[Asterisk-Users] call screening

[C F]

Right now I'm stuck at this point:
[default]
exten => 1002,Macro(stdcs,1002,SIP/1002)

[macro-stdcs]
;; arg1 exten
;; arg2 device
exten => s,1,Wait(0.2)
exten => s,2,Playback(vm-rec-name)
exten => s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten => s,4,Record(${SCREEN_FILE}:gsm|2|4)
exten => s,5,Playback(pls-wait-connect-call)
exten => s,6,Dial(${ARG2},30,gM(screen^${SCREEN_FILE}))
exten => s,7,Voicemail(u${ARG1})
exten => s,8,Playback(Goodbye)
exten => s,9,Hangup
exten => s,107,Voicemail(b${ARG1})
exten => s,108,Playback(Goodbye)
exten => s,109,Hangup


[macro-screen]
exten => s,1,Wait(0.2)
exten => s,2,Playback(${ARG1})
;1 TO ACCEPT, 2 TO REJECT, 3 TO TRANSFER
exten => s,3,Read(ACCEPT1|custom/2) ;this file contains the phrase'you
have an incoming call from'
exten => s,4,Noop(${ACCEPT1})
exten => s,5,Gotoif($[${ACCEPT1}=1] ?50) ;connect
exten => s,6,Gotoif($[${ACCEPT1}=2] ?30) ;reject to vm
;exten => s,6,Gotoif($[${ACCEPT1}=3] ?40) ;TRANSFER
exten => s,7,Gotoif($[${ACCEPT1}=4] ?50:50) ;any thing else connect

exten => s,30,SetVar(MACRO_RESULT=CONTINUE)
exten => s,31,System(/bin/rm ${ARG1})
;not yet written
;exten => s,40,	;ask for extension then set macro to goto that and continue
exten => s,50,System(/bin/rm ${ARG1})

when I dial exten 1002 I get the follwoing in the CLI:
 -- Executing Macro("SIP/1000-906f", "stdcs|1002|SIP/1002") in new stack
    -- Executing Wait("SIP/1000-906f", "0.2") in new stack
    -- Executing Playback("SIP/1000-906f", "vm-rec-name") in new stack
    -- Playing 'vm-rec-name' (language 'en')
    -- Executing SetVar("SIP/1000-906f",
"SCREEN_FILE=/tmp/1000-1103501744") in new stack
    -- Executing Record("SIP/1000-906f",
"/tmp/1000-1103501744:gsm|2|4") in new stack
    -- Playing 'beep' (language 'en')
    -- Executing Playback("SIP/1000-906f", "pls-wait-connect-call") in
new stack    -- Playing 'pls-wait-connect-call' (language 'en')
    -- Executing Dial("SIP/1000-906f",
"SIP/1002|30|gM(screen^/tmp/1000-1103501744)") in new stack
    -- Called 1002
    -- SIP/1002-1507 is ringing
    -- SIP/1002-1507 answered SIP/1000-906f
    -- Executing Wait("SIP/1001-1507", "0.2") in new stack
    -- Executing Playback("SIP/1002-1507", "/tmp/1000-1103501744") in new stack
    -- Playing '/tmp/1000-1103501744' (language 'en')
    -- Executing Read("SIP/1002-1507", "ACCEPT1|custom/2") in new stack
    -- Playing 'custom/2' (language 'en')
    -- User entered ''
    -- Executing NoOp("SIP/1001-1507", "") in new stack
    -- Executing GotoIf("SIP/1001-1507", "=1 50") in new stack
    -- Executing GotoIf("SIP/1001-1507", "=2 30") in new stack
    -- Attempting native bridge of SIP/1000-906f and SIP/1002-1507
    -- Executing VoiceMail("SIP/1002-906f", "u1002") in new stack
    -- Playing 'voicemail/default/1002/unavail' (language 'en')
  == Spawn extension (macro-stdcs, s, 7) exited non-zero on
'SIP/1000-906f' in macro 'stdcs'
  == Spawn extension (default, 1002, 1) exited non-zero on 'SIP/1000-906f'

I have no clue why the Read doesn't work, for some reason it refuses
to work from within this macro but works from any where else. Need
help ASAP.


On Sun, 19 Dec 2004 18:37:40 -0500, C F <shmaltz at gmail.com> wrote:
> According to this it exists:
> http://www.voip-info.org/wiki-Asterisk+cmd+Dial
> However I'm testing it for the last 8 hours with no  success.
> Recompiling after reading this:
> http://bugs.digium.com/bug_view_page.php?bug_id=0002905
> will post back
> 
> 
> On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed
> <treed at copilotconsulting.com> wrote:
> > On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
> > > Is there a way to use asterisk for call screening?
> > >
> > > Meaning, a call comes in, asterisk answers with voicemail after I don't
> > > pickup, and the voicemail prompt + the caller's message a played via the
> > > sound card on asterisk. If I wan't to pick up, I do so by picking up the
> > > phone and dialing something.
> > > Is it doable?
> >
> > I think I would try something like inviting the voicemail, the caller, and
> > an auto-answer (intercom) channel on your VOIP phone into a MeetMe where
> > your voiphone is not allowed to talk, only listen. Then you would hear
> > what is going on and if you wanted to talk to the person you could join
> > the MeetMe on a different line and talk to the person.
> >
> > --
> > Tracy Reed    http://copilotcom.com
> > This message is cryptographically signed for your protection.
> > Info: http://copilotconsulting.com/sig
> >
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
> >
> >
> >
>