[asterisk-users] missed call notification
Warren Selby
wcselby at selbytech.com
Thu May 5 16:55:12 CDT 2011
Make it to Astricon this year, I'll buy you a drink and you can take the test there! That's what I did last year. :)
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:27 PM, Sherwood McGowan <sherwood.mcgowan at gmail.com> wrote:
> 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> 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
>> To: 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
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] missed call notification
>>
>> if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the macro, you'd get 's'....do it while you still have the called number as the EXTEN
>>
>> On Thu, May 5, 2011 at 12:42 PM, satish patel <satish_lx at hotmail.com> wrote:
>>
>> Also check for CANCEL, since this should be the status if the caller
>> hangs up before the call is picked up.
>>
>> But CANCEL is return nothing
>>
>>
>> [macro-stdexten]
>> exten => s,1,Dial(${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
>>
>> 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
>> > >
>> > > --
>> > > _____________________________________________________________________
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>> >
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>> -- _____________________________________________________________________ -- 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
>>
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>> _____________________________________________________________________
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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: 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 --
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>> http://www.asterisk.org/hello
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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:
>> http://www.asterisk.org/hello
>>
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>> 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
>
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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:
> http://www.asterisk.org/hello
>
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