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 _______________________________________________ --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
On 6/21/07, Kyle Vorster <kyle at serv.co.za> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070621/a82744c0/attachment.htm
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 Pic?o Link Consulting - Redes&Seguran?a Tel: 213 100 182 Av. Duque de ?vila, 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 _______________________________________________ --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 _______________________________________________ --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