[Asterisk-Users] call screening

[C F]

OK I now know what was/is worng, my SIP is wrong it doesn't give 2 way
audio, so first I'm going to fix this and then we will see.


On Sun, 19 Dec 2004 19:26:59 -0500, C F <shmaltz at gmail.com> wrote:
> 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
> > >
> > >
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