[Freeswitch-branches] [commit] r11136 - freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax
FreeSWITCH SVN
gmaruzz at freeswitch.org
Mon Jan 12 04:12:05 PST 2009
Author: gmaruzz
Date: Mon Jan 12 06:12:05 2009
New Revision: 11136
Log:
docs
Modified:
freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/README
Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/README
==============================================================================
--- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/README (original)
+++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/README Mon Jan 12 06:12:05 2009
@@ -11,22 +11,38 @@
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-Skypiax is an endpoint (channel driver) that use the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls from/to FreeSWITCH.
+Skypiax is an endpoint (channel driver) that use the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls from/to FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, eg sofia/SIP).
-Think at Skypiax as similar to OpenZAP for analog lines: for each channel you need an interface (a Skype client). So, for eg, for two concurrent calls, you will need two channels, and two Skype clients running on your machine.
+Think at Skypiax as similar to OpenZAP for analog lines: for each channel you need an interface (a Skype client). So, for eg, for two concurrent calls, you will need two channels, two Skype clients running on server.
-On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more important) the snd-dummy ALSA "fake" sound driver.
+If your server's Skype client(s) has got the Skype credits, Skypiax works for SkypeOut calls too.
-On a Linux machine with 3GB ram and a quad core intel6600, we got no problem with 20 concurrent calls, and plenty of room for adding more Skypiax channels.
+You can use it from the dialplan, eg with the provided modified "default.xml" dialplan, you can call "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via the Skype network from a SIP softphone to remote_skypename or to a phone number via SkypeOut, or you can call the "2908" extension from any phone to be bridged to the Skype Test Call).
+
+With the provided skypiax.conf.xml all incoming Skype calls will be routed to the "5000" extension, the IVR in default FreeSWITCH installation.
+
+On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more important) the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. But for a low number of channels it would works with regular X servers and ALSA drivers.
+
+On a Linux machine with 3GB ram and a quad core intel6600, we got no problem with 20 concurrent calls, and plenty of room for adding more Skypiax channels (100? not tested).
On Windows, no need to do anything special, the Skype client is lighter on CPU.
-More info on skypiax: http://wiki.freeswitch.org/wiki/Skypiax
- http://www.celliax.org
+
+''Skypiax is now pre-beta, but usable for testing and finding bugs :-).''
+
+
+'''You can download Skypiax source code with subversion with the command:'''
+
+svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax mod_skypiax
+then, follow the README file in the mod_skypiax directory.
+
+More info on skypiax:
+http://wiki.freeswitch.org/wiki/Skypiax
+http://www.celliax.org
Skypiax has been contributed to the community by:
- Giovanni Maruzzelli
- (gmaruzz at gmail dot com)
+Giovanni Maruzzelli (gmaruzz at gmail dot com)
+
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