Hello Roland, You can use the cmd Read for this. http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user collect the required digits using Read; check the collected digits; if yes jump to required extension; else reject user or whatever you want to do. I could've written out the dialplan, but well... you are a newbie you said, so you gotta learn ;-) . Hope this helps. - Ben. --- On Sun, 8/24/08, RoLaNd RoLaNd <[EMAIL PROTECTED]> wrote: > From: RoLaNd RoLaNd <[EMAIL PROTECTED]> > Subject: [asterisk-users] entering a password to have access to a sip > account?! > To: asterisk-users@lists.digium.com > Date: Sunday, August 24, 2008, 3:26 PM > Hi all, > > i;m obviously a newbie, its been 2 days that im trying to > figure out a way to deny a specific extension (300) from > calling another specific extensions (03) except if the > caller punch a specified password.. sorry if im not > explaining myself well.. heres an example: > > i called my pstn line(with 300 as its sip account), an > attendant answers and asks me to punch in an extension > number right now if i dial "03" it rings at the > other end! though i dont want that to happen! i want to set > asterisk up in a way tht if i dial "03" from > "300" to ask me for a password... or it wont let > the line go through! > > > can anyone guide me through this issue! im really going > crazy to get this done! any help would truly and utterly be > appreciated:) > > > > ps: find below my extensions.conf > > > [sipura-line] > exten => 301,1,Answer() ; Answer inbound calls > exten => 301,2,Playback(silence/1) > exten => 301,3,Background(simzy1) ; input an extension > exten => 301,4,WaitExten(8) > exten => 301,5,Dial(SIP/100,15) ; goes to operator > exten => 301,4,Wait(8) > include => spa > exten => _XXX,6,VoiceMail([EMAIL PROTECTED]) > exten => 301,n,Hangup() > > > > > [spa] > exten =>_301,1,GoTo(sipura-line,${EXTEN},1) > exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals > to 5 seconds so it will ring 3 times > exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 > voicemail box if line is busy or unavailable > exten => _1XX,3,HangUp() > exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals > to 5 seconds so it will ring 3 times > exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to > voicemail box if line is busy or unavailable > exten => _2XX,3,HangUp() > exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals > to 5 seconds so it will ring 3 times > exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 > voicemail box if line is busy or unavailable > exten => _3XX,3,HangUp() > exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line > ;exten =>_01,2,Set(TIMEOUT(absolute)=5) > exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line > exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer > exten =>_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte > exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference > exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4 > exten => 303,1,VoicemailMain ; voicemail box to be > redirected to > > > _________________________________________________________________ > News, entertainment and everything you care about at > Live.com. Get it now! > http://www.live.com/getstarted.aspx_______________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users