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

Wed Dec 29 08:56:47 MST 2004

Thanks for the example! I was using something similar to this that I found
in the Wiki but the problem I ran into was the Record() part. Each time *
got to the record part I got some error saying, can't remember what it was,
I will dig it up and post it in a reply.

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: Wednesday, December 29, 2004 9:41 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?

> [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
> > > > > > >
> > > > > > >
> > > > > > > _______________________________________________
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> > > > >
> > > > >
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> > >
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