[asterisk-users] missed call notification

Thu May 5 13:27:29 CDT 2011

Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have
dCAP by your name! I've been at this 6-7 years and still haven't gotten off
my butt and taken the tests :D

On Thu, May 5, 2011 at 1:20 PM, Warren Selby <wcselby at selbytech.com> wrote:

> And Sherwood beats me to the punch again :).
>
> Thanks,
> --Warren Selby, dCAP
>
> On May 5, 2011, at 1:15 PM, Sherwood McGowan <sherwood.mcgowan at gmail.com>
> wrote:
>
> No, the variables are channel specific except for when they're inherited,
> which doesn't affect you here
>
> On Thu, May 5, 2011 at 1:02 PM, satish patel < <satish_lx at hotmail.com>
> satish_lx at hotmail.com> wrote:
>
>>  After google i found something and i tried following. I set variable
>> before Dial and its give me proper value in "h" extension but now question
>> is if multiple user dial multiple extension then will it  overwrite current
>> variable value ?
>>
>> exten => s,1,Set(_CALLED_EXT=${ARG2})
>> exten => s,n,Dial(${ARG2}&iax2/${ARG1},20,t)
>>
>> ------------------------------
>> From: <satish_lx at hotmail.com>satish_lx at hotmail.com
>> To: <asterisk-users at lists.digium.com>asterisk-users at lists.digium.com
>> Date: Thu, 5 May 2011 17:52:54 +0000
>>
>> Subject: Re: [asterisk-users] missed call notification
>>
>> Could you please tell me how ( Syntax ) and where in macro ?
>>
>> I am not expert in dialplan variables. I appreciate your help
>>
>> ------------------------------
>> Date: Thu, 5 May 2011 12:44:19 -0500
>> From: <sherwood.mcgowan at gmail.com>sherwood.mcgowan at gmail.com
>> To: <asterisk-users at lists.digium.com>asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] missed call notification
>>
>> if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the
>> macro, you'd get 's'....do it while you still have the called number as the
>> EXTEN
>>
>> On Thu, May 5, 2011 at 12:42 PM, satish patel < <satish_lx at hotmail.com>
>> satish_lx at hotmail.com> wrote:
>>
>>
>> Also check for CANCEL, since this should be the status if the caller
>> hangs up before the call is picked up.
>>
>> But CANCEL is return nothing
>>
>>
>> [macro-stdexten]
>> exten => s,1,Dial(${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>satish_lx at hotmail.com
>>
>> To: <asterisk-users at lists.digium.com>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>wcselby at selbytech.com
>> > Date: Thu, 5 May 2011 12:10:09 -0500
>> > To: <asterisk-users at lists.digium.com>asterisk-users at lists.digium.com
>> > Subject: Re: [asterisk-users] missed call notification
>> >
>> > Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call,
>> then reference that variable in your h exten.
>> >
>> > Thanks,
>> > --Warren Selby, dCAP
>> >
>> > On May 5, 2011, at 11:59 AM, satish patel < <satish_lx at hotmail.com>
>> satish_lx at hotmail.com> wrote:
>> >
>> > > Hi All,
>> > >
>> > > I am using
>> <http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/>
>> http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/to implement missed call feature. and i modify script to grab email address
>> from voicemail.conf
>> > >
>> > > But i am not able to see DEST extension in this script ? what would be
>> the variable to get destination extension so base on that i can grab email
>> address of user from voicemail.conf
>> > >
>> > > exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ""
>> "${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
>> > >
>> > > --
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>> http://www.api-digital.com --
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>>
>>
>> --
>> Sherwood McGowan
>> Telecommunications and VOIP Consultant
>>
>>
>> -- _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
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>
>
>
> --
> Sherwood McGowan
> Telecommunications and VOIP Consultant
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
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-- 
Sherwood McGowan
Telecommunications and VOIP Consultant
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