Hi, strange.... -- Goto (noanswer,s,1) -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-00000418", "echo "Verpasster Anruf vom +493511111111 um 19:13" | mail -s "Verpasster Anruf" i...@mydomain.de") in new stack -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-00000268;2", "UID CALL: 1636222382.6032/ DATE: 20211106-191306)") in new stack -- Executing [s@noanswer:2] System("Local/123456@main_incoming-00000268;2", "echo "Verpasster Anrufvom 03511111111 um 19:13" | mail -s "Verpasster Anruf" i...@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" i...@xxxx.de) exten => s,n,Hangup() LG Lukasz On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello <lucab...@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 - [${CALLERID(num)}]) > exten => _00493529123456,n,Dial(local/123456@main_incoming,,xX) > exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}]) > exten => _03529123456,n,Dial(local/123456@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@michael_incoming:1] > Verbose("PJSIP/pbxmichael_in-00000418", "2,Call for Main - > [+493511111111]") in new stack > == Call for Main - [+493511111111] > -- Executing [00493529123456@michael_incoming:2] > Dial("PJSIP/pbxmichael_in-00000418", "local/123456@main_incoming,,xX") > in new stack > -- Called local/123456@main_incoming > -- Executing [123456@main_incoming:1] > Verbose("Local/123456@main_incoming-00000268;2", "2,Call for Main - > [+493511111111]") in new stack > == Call for Main - [+493511111111] > -- Executing [123456@main_incoming:2] > Set("Local/123456@main_incoming-00000268;2", > "CALLERID(num)=03511111111") in new stack > -- Executing [123456@main_incoming:3] > Set("Local/123456@main_incoming-00000268;2", > "CHANNEL(musicclass)=default") in new stack > -- Executing [123456@main_incoming:4] > Dial("Local/123456@main_incoming-00000268;2", "SIP/74,39,RcxX") in new > stack > == Using SIP RTP CoS mark 5 > -- Called SIP/74 > -- Local/123456@main_incoming-00000268;1 is ringing > -- SIP/74-00000462 is ringing > -- Local/123456@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@michael_incoming:1] > GotoIf("PJSIP/pbxmichael_in-00000418", "0?done") in new stack > -- Executing [h@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@main_incoming-00000268;2' > -- Executing [h@main_incoming:1] > GotoIf("Local/123456@main_incoming-00000268;2", "0?done") in new stack > -- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-00000418", > "UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack > -- Executing [h@main_incoming:2] > Goto("Local/123456@main_incoming-00000268;2", "noanswer,s,1") in new stack > -- Goto (noanswer,s,1) > -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-00000418", > "echo "Verpasster Anruf vom +493511111111 um 19:13" | mail -s > "Verpasster Anruf" i...@mydomain.de") in new stack > -- Executing [s@noanswer:1] > NoOp("Local/123456@main_incoming-00000268;2", "UID CALL: 1636222382.6032 > / DATE: 20211106-191306)") in new stack > -- Executing [s@noanswer:2] > System("Local/123456@main_incoming-00000268;2", "echo "Verpasster Anruf > vom 03511111111 um 19:13" | mail -s "Verpasster Anruf" > i...@mydomain.de") in new stack > > Any other idea? > > Thanks > Luca Bertoncello > (lucab...@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
-- _____________________________________________________________________ -- 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