[Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

C F shmaltz at gmail.com
Wed Dec 29 08:41:52 MST 2004 

[macro-stdcs]
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;; Call a device with cs			       ;;
;; Takes 2 arguments				     ;;
;; arg1 exten						  ;;
;; arg2 device						  ;;
;; tnen goes to vm				       ;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;screen-record: Please record your name press pound when finished.
;screen-from: You have a call from
;screen-accept: Press 1 to accept 2 to reject, and 3 to transfer.
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,mtM(screen^${SCREEN_FILE}))
exten => s,7,Goto(17);VM
'I always leaeve room for more in case the dial plan changes
exten => s,17,Voicemail(u${ARG1})
exten => s,18,Playback(goodbye)
exten => s,19,Hangup
exten => s,107,Goto(17)

exten => h,1,System(/bin/rm ${ARG1}.gsm)

 [macro-screen]
;this is called in the Dial statement using M
;ARG1 recorded name to play back
;TODO: add a response timeout, after which the message is repeated
(needed for outgoing zap fxo channels) and absolute timeout, after
which VM is used
exten => s,1,noop(${ARG1})
exten => s,2,Playback(custom/screen-from) ;you have an incoming call from:
exten => s,3,Playback(${ARG1})
;press 1 to accept 2 to reject 3 to transfer
exten => s,4,Read(ACCEPT|custom/screnn-accept|1)
exten => s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
exten => s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
exten => s,7,Gotoif($[${ACCEPT} = 3] ?40) ;TRANSFER
exten => s,8,Gotoif($[${ACCEPT} = 4] ?30:30) ;any thing else vm

exten => s,30,SetVar(MACRO_RESULT=CONTINUE)
exten => s,31,Goto(50)

exten => s,40,Read(TEXTEN|custom/screen-exten|3)
;ask for extension then set macro to goto that and continue
exten => s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)  
exten => s,42,SetVar(MACRO_RESULT=GOTO:internaldial^${TEXTEN}^1)
exten => s,43,Goto(50)
exten => s,45,Gotoif($[${TEXTEN} = 0] ?46:46)
;the logic is here to allow transfer to operator, i just didn't imlepent it yet
exten => s,46,SetVar(MACRO_RESULT=CONTINUE)
exten => s,47,Goto(50)

exten => s,50,System(/bin/rm ${ARG1}.gsm)

exten => h,1,System(/bin/rm ${ARG1}.gsm)




On Wed, 29 Dec 2004 00:35:34 -0600, Me <mylist at lightwavetech.com> wrote:
> Nevermind, it looks like "Asterisk cmd Read" is my lucky command :)
> 
> Thanks!
> 
> Start Your Own Internet Service!
> http://www.YourOwnISP.com
> 
> ----- Original Message -----
> From: "Me" <mylist at lightwavetech.com>
> To: "C F" <shmaltz at gmail.com>; "Asterisk Users Mailing List - Non-Commercial
> Discussion" <asterisk-users at lists.digium.com>
> Sent: Wednesday, December 29, 2004 12:19 AM
> Subject: Re: [Asterisk-Users] Sending call to analog then to
> Vmailaftertimeout?
> 
> > I was trying this logic before, I got as far as going into the Macro,
> > playing a message and then.. Well... I got lost, I am not sure how to go
> > about require them to press a button. Normally I can make someone press an
> > extension but from what I read about Macros in * you have to stay within
> the
> > "s" extension.
> >
> > Any idea where I can find an example of this sort of thing?
> >
> > Thanks!
> >
> > Start Your Own Internet Service!
> > http://www.YourOwnISP.com
> > ----- Original Message -----
> > From: "C F" <shmaltz at gmail.com>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Tuesday, December 28, 2004 11:34 PM
> > Subject: Re: [Asterisk-Users] Sending call to analog then to
> > Vmailaftertimeout?
> >
> >
> > > ---------- Forwarded message ----------
> > > From: C F <shmaltz at gmail.com>
> > > Date: Wed, 29 Dec 2004 00:34:28 -0500
> > > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
> > aftertimeout?
> > > To: Me <mylist at lightwavetech.com>
> > >
> > >
> > > try the M option which will do a macro and will not connect the caller
> > > unless s/he presses some button. and if no button is pressed then it
> > > goes to VM. now remember to replay the message (to press the button) a
> > > few times b4 going to VM otherwise they will never hear it, since *
> > > considers it answered .
> > > http://www.voip-info.org/wiki-Asterisk+cmd+dial
> > >
> > >
> > > On Tue, 28 Dec 2004 23:29:54 -0600, Me <mylist at lightwavetech.com> wrote:
> > > > I was aware of the "c" option but it's a pain for people to have to
> > press
> > > > the # sign plus they have to know they are suppose to do that. In
> > addition,
> > > > I tried to use the "A" option to play a sound to them when they answer
> > > > reminding them to press pound at the end of the message but the sound
> > > > doesn't play until they press pound :)
> > > >
> > > > So.. It appears I am still stuck with * considering the call answered
> > when
> > > > the Zap channels grabs it and connects the other leg of the call.
> > Hopefully
> > > > there is some other way to make this happen.
> > > >
> > > > Thanks for the feedback though.
> > > >
> > > > Start Your Own Internet Service!
> > > > http://www.YourOwnISP.com
> > > >
> > > > ----- Original Message -----
> > > > From: "C F" <shmaltz at gmail.com>
> > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > <asterisk-users at lists.digium.com>
> > > > Sent: Tuesday, December 28, 2004 6:26 PM
> > > > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
> > > > aftertimeout?
> > > >
> > > > > Follow these:
> > > > > http://www.voip-info.org/wiki-Asterisk+zap+channels
> > > > > looks like this would work:
> > > > >  exten => 1200,1,playback(pls-wait-connect-call)
> > > > >  exten => 1200,2,Dial(Zap/1c/5555551212,20,rTt) ;note the c after
> the
> > > > > channel number
> > > > >  exten => 1200,3,VoiceMail(u100 at lightwavetech.com)
> > > > >  exten => 1200,4,Goto,t|1
> > > > >
> > > > >
> > > > > On Tue, 28 Dec 2004 14:20:02 -0600, Me <mylist at lightwavetech.com>
> > wrote:
> > > > > > Sorry about the HTML emails, on my laptop and forgot to change the
> > > > sending
> > > > > > format from the default.
> > > > > >
> > > > > >
> > > > > > ----- Original Message -----
> > > > > > From: Me
> > > > > > To: asterisk-users at lists.digium.com
> > > > > > Sent: Tuesday, December 28, 2004 2:01 PM
> > > > > > Subject: [Asterisk-Users] Sending call to analog then to Vmail
> after
> > > > > > timeout?
> > > > > >
> > > > > > I have one analog line hooked in my Asterisk box using an x100p (I
> > think
> > > > > > that's the model number).
> > > > > >
> > > > > > When I do this in my extensions.conf:
> > > > > >
> > > > > > exten => 1200,1,playback(pls-wait-connect-call)
> > > > > > exten => 1200,2,Dial(Zap/1/5555551212,20,rTt)
> > > > > > exten => 1200,3,VoiceMail(u100 at lightwavetech.com)
> > > > > > exten => 1200,4,Goto,t|1
> > > > > >
> > > > > > The phone rings beyond the 20 second timeout and never really goes
> > to
> > > > the *
> > > > > > voicemail. I can't seem to get it to timeout regardless of how
> many
> > > > seconds
> > > > > > I set it to.
> > > > > >
> > > > > > I assume this has something to do with the fact that * considers
> the
> > > > call
> > > > > > answered as soon as the zap channel picks it up, right?
> > > > > >
> > > > > > Anyhow, is there a way to make the above config work and go to the
> *
> > > > > > voicemail after 20 seconds if the called party does not answer
> after
> > 20
> > > > > > seconds? Also, what happens if the called party's line is busy,
> have
> > not
> > > > run
> > > > > > into this yet so I am curious.
> > > > > >
> > > > > > Thanks!
> > > > > >
> > > > > > --
> > > > > > Start Your Own Internet Service!
> > > > > > http://www.YourOwnISP.com
> > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > Asterisk-Users mailing list
> > > > > > Asterisk-Users at lists.digium.com
> > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > > > > >
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> > > > > >
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> > > >
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