[asterisk-users] Notifying missed calls

[Łukasz Grzywański]

Hi,
strange....

  -- Goto (noanswer,s,1)
  -- Executing [s at noanswer:2] System("PJSIP/pbxmichael_in-00000418", "echo
"Verpasster Anruf vom +493511111111 um 19:13" | mail -s "Verpasster Anruf"
info at mydomain.de") in new stack
  -- Executing [s at noanswer:1] NoOp("Local/123456 at main_incoming-00000268;2",
"UID CALL: 1636222382.6032/ DATE: 20211106-191306)") in new stack
  -- Executing [s at noanswer:2] System("Local/123456 at main_incoming-00000268;2",
"echo "Verpasster Anrufvom 03511111111 um 19:13" | mail -s "Verpasster
Anruf" info at mydomain.de") in new stack

pls run

asterisk -rx "dialplan show noanswer"

and please check:

[noanswer]
exten => s,1,NoOp(UID CALL: ${UNIQUEID} /
DATE:${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" info at xxxx.de)
exten => s,n,Hangup()


 LG
Lukasz

On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello <lucabert at lucabert.de> wrote:

> Am 06.11.2021 um 15:06 schrieb Frank Vanoni:
>
> Hi Frank
>
> > The "h" extension is executed whenever a call is hang up in that
> > contexts.
> >
> > In your configuration it executes first the "s" extension (where you
> > GoTo h,1) and once that is executed, the "h" extension is executed
> > again.
>
> OK, I modified my configuration so:
>
> [main_incoming]
> exten => _00493529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _00493529123456,n,Dial(local/123456 at main_incoming,,xX)
> exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _03529123456,n,Dial(local/123456 at main_incoming,,xX)
> exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
> exten => _123456,n,Set(CHANNEL(musicclass)=default)
> exten => _123456,n,Dial(SIP/74,39,RcxX)
> exten => _123456,n,Verbose(2,Voicemail for Main)
> exten => _123456,n,Set(CALLERID(name)=)
> exten => _123456,n,VoiceMail(74,us)
> exten => _123456,n,Hangup
> include => fax_incoming
> include => michael_incoming
> include => internal_calls
>
> exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
> exten => h,n,Goto(noanswer,s,1)
> exten => h,n(done),NoOp()
>
> Unfortunately two E-Mails are sent anyway...
> This is the Asterisk log:
>
>     -- Executing [00493529123456 at michael_incoming:1]
> Verbose("PJSIP/pbxmichael_in-00000418", "2,Call for Main -
> [+493511111111]") in new stack
>   == Call for Main - [+493511111111]
>     -- Executing [00493529123456 at michael_incoming:2]
> Dial("PJSIP/pbxmichael_in-00000418", "local/123456 at main_incoming,,xX")
> in new stack
>     -- Called local/123456 at main_incoming
>     -- Executing [123456 at main_incoming:1]
> Verbose("Local/123456 at main_incoming-00000268;2", "2,Call for Main -
> [+493511111111]") in new stack
>   == Call for Main - [+493511111111]
>     -- Executing [123456 at main_incoming:2]
> Set("Local/123456 at main_incoming-00000268;2",
> "CALLERID(num)=03511111111") in new stack
>     -- Executing [123456 at main_incoming:3]
> Set("Local/123456 at main_incoming-00000268;2",
> "CHANNEL(musicclass)=default") in new stack
>     -- Executing [123456 at main_incoming:4]
> Dial("Local/123456 at main_incoming-00000268;2", "SIP/74,39,RcxX") in new
> stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/74
>     -- Local/123456 at main_incoming-00000268;1 is ringing
>     -- SIP/74-00000462 is ringing
>     -- Local/123456 at main_incoming-00000268;1 is ringing
>     -- SIP/74-00000462 is ringing
>     -- SIP/74-00000462 is ringing
>     -- SIP/74-00000462 is ringing
>   == Spawn extension (michael_incoming, 00493529123456, 2) exited
> non-zero on 'PJSIP/pbxmichael_in-00000418'
>     -- Executing [h at michael_incoming:1]
> GotoIf("PJSIP/pbxmichael_in-00000418", "0?done") in new stack
>     -- Executing [h at michael_incoming:2]
> Goto("PJSIP/pbxmichael_in-00000418", "noanswer,s,1") in new stack
>     -- Goto (noanswer,s,1)
>   == Spawn extension (main_incoming, 123456, 4) exited non-zero on
> 'Local/123456 at main_incoming-00000268;2'
>     -- Executing [h at main_incoming:1]
> GotoIf("Local/123456 at main_incoming-00000268;2", "0?done") in new stack
>     -- Executing [s at noanswer:1] NoOp("PJSIP/pbxmichael_in-00000418",
> "UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack
>     -- Executing [h at main_incoming:2]
> Goto("Local/123456 at main_incoming-00000268;2", "noanswer,s,1") in new stack
>     -- Goto (noanswer,s,1)
>     -- Executing [s at noanswer:2] System("PJSIP/pbxmichael_in-00000418",
> "echo "Verpasster Anruf vom +493511111111 um 19:13" | mail -s
> "Verpasster Anruf" info at mydomain.de") in new stack
>     -- Executing [s at noanswer:1]
> NoOp("Local/123456 at main_incoming-00000268;2", "UID CALL: 1636222382.6032
> / DATE: 20211106-191306)") in new stack
>     -- Executing [s at noanswer:2]
> System("Local/123456 at main_incoming-00000268;2", "echo "Verpasster Anruf
> vom 03511111111 um 19:13" | mail -s "Verpasster Anruf"
> info at mydomain.de") in new stack
>
> Any other idea?
>
> Thanks
> Luca Bertoncello
> (lucabert at lucabert.de)
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



-- 

Pozdrawiam,

Łukasz Grzywański
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20211106/88659139/attachment.html>