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Difference between revisions of "Asterisk"

(SIP)
(SIP)
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SIP clients we've used successfully:
 
SIP clients we've used successfully:
 
* '''Platform independent''': [http://download.jitsi.org/ Jitsi]
 
* '''Platform independent''': [http://download.jitsi.org/ Jitsi]
* '''Linux''' [http://www.twinklephone.com/ twinkle] - it doesn't require you authenticate against some system like many other clients which is handy.  
+
* '''Linux''' [http://www.twinklephone.com/ twinkle] - it doesn't require you authenticate against some VoIP system like many other clients which is handy.  
 
* '''Windows''': [http://www.counterpath.com/x-lite.html X-Lite], [http://www.3cx.com/products/3CXPhone-for-iPhone.html 3cx],  [http://www.sipgate.de/team/apps/iphone sipgate]
 
* '''Windows''': [http://www.counterpath.com/x-lite.html X-Lite], [http://www.3cx.com/products/3CXPhone-for-iPhone.html 3cx],  [http://www.sipgate.de/team/apps/iphone sipgate]
 
* '''MacOS''': TBD
 
* '''MacOS''': TBD

Revision as of 09:26, 13 April 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 (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 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)

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 iptel.org.

SIP clients we've used successfully:

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 Eclipse Foundation => Community => Servers.

If you need a bridge for your project, please request it via. the same Bugzilla coordinates.