[asterisk-users] ChanSkype

Hugo Miguel de Almeida Teixeira Picao hugo.picao at link.pt
Mon Jun 25 04:32:46 CDT 2007


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 - Redes&Segurança
Tel: 213 100 182
Av. Duque de Ávila, 23
1000-138 Lisboa

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kyle Vorster
Sent: quinta-feira, 21 de Junho de 2007 14:07
To: asterisk-users at 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 -rvvvvvvvvvvvvvv

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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