[asterisk-users] missed call notification

satish patel satish_lx at hotmail.com
Thu May 5 12:42:07 CDT 2011



Also check for CANCEL, since this should be the status if the caller
hangs up before the call is picked up.

But CANCEL is return nothing 

[macro-stdexten]
exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t)             ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)
exten => s,n,Goto(s-${DIALSTATUS},1)                     ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
;exten => s,n,Hangup()

exten => s-CANCEL,1,Verbose(Hangup call)



CLI
 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-00000023' in macro 'stdexten'
  == Spawn extension (from-sip, 7516, 1) exited non-zero on 'SIP/7527-00000023'


Look like its going back to original extension :( I hate macro 


From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Thu, 5 May 2011 17:15:53 +0000
Subject: Re: [asterisk-users] missed call notification








You want me to do this in macro-stdexten ? I have following dialplan.  I have used "h" extension in original context because you can't you "h" inside macro right ? 

[macro-stdexten]
exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t)             ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)
exten => s,n,Goto(s-${DIALSTATUS},1)                     ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s,n,Hangup()
exten => s-NOANSWER,1,Voicemail(${ARG1},u)               ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,Voicemail(${ARG1},b)                   ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Hangup()
exten => s-CONGESTION,1,Voicemail(${ARG1},u)             ; Like above, write a macro for this case
exten => s-CONGESTION,n,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)                       ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1})                      ; If they press *, send the user into VoicemailMain


[from-sip]
...blah...blah..

exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh "" 
"${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}")





> From: wcselby at selbytech.com
> Date: Thu, 5 May 2011 12:10:09 -0500
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] missed call notification
> 
> Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, then reference that variable in your h exten. 
> 
> Thanks,
> --Warren Selby, dCAP
> 
> On May 5, 2011, at 11:59 AM, satish patel <satish_lx at hotmail.com> wrote:
> 
> > Hi All,
> > 
> > I am using http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/   to implement missed call feature. and i modify script to grab email address from voicemail.conf 
> > 
> > But i am not able to see DEST extension in this script ?  what would be the variable to get destination extension so base on that i can grab email address of user from voicemail.conf 
> > 
> > exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh "" "${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}")
> > 
> > Calling from 7527<--to--->7101  but i can see only 7527 not dest 7101
> > 
> > 
> > CLI outout
> >     -- Executing [h at from-sip:1] System("SIP/7527-0000000d", "/var/lib/asterisk/agi-bin/processcallemail.sh "" "7527" "Guest" "CANCEL" """) in new stack
> > shirley*CLI> exit
> > 
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >               http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
 		 	   		  

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110505/9a929dc3/attachment.htm>


More information about the asterisk-users mailing list