VoIP and Skype
Man, we really need the asterisk module for skype. I wonder if anyone’s working on it. Pretty high bountry already.
I’ve already heard about several companies using SkypeOut, especially in Europe. Their prices are very competitive with voipjet, I haven’t checked vs any other companies yet. Not bad, they’re not necessairily the biggest competitor for what I plan on doing, in fact I can use them a lot for stuff, but a module would make things much, much easier. I’m surprised they haven’t gone into releasing one, but I guess they want to keep everything as closed as possible.
Comments
Comment from EuroHacker editor
Time: April 11, 2005, 4:59 pm
One thing I’ve been wanting to do is encrypted Skype, would this be possible with an Asterisk module?
Btw, check out EuroHacker. I’m working on issue 2 right now. If you want to write something, get in touch 
Comment from EuroHacker editor
Time: May 18, 2005, 10:45 am
Was that directed at me?
Comment from admin
Time: May 18, 2005, 11:06 am
No, spam sorry.
Been to busy to clean house/make any new posting. Skype and Asterisk are not compatible right now.
Comment from BriSch
Time: June 14, 2005, 3:52 pm
What about a Skype-Plugin which is able to put Asterisk commands
(like: asterisk -rx dial s or asterisk -rx hangup)?
The easiest way I think is to bridge the “line out” to the “line in” of the sound card - the Alsa or OSS channel must be enabled in Asterisk.
To avoid ‘device busy’ a virtual sound device (/dev/dsp1) has to be
created or a second sound card is needed.
the disadvantages:
- only one person is able to make a phone call (because it is
transferred through the audio cable).
- there is no way to send DTMF signals - voice recognisation must be
set up on the Asterisk.
advantages:
- Skype works with all nesty internet connections
(Proxies, NATs, Firewalls) - so calling the Asterisk PBX
is no problem.
- consumes low bandwidth - so the codec of Skype is used
(Asterisk don’t need any bandwidth because it is connected to the
sound card - extra bandwith is only needed when making a SIP-Call)
Well tested with Asterisk ‘never hangup extension’ and Skype autoanswer
function.
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