asterisk users - Jun 2006 - AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

Asterisk handling My Skype Calls

This is for me, once more, Asterisk as the Future of Telephony.

Today I've integrated my Skype Account as SIP extension in my * Box.
This has been possible using "Uplink Skype to SIP Adapter", available
for free at http://www.nch.com.au/skypetosip/index.html .

Main features that any one can easily integrate into Asterisk:

- Route skype incoming calls into Asterisk DialPlan, then you just can
do ANYThing route to your mobile, Meetme rooms, IVRs do it in your
way.

- Dialout calls from any SIP extension through Skype (reaching Skype
contacts or outgoing calls to landline through Skype Outgoing calls
prices.

- Enable your website with SkypeMe Button and route it to Asterisk!
Feel free to listen MusicOnHold from my Asterisk Box through my Skype
Account.

Check this in http://asteriskpt.blogspot.com - AsteriskPT - Asterisk
Portuguese Users Group.

Please feel free to contact me if you have more ideas to improve this
solution, currently i didn't test more than one simultaneous calls
incoming and outgoing through Skype.

MoutaPT

http://asteriskpt.blogspot.com - AsteriskPT - Asterisk Portuguese Users Group.

>>
>> Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
>> Engels praten!
>>  ;-)
>>
>
> Pues my punto fue que un poquito de correo en otro idioma no hace 
> da?o, y si ayuda mucho y molesta poco, ?por qu? quejarse?
Quel bordel, sacrebleu!

-- 
Jean-Michel Hiver - http://ykoz.net/
D?couvrez la R?union des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Ok, on peut parler fran?ais alors ;)

Olivier

Jean-Michel Hiver a ?crit :
>
>>>
>>> Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij
aardig
>>> Engels praten!
>>>  ;-)
>>>
>>
>> Pues my punto fue que un poquito de correo en otro idioma no hace 
>> da?o, y si ayuda mucho y molesta poco, ?por qu? quejarse?
>
> Quel bordel, sacrebleu!
>

Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port.  Run Linux off a CF card and have it setup to *only*
interface with Skype and Asterisk.  Basically, make a Skype ATA, but it would
convert Skype to SIP.  I think that could still be considered an ATA, right?
 Or a gateway at least.

Since you can make a Skype account for free and
can (for right now) make US and Canada LD calls for free, I think the cost
and time to make them would be worth it.  :)  And if you figure out a good
price for them, people might even buy them from you....

Undrhil

---
Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com
wrote:
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
> US NY +1 516 687 5200          FreeWorldDialup: 635378
> http://www.trxtel.com
the VoIP provider that pays you!
> 
> 
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users
mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
>

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
US NY +1 516 687 5200          FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!
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On Wed, June 28, 2006 10:14, undrhil.1528785@bloglines.com
said:
> Well, look at it this way: if you get the working, you can buy one of
> those
> tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia
> soundcard
> and a ethernet port.  Run Linux off a CF card and have it setup to *only*
> interface with Skype and Asterisk.  Basically, make a Skype ATA, but it
> would
> convert Skype to SIP.  I think that could still be considered an ATA,
> right?
>  Or a gateway at least.
>
> Since you can make a Skype account for free and
> can (for right now) make US and Canada LD calls for free, I think the cost
> and time to make them would be worth it.  :)  And if you figure out a good
> price for them, people might even buy them from you....
>
> Undrhil
>
Another advantage is that you can reach all those people who have Skype
and are not willing to try Voipbuster or similar SIP based providers, and
tell them that SIP/IAX/Asterisk *is* the better solution, because they
cannot do the same with Skype the other way round!   ;-p

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
  AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
  2 Sweex HFC-PCI cards

On Wed, Jun 28, 2006 at 08:14:56AM -0000, undrhil.1528785@bloglines.com
wrote:
> Well, look at it this way: if you get the working, you can buy one of those
> tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia
soundcard
> and a ethernet port.  Run Linux off a CF card and have it setup to *only*
> interface with Skype and Asterisk.  Basically, make a Skype ATA, but it
would
> convert Skype to SIP.  I think that could still be considered an ATA,
right?
>  Or a gateway at least.
> 
> Since you can make a Skype account for free and
> can (for right now) make US and Canada LD calls for free, I think the cost
> and time to make them would be worth it.  :)  And if you figure out a good
> price for them, people might even buy them from you....
You would be violating the terms of usage of their API if you want to
use (let alone sell) such a device.

-- 
Tzafrir Cohen      sip:tzafrir@local.xorcom.com
icq#16849755       iax:tzafrir@local.xorcom.com
+972-50-7952406           
tzafrir.cohen@xorcom.com  http://www.xorcom.com

On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote:
> > Since you can make a Skype account for free and
> > can (for right now) make US and Canada LD calls for free, I think the
cost
> > and time to make them would be worth it.  :)  And if you figure out a
good
> > price for them, people might even buy them from you....
> 
> You would be violating the terms of usage of their API if you want to
> use (let alone sell) such a device.
> 
I am unsure if all the hardware devices are basically usb soundcards or
not, havent really looked, but if they arent then it would seem to me
that its possible to do.  Further I dont think it would be against their
api to write sofeware that uses their "api".  That is what was being
discussed when this comment came out, so ...


-- 
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
US NY +1 516 687 5200          FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!
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Hi Marco,

Marco Mouta wrote:
> Please feel free to contact me if you have more ideas to improve this
> solution, currently i didn't test more than one simultaneous calls
> incoming and outgoing through Skype.
get it running on unix so you can run it on the asterisk server.



Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook