I have set up extension.conf and sip.con with default parameter of UNIVOICE server, but Asterisk show this message when I call a number:
Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899 create_addr: No such host: univoice,Ttr Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup("SIP/100-2331", "") in new stack ___________________________________ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users