[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

[trixter aka Bret McDanel]

How many channels have you guys been able to get with this?  

The only problem I have with this is that it takes skype and a soundcard
(virtual or otherwise) and the "API" is really executing commands on a
running skype process.  In my opinion its not worth it for 1 concurrent
call per account.

I have written code that works with skype in linux that simulates a
virtual sound device.  I have used that and successfully done calls out
with this.  I havent played with the dbus stuff (how you control the
skype app from within linux) but since I have a "soundcard" that I know
the audio format of it wouldnt be difficult to integrate this into
asterisk, I could tweak chan_oss and make it into chan_skype fairly
easily since that takes care of the other half of the equation.  The
only thing missing would be the events via dbus, which there are plenty
of examples on so its not like all new code would have to be written.

But its just not worth it if you have to have skype running for each
call.  And then you would potentially have to have a new username for
each running process, and skype really wants X on linux so you would
have to at least have the X virtual frame buffer (it works and acts like
X but never displays anything or uses any hardware).  That seems like an
aweful lot of wasted resources on a box to connect to skype.


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
Belfast IE +44 28 9099 6461    DE +49 801 777 555 3402
Utrecht NL +31 306 553058      US WA +1 360 207 0479
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http://www.trxtel.com the VoIP provider that pays you!
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