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= Asterisk Server (beta) =
= Asterisk Server (beta) =


== Introduction ==
== OBSOLETE ==
Asterisk is a sip complaint VOIP solution which is available (in a beta form) to Fedora developers.  In a way it is an identical solution to the IRC chat rooms except it uses voice content.  It is still in beta but we are happy to share it with everyone until we can obtain proper infrastructure for a production based deployment.


As of 2011 Fedora no longer runs asterisk servers. If at some point in the future a team comes together to do this, this page may not be obsolete any more.


== Introduction ==
Asterisk is a [http://en.wikipedia.org/wiki/Session_Initiation_Protocol SIP] compliant [http://en.wikipedia.org/wiki/Voip VoIP] solution which is available (in a beta form) to Fedora developers.  In a way it is an identical solution to the IRC chat rooms except it uses voice content.  It is still in beta but we are happy to share it with everyone until we can obtain proper infrastructure for a production based deployment.


== Supported clients ==
== Supported clients ==
There are many clients that support SIP based communication.  The clients supported by the infrastructure team are ekiga (Gnome), twinkle (KDE), and linphone.  Other clients will work but troubleshooting will be difficult as the team may be unfamiliar with them.
There are many clients that support SIP based communication.  The clients supported by the infrastructure team are ekiga (Gnome), twinkle (KDE), and linphone.  Other clients will work but troubleshooting will be difficult as the team may be unfamiliar with them.
== Configuring Clients ==
Configuring ekiga or twinkle is easy. If you have still have problem configuring, please follow [[:Infrastructure/Asterisk/Configure_voip_client|this]] link.


== Read Me First! ==
== Read Me First! ==
Asterisk is actually fairly easy to use, your headset and hardware may be a different story.  Before joining ANY conference please make sure to test your equipment.  This is especially true if you are presenting!
Asterisk is actually fairly easy to use, your headset and hardware may be a different story.  Before joining ANY conference please make sure to test your equipment.  This is especially true if you are presenting!


* (./) Do test your connection in <code>sip:infrastructure@fedoraproject.org</code> Ask someone in #fedora-admin to connect and verify your settings.
* {{Template:Important}} Do test your connection in <code>sip:infrastructure@fedoraproject.org</code> Ask someone in #fedora-admin to connect and verify your settings.
* (./) Do use mute when you are not talking (it is preferred you use mute on the sip client and not through other means unless you are confident in your settings.  It is possible for your sound system to loop back into itself.)
* {{Template:Important}} Do use mute when you are not talking (it is preferred you use mute on the sip client and not through other means unless you are confident in your settings.  It is possible for your sound system to loop back into itself.)
* (./) Do pay attention when you join.
* {{Template:Important}} Do pay attention when you join.
* {{Template:Warning}} Do not assume your equipment works
* {{Template:Warning}} Do not assume your equipment works
* {{Template:Warning}} If you cannot connect to <code>sip:infrastructure@fedoraproject.org</code> also try <code>sip:infrastructure@talk.fedoraproject.org</code>
* {{Template:Warning}} If you cannot connect to <code>sip:infrastructure@fedoraproject.org</code> also try <code>sip:infrastructure@talk.fedoraproject.org</code>
== Connecting ==
== Connecting ==
Connecting to the conference is easy.  Just start up the client and join sip:conference@fedoraproject.org.  If you hear a voice asking you to enter your conference number you are all set, just hang up.
Connecting to the conference is easy.  Just start up the client and join sip:conference@fedoraproject.org.  If you hear a voice asking you to enter your conference number you are all set, just hang up.

Latest revision as of 20:08, 16 November 2011

Asterisk Server (beta)

OBSOLETE

As of 2011 Fedora no longer runs asterisk servers. If at some point in the future a team comes together to do this, this page may not be obsolete any more.

Introduction

Asterisk is a SIP compliant VoIP solution which is available (in a beta form) to Fedora developers. In a way it is an identical solution to the IRC chat rooms except it uses voice content. It is still in beta but we are happy to share it with everyone until we can obtain proper infrastructure for a production based deployment.

Supported clients

There are many clients that support SIP based communication. The clients supported by the infrastructure team are ekiga (Gnome), twinkle (KDE), and linphone. Other clients will work but troubleshooting will be difficult as the team may be unfamiliar with them.

Configuring Clients

Configuring ekiga or twinkle is easy. If you have still have problem configuring, please follow this link.

Read Me First!

Asterisk is actually fairly easy to use, your headset and hardware may be a different story. Before joining ANY conference please make sure to test your equipment. This is especially true if you are presenting!

Connecting

Connecting to the conference is easy. Just start up the client and join sip:conference@fedoraproject.org. If you hear a voice asking you to enter your conference number you are all set, just hang up.

Place a call

Once a conference or meeting has been called, contact information will be given. This typically takes the form of infrastructure@fedoraproject.org or sip:infrastructure@fedoraproject.org Just place this information into the call fields in your client and place the call. You should hear a beep for each person that enters or exits a call.

Twinkle (KDE)

File:Infrastructure Asterisk twinkleNotConnected.png

When asked to make a call put it in the "call" field and click on "dial"

Ekiga (Gnome)

File:Infrastructure Asterisk EkigaConnected.png

When asked to make a call put it in the typable field and click on the plug button on the right.

Linphone

It is recommended to use twinkle or ekiga over linphone but some may not have that option.

File:Infrastructure Asterisk linphoneConnected.png

    • Note that with linphone any time you are asked for a call id like infrastructure@fedoraproject.org You need to change the "@fedoraproject.org" to "@talk.fedoraproject.org"

Known Conferences

Troubleshooting

Cannot connect

Being unable to connect is a common issue. Try the following solutions:

  • Disable your firewalls (If it starts working, then you need to re-configure your firewall)
  • Try to connect to @talk.fedoraproject.org instead of @fedoraproject.org
  • Sometimes a client (like ekiga) might bind to an interface it shouldn't. Please verify it is using a valid and proper interface.
    • Ekiga - "Edit -> Preferences -> Protocols -> Network Settings -> "Listen on".
    • Twinkle - "Edit -> System Settings -> General -> Default Network Interface"

Can't hear anything

First ensure that you have your audio configured properly and it isn't muted. Just try to play a song or find some other means to make noise. These issues are typically on the client. To test audio please join:

sip:conferences@fedoraproject.org

You should hear a female voice asking for your conference ID. If you do, just hang up.

No one can hear me

Having a mic configured in properly is a bit of an art. Ensure that you can record your own audio using something like audacity. If you'd like to test your mic, just stop by #fedora-admin on irc.freenode.net and we'll be happy to assist.

Echoing

There are a few common issues that lead to echoing. The most common of which is not using a headset and microphone. Using actual speakers on your laptop or desktop will cause your mic (sometimes embedded in your computer) will pick up the audio again. Unfortunately testing for this yourself is difficult. If you hear someone complaining please mute yourself or drop the call until you can re-configure.

Another common problem that leads to horrible echoing is from your sound system looping back on itself. This issue actually makes the conference unusable. If you join and people start complaining about an echo, please mute yourself or drop the call so those in the meeting can continue.

Ekiga dialpad doesn't work

Try disabling silence detection in your audio configuration. With silence detection enabled, the dialpad may not send tones properly. Furthermore, the ekiga dialpad is prone to sending repeat tones which makes it difficult to enter PINs properly.

Tips

  • People without a mic are encouraged to join meetings and ask questions via irc
  • Those looking to talk or hold meetings on the conference system are encouraged to get at least headphones and a microphone.
  • Use the mute that comes with your client (ekiga, linphone, twinkle, etc). It will be better then muting your mic.
  • Don't forget to disconnect after you're done! No one wants to hear you watching TV that night :)