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" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@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 <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > > To: "C F" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial > > Discussion" <asterisk-users@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" <[EMAIL PROTECTED]> > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > <asterisk-users@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 <[EMAIL PROTECTED]> > > > > Date: Wed, 29 Dec 2004 00:34:28 -0500 > > > > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail > > > aftertimeout? > > > > To: Me <[EMAIL PROTECTED]> > > > > > > > > > > > > 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 <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > > > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > > <asterisk-users@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([EMAIL PROTECTED]) > > > > > > exten => 1200,4,Goto,t|1 > > > > > > > > > > > > > > > > > > On Tue, 28 Dec 2004 14:20:02 -0600, Me <[EMAIL PROTECTED]> > > > 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@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([EMAIL PROTECTED]) > > > > > > > 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@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 > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Asterisk-Users mailing list > > > > > > > Asterisk-Users@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 > > > > > > > > > > > > > _______________________________________________ > > > > > > Asterisk-Users mailing list > > > > > > Asterisk-Users@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 > > > > > > > > > > > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@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 > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@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 > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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 > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
