[asterisk-users] entering a password to have access to a sip account?!
Steve Totaro
stotaro at totarotechnologies.com
Sun Aug 24 20:55:49 CDT 2008
Roland,
The simple solution is to utilize the power of contexts (put exten 300
in a different context in sip.conf or db) and includes to separate yet
include 300 (so 300 can be called and call other internal extensions).
Add authenticate before the dial statement.
The easiest way to do it, is just copy the [spa] context in your
dialplan and then change the context from [spa] to [restricted_300]
(or whatever) and then just add the authenticate statement as below
and renumber the dial prio to 2 or (n for next). Make sure your
context in sip.conf for that sip extension matches this newly created
context. There are probably cleaner ways of doing it, but one thing
at a time :)
exten =>_01,1,Authenticate(1234)
exten =>_01,2,Dial(SIP/$(EXTEN)@300) ; old ogero line
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 4:20 PM, RoLaNd RoLaNd <r_o_l_a_n_d at hotmail.com> wrote:
> Hello Steve,
>
> thanks for the advice :)
>
> though one prob! if i add the authenticate line itll require all callers to
> enter 1234 to access *ANY* sip account..
> even though this would come in handy at some point but at the moment i just
> want to deny the extension 300 from being able to call "01" unless the
> caller entered a password..
> find below wht i did so far..
>
>
>
>
>
> [sipura-line]
> exten => 301,1,Answer() ; Answer inbound calls
> exten => 301,2,Playback(silence/1)
> exten => 301,3,Background(simzy1) ; input an extension
> exten => 301,4,authenticate(1234)
> exten => 301,5,WaitExten(8)
> exten => 301,6,Dial(SIP/100,15) ; goes to operator
> exten => 301,3,Wait(8)
> include => spa
> exten => _XXX,6,VoiceMail(100 at default)
> exten => 301,n,Hangup()
>
>
>
>
> [spa]
> exten =>_301,1,GoTo(sipura-line,${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 => 303,1,VoicemailMain ; voicemail box to be redirected to
>
>
>
>> Date: Sun, 24 Aug 2008 12:05:02 -0400
>> From: stotaro at totarotechnologies.com
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] entering a password to have access to a sip
>> account?!
>>
>> 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
>> >
>>
>> _______________________________________________
>> -- 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
>
> ________________________________
> Connect to the next generation of MSN Messenger Get it now!
> _______________________________________________
> -- 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