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

Grygoriy Dobrovolskyy megahohol at gmail.com
Sun Aug 24 17:21:55 CDT 2008


I have one solution in mind, maybe it is an overkill but:

You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all other sip accounts DB(300/NNN)=0 where NNN are all others sip
accounts numbers. You can use set for this, example

exten => 75,1,Set(DB(300/301)=1)
or
exten => 75,1,Set(DB(300/${Callerid(num)}=1)
exten => 76,1,Set(DB(300/${Callerid(num)}=0)
And just go and call from each phone 75 or 76 , i assume that you callerid
is the same as callerid(num) var. The methos is somehow primitive and will
not work if you have 500 extensions, but for 5 sip accounts  is a way to go.

Or create external bash script to speed up.

After this you will have as much db entryes as sip accounts in you astdb,
all we need to is is to verify the value before call

exten => 300,1,GotoIf($[${DB(300/${Callerid(num)})}=1]?2:3)
exten => 300,2,Playback(stop_calling_me)
exten => 300,3,Dial(Sip/300)

And again i assume that your sip peers have the same
Callerid(num)=extensions

Maybe i got some syntax errors, but you get the idea.

Have fun



2008/8/24 RoLaNd RoLaNd <r_o_l_a_n_d at hotmail.com>

>  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
> > >
> > >
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