> i have just started to configure access to the * over SIP-Phones. > Therefore I have defined this SIP-Phone in sip.conf: > > [tobias] > type=friend > username=tobias > secret=tobias > auth=md5 > host=dynamic > reinvite=no > dtmfmode=inband > callerid="Tobias" <1087006> > allow=all > context=javaAgi > dtmfmode=rfc2833 > > > As you can see i am directing calls from this user to the context > [javaAgi] which is defined here in extension.conf: > > [javaAgi] > exten => s,1,Answer() > exten => s,2,Playback(code1000) > exten => s,3,Hangup() > exten => 1,1,Answer() > exten => 1,2,Playback(code1000) > exten => 1,3,Hangup() > > If i dial 1 on my SIP Phone everything works as suspected, the call is > answered and the gsm-file is played. My understanding of the > 's'-extension is, that it is executed then a call comes in an there is > no extension wich matches the called number. But if i dial a random > number i get an "404 Not found" error.
The "s" extension matches only when "no" digits are dialed. Dialing a "1" is a digit, so no match. Try playing around with exten=>_XXXX.,1,Answer() and understand what the differences are. _______________________________________________ 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
