Re: [asterisk-users] Chanskype Support

2009-09-06 Thread Tilghman Lesher
On Saturday 05 September 2009 23:10:19 Luke McKee wrote:
 I bought paid for and used to use ChanSkype for skype connectivity.
 Now I tried to download the software again after the my old server got
 retired and someone else is squatting on their website. Can someone
 give me the latest ChanSkype binaries?

That's not a squatter.  That's the site of the company that developed that
solution.  See this archive post, for context:
http://lists.digium.com/pipermail/asterisk-biz/2006-October/017891.html

 Why do I have to rebuy the functionality I've already paid for too?

That's a question for Instant Solutions.

 Is digium going to feel for all the people who have to re-buy what they
 have alredy paid for.

You bought a solution from a particular provider.  I'm not sure why Digium
should give you their product for free, especially when they haven't had
anything to do with the other product and, beyond that, cannot verify that
you paid anything to anybody.
 
-- 
Tilghman  Teryl
with Peter, Cottontail, Midnight, Thumper,  Johnny (bunnies)
and Harry, BB,  George (dogs)

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[asterisk-users] Chanskype Support

2009-09-05 Thread Luke McKee
Hello,

I bought paid for and used to use ChanSkype for skype connectivity.
Now I tried to download the software again after the my old server got
retired and someone else is squatting on their website. Can someone
give me the latest ChanSkype binaries?

Why do I have to rebuy the functionality I've already paid for too? Is
digium going to feel for all the people who have to re-buy what they
have alredy paid for. I guess the same happened when ALSA replaced
commercial OSS from 4front. I paid $99 to them. Does anyone know where
the ChanSkype people are?

Cheers,

Luke

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Re: [asterisk-users] ChanSkype

2007-06-25 Thread Hugo Miguel de Almeida Teixeira Picao
Hi Kyle,

You need to set up a inbound route from DID=skype1 and tell him where to finish.

Something like:

exten = skype1,1,Set(FROM_DID=skype1)
exten = skype1,n,Goto(ext-local,1000,1)


Hope it helps.

Best Regards,

Hugo Picão
Link Consulting - RedesSegurança
Tel: 213 100 182
Av. Duque de Ávila, 23
1000-138 Lisboa

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Vorster
Sent: quinta-feira, 21 de Junho de 2007 14:07
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ChanSkype

Hello,

I recently installed chanskype on my asterisk box and it works like a
dream, can phone out.

But no idea how to setup the incoming calls, every time I phone my skype
name it just connects and disconnect the call right away.

I get the following on asterisk -rvv

Verbosity was 1 and is now 14
   == Sent cmd 'GET CALL 175 TYPE' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 PARTNER_HANDLE' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 PSTN_NUMBER' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'ALTER CALL 175 END HANGUP' to fd 18 on Skype dev 'skype1'
   == Unknown event 'ALTER CALL 175 END HANGUP' from Skype device 'skype1'
   == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1'

Any one got some advice ?

Kind Regards,
Kyle Vorster

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[asterisk-users] ChanSkype

2007-06-21 Thread Kyle Vorster
Hello,

I recently installed chanskype on my asterisk box and it works like a
dream, can phone out.

But no idea how to setup the incoming calls, every time I phone my skype
name it just connects and disconnect the call right away.

I get the following on asterisk -rvv

Verbosity was 1 and is now 14
   == Sent cmd 'GET CALL 175 TYPE' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 PARTNER_HANDLE' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 PSTN_NUMBER' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1'
   == Sent cmd 'ALTER CALL 175 END HANGUP' to fd 18 on Skype dev 'skype1'
   == Unknown event 'ALTER CALL 175 END HANGUP' from Skype device 'skype1'
   == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1'

Any one got some advice ?

Kind Regards,
Kyle Vorster

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Re: [asterisk-users] ChanSkype

2007-06-21 Thread David Gomillion

On 6/21/07, Kyle Vorster [EMAIL PROTECTED] wrote:


Hello,

I recently installed chanskype on my asterisk box and it works like a
dream, can phone out.

But no idea how to setup the incoming calls, every time I phone my skype
name it just connects and disconnect the call right away.

...
Any one got some advice ?



My advice: contact the developer of ChanSkype. You have to pay for that,
right? Hopefully, it comes with some support.

In the mean time, make sure your incoming call's context exists, ensure that
you have an s and i extension in that extension just in case the number
comes in differently than how you expect, and put some no-ops in, maybe have
it echo the EXTEN variable. You know, basic troubleshooting.

Good luck,
David
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Re: [asterisk-users] chanskype

2007-01-20 Thread Moises Silva

Hi,
I tried the try version of chanskype, however, everytime that I make a call
asterisk generate an error

So you think is easy to us guess wich error you are getting?
Seriously, I think you should read this:
http://www.catb.org/~esr/faqs/smart-questions.html


Anyone has experience with this? Since I tried to contact the support but
they never replied.


Please provide more information about the error, and search in
voip-info.org how to raise the verbosity level of asterisk in the
console.

Also sometimes help searching the error in google.

Regards


--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] chanskype

2007-01-20 Thread Il Neofita

Hi,
I was wondering if someone had  problems with chanskype.
Since I am wondering if they are a credible company or not.

See you
On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote:


 Hi,
 I tried the try version of chanskype, however, everytime that I make a
call
 asterisk generate an error
So you think is easy to us guess wich error you are getting?
Seriously, I think you should read this:
http://www.catb.org/~esr/faqs/smart-questions.html

 Anyone has experience with this? Since I tried to contact the support
but
 they never replied.

Please provide more information about the error, and search in
voip-info.org how to raise the verbosity level of asterisk in the
console.

Also sometimes help searching the error in google.

Regards


--
Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org;
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Re: [asterisk-users] chanskype

2007-01-20 Thread Andrew Joakimsen

That's a great site! Perhaps it should be auto-sent to every poster
for the first 30 days of their membership:

Despite this, hackers have a reputation for meeting simple questions
with what looks like hostility or arrogance. It sometimes looks like
we're reflexively rude to newbies and the ignorant. But this isn't
really true.

What we are, unapologetically, is hostile to people who seem to be
unwilling to think or to do their own homework before asking
questions. People like that are time sinks — they take without giving
back, and they waste time we could have spent on another question more
interesting and another person more worthy of an answer. We call
people like this losers (and for historical reasons we sometimes
spell it lusers).

On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote:

 Hi,
 I tried the try version of chanskype, however, everytime that I make a call
 asterisk generate an error
So you think is easy to us guess wich error you are getting?
Seriously, I think you should read this:
http://www.catb.org/~esr/faqs/smart-questions.html

 Anyone has experience with this? Since I tried to contact the support but
 they never replied.

Please provide more information about the error, and search in
voip-info.org how to raise the verbosity level of asterisk in the
console.

Also sometimes help searching the error in google.

Regards


--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[asterisk-users] chanskype

2007-01-19 Thread Il Neofita

Hi,
I tried the try version of chanskype, however, everytime that I make a call
asterisk generate an error

Anyone has experience with this? Since I tried to contact the support but
they never replied.

Thank you
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