | ||
Latest revision as of 17:09, 24 November 2017
NOTE: Asterisk is being replaced by service from Zoom.us. The service will be shutdown Jan 5th, 2018
Contents
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.
Eclipse projects can request a dedicated bridge by opening a bug against Community/Servers.
How do I access it?
Telephone
We have numbers in Canada, North America (toll-free), Germany, France, and the UK (toll-free). The European numbers are a local call anywhere in that country. Calling these numbers from any phone or VoIP phone will enable you to join a conference call.
See a list of Phone Numbers.
Your project or IWG leader will provide the bridge 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)
Some clients require you to register/authenticate against a system. If you don't already have voip, you may need an account somewhere such as iptel.org - iptel.org is completely free of charge and has successfully been used with Jitsi on Windows (account comes preconfigured).
SIP clients we've used successfully:
- Windows, Linux, MacOS: Jitsi
- Linux twinkle - it doesn't require you authenticate against some VoIP system like many other clients which is handy.
- Android: sipdroid
- iPhone: 3cx
- Hardware VOIP phones: SNOM, Digium
Conference commands
The following commands are available when connected to a conference call.
*1 to mute/unmute yourself *2 lock/unlock 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 volume *9 increase your volume volume *8 exit the conference
Problems?
Should you detect any issues with Asterisk, please raise a bug in Bugzilla to Community/Servers.
If you need a bridge for your project, please request it via the same Bugzilla coordinates.