well, aint that a bugger. Just looked at my contacts list on skype and the bloody thing is working ... wtf ?
Another question: I just tried calling in like this external => ddi => dial(skype) and got a load of static with WARNING[15328]: channel.c:3098 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) on the console. Fired up a sip client, made the same call, and all was ok. Any clues ? 2009/12/5 Julian Lyndon-Smith <[email protected]>: > As I have no friends and no life I thought that I would set up my > asterisk server with Skype. > > 1) Paid the $, got the licence, built and installed > 2) create a business skype account (called company "foo") > 3) created a member of the business called "bar" > 4) updated the skype conf file > 5) restarted asterisk > > > > => skype show settings > Skype For Asterisk Settings: > engine_directory: /tmp > data_directory: /var/spool/asterisk/skype > defaultuser: bar > bind_address: 0.0.0.0 > bind_port: 0 > rtp_address: 127.0.0.1 > https_proxy: > https_proxy_user: > https_proxy_password: > socks5_proxy: > socks5_proxy_user: > socks5_proxy_password: > disable_tcpauto: no > disable_udp: no > debug: no > > => skype show users > Skype Users> > bar: Logged In > > 6) added a test to extensions.conf > > exten => 123650,1,Dial(Skype/[email protected]) > exten => 123650,2,Hangup() > > and get a > > Everyone is busy/congested at this time (1:0/0/1) > [Dec 5 20:15:08] -- Executing [123...@isdnspan1:2] > Hangup("Zap/1-1", "") in new stack > > My skype client can find "bar", but it is "offline", so I can't place > calls either > > Anyone know what I am doing wrong ?? (1.4 source svn trunk) > > TIA > > Julian > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
