[asterisk-users] entering a password to have access to a sip account?!

Steve Totaro stotaro at totarotechnologies.com
Sun Aug 24 11:05:02 CDT 2008


You want to use Authenticate() between answer and dial.

http://www.google.com/search?q=asterisk+authenticate&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a

Thanks,
Steve Totaro

On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd <r_o_l_a_n_d at hotmail.com> wrote:
>
>
> 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(100 at default)
> 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(${EXTEN}@default) ; direct 2 voicemail box if line
> is busy or unavailable
> exten => _1XX,3,HangUp()
> exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
> will ring 3 times
> exten => _2XX,2,VoiceMail(${EXTEN}@default) ; directs to voicemail box if
> line is busy or unavailable
> exten => _2XX,3,HangUp()
> exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
> will ring 3 times
> exten => _3XX,2,VoiceMail(${EXTEN}@default) ; directs 2 voicemail box if
> line is busy or unavailable
> exten => _3XX,3,HangUp()
> exten =>_01,1,Dial(SIP/$(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
>
>
> ________________________________
> Get news, entertainment and everything you care about at Live.com. Check it
> out!
> _______________________________________________
> -- 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
>



More information about the asterisk-users mailing list