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Difference between revisions of "Asterisk"
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The system was implemented in 2012 to reduce costs and offer greater flexibility. | The system was implemented in 2012 to reduce costs and offer greater flexibility. | ||
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| + | Eclipse projects can request a dedicated bridge by opening a bug against [https://bugs.eclipse.org/bugs/enter_bug.cgi?product=Community&component=Servers Community/Servers]. | ||
==How do I access it?== | ==How do I access it?== | ||
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===Telephone=== | ===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. Your project or IWG leader will provide the | + | 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. |
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| + | See a list of [http://wiki.eclipse.org/Asterisk/Numbers Phone Numbers]. | ||
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| + | Your project or IWG leader will provide the bridge and passcode as part of the invitation to a meeting. | ||
===SIP=== | ===SIP=== | ||
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[http://en.wikipedia.org/wiki/List_of_SIP_software#Clients See a list of clients]. | [http://en.wikipedia.org/wiki/List_of_SIP_software#Clients See a list of clients]. | ||
| − | + | 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 [http://www.iptel.org/service iptel.org] - iptel.org is completely free of charge and has successfully been used with '''Jitsi''' on Windows (account comes preconfigured). | |
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| − | + | SIP clients we've used successfully: | |
| + | * '''Windows, Linux, MacOS''': [http://download.jitsi.org/ Jitsi] | ||
| + | * '''Linux''' [http://www.twinklephone.com/ twinkle] - it doesn't require you authenticate against some VoIP system like many other clients which is handy. | ||
| + | * '''Android''': [http://sipdroid.org/ sipdroid] | ||
| + | * '''iPhone''': [http://www.3cx.com/products/3CXPhone-for-iPhone.html 3cx] <!-- [http://www.sipgate.de/team/apps/iphone sipgate] --> | ||
| + | * '''Hardware VOIP phones''': [http://www.snom.com/ SNOM], [http://www.digium.com Digium] | ||
==Conference commands== | ==Conference commands== | ||
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==Problems?== | ==Problems?== | ||
| − | Should you detect any issues with Asterisk, please raise a bug in Bugzilla to | + | Should you detect any issues with Asterisk, please raise a bug in Bugzilla to [https://bugs.eclipse.org/bugs/enter_bug.cgi?product=Community&component=Servers Community/Servers]. |
| − | If you need a bridge for your project, please request it via | + | If you need a bridge for your project, please request it via the same Bugzilla coordinates. |
Revision as of 15:15, 31 July 2012
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.