Dear all, I have small question in sip.conf I added
[service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip); exten => 1,3,Hangup; but when I use next construction(As I understand it is used to allow to process any extension dialed by user) exten => s,1,Answer; exten => s,2,Playback(hello-world,skip); exten => s,3,Hangup; Asterisk says call rejected due to no extension. What is wrong? any body can make spot lighter. Thank in advance. _______________________________________________ -- 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
