Re: [asterisk-users] Notifying missed calls
Am 06.11.2021 um 21:15 schrieb Łukasz Grzywański: Hi Łukasz, Dziękuję > two legs in this same context > ( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) ) > > PJSIP/pbxmichael_in-0418 > and > Local/123456@main_incoming-0268 > > [main_incoming] > exten => _+49X.,1,goto(${EXTEN:3},1) > exten => _0049X.,1,goto(${EXTEN:4},1) > exten => _03529X.,1,goto(${EXTEN:1},1) > exten => _3529X.,1,goto(${EXTEN:4},1) > > 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 You are my hero! It works as expected! Thank you very very much! Luca -- _ -- 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
Re: [asterisk-users] Notifying missed calls
Hi ;-) now I see 😂 Luca two legs in this same context ( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) ) PJSIP/pbxmichael_in-0418 and Local/123456@main_incoming-0268 [main_incoming] exten => _+49X.,1,goto(${EXTEN:3},1) exten => _0049X.,1,goto(${EXTEN:4},1) exten => _03529X.,1,goto(${EXTEN:1},1) exten => _3529X.,1,goto(${EXTEN:4},1) 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 LG Lukasz On Sat, 6 Nov 2021 at 21:02, Łukasz Grzywański wrote: > Hi, > strange > > -- Goto (noanswer,s,1) > -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418", > "echo "Verpasster Anruf vom +4935 um 19:13" | mail -s "Verpasster > Anruf" i...@mydomain.de") in new stack > -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-0268;2", > "UID CALL: 1636222382.6032/ DATE: 20211106-191306)") in new stack > -- Executing [s@noanswer:2] System("Local/123456@main_incoming-0268;2", > "echo "Verpasster Anrufvom 035 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...@.de) > exten => s,n,Hangup() > > > LG > Lukasz > > On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello > 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-0418", "2,Call for Main - >> [+4935]") in new stack >> == Call for Main - [+4935] >> -- Executing [00493529123456@michael_incoming:2] >> Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX") >> in new stack >> -- Called local/123456@main_incoming >> -- Executing [123456@main_incoming:1] >> Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main - >> [+4935]") in new stack >> == Call for Main - [+4935] >> -- Executing [123456@main_incoming:2] >> Set("Local/123456@main_incoming-0268;2", >> "CALLERID(num)=035") in new stack >> -- Executing [123456@main_incoming:3] >> Set("Local/123456@main_incoming-0268;2", >> "CHANNEL(musicclass)=default") in new stack >> -- Ex
Re: [asterisk-users] Notifying missed calls
Hi, strange -- Goto (noanswer,s,1) -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418", "echo "Verpasster Anruf vom +4935 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-0268;2", "echo "Verpasster Anrufvom 035 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...@.de) exten => s,n,Hangup() LG Lukasz On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello 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-0418", "2,Call for Main - > [+4935]") in new stack > == Call for Main - [+4935] > -- Executing [00493529123456@michael_incoming:2] > Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX") > in new stack > -- Called local/123456@main_incoming > -- Executing [123456@main_incoming:1] > Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main - > [+4935]") in new stack > == Call for Main - [+4935] > -- Executing [123456@main_incoming:2] > Set("Local/123456@main_incoming-0268;2", > "CALLERID(num)=035") in new stack > -- Executing [123456@main_incoming:3] > Set("Local/123456@main_incoming-0268;2", > "CHANNEL(musicclass)=default") in new stack > -- Executing [123456@main_incoming:4] > Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new > stack > == Using SIP RTP CoS mark 5 > -- Called SIP/74 > -- Local/123456@main_incoming-0268;1 is ringing > -- SIP/74-0462 is ringing > -- Local/123456@main_incoming-0268;1 is ringing > -- SIP/74-0462 is ringing > -- SIP/74-0462 is ringing > -- SIP/74-0462 is ringing > == Spawn extension (michael_incoming, 00493529123456, 2) exited > non-zero on 'PJSIP/pbxmichael_in-0418' > -- Executing [h@michael_incoming:1] > GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack > -- Executing [h@michael_incoming:2] > Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack > -- Goto (noanswer,s,1) > == Spawn extension (main_incoming, 123456, 4) exited non-zero on > 'Local/123456@main_incoming-0268;2' > -- Executing [h@main_incoming:1] > GotoIf("Local/123456@main_incoming-0268;2", "0?done") in new stack > -- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-0418", > "UID CALL: 1636222382.6030 / DATE:
Re: [asterisk-users] Notifying missed calls
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-0418", "2,Call for Main - [+4935]") in new stack == Call for Main - [+4935] -- Executing [00493529123456@michael_incoming:2] Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX") in new stack -- Called local/123456@main_incoming -- Executing [123456@main_incoming:1] Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main - [+4935]") in new stack == Call for Main - [+4935] -- Executing [123456@main_incoming:2] Set("Local/123456@main_incoming-0268;2", "CALLERID(num)=035") in new stack -- Executing [123456@main_incoming:3] Set("Local/123456@main_incoming-0268;2", "CHANNEL(musicclass)=default") in new stack -- Executing [123456@main_incoming:4] Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/74 -- Local/123456@main_incoming-0268;1 is ringing -- SIP/74-0462 is ringing -- Local/123456@main_incoming-0268;1 is ringing -- SIP/74-0462 is ringing -- SIP/74-0462 is ringing -- SIP/74-0462 is ringing == Spawn extension (michael_incoming, 00493529123456, 2) exited non-zero on 'PJSIP/pbxmichael_in-0418' -- Executing [h@michael_incoming:1] GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack -- Executing [h@michael_incoming:2] Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack -- Goto (noanswer,s,1) == Spawn extension (main_incoming, 123456, 4) exited non-zero on 'Local/123456@main_incoming-0268;2' -- Executing [h@main_incoming:1] GotoIf("Local/123456@main_incoming-0268;2", "0?done") in new stack -- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-0418", "UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack -- Executing [h@main_incoming:2] Goto("Local/123456@main_incoming-0268;2", "noanswer,s,1") in new stack -- Goto (noanswer,s,1) -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418", "echo "Verpasster Anruf vom +4935 um 19:13" | mail -s "Verpasster Anruf" i...@mydomain.de") in new stack -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-0268;2", "UID CALL: 1636222382.6032 / DATE: 20211106-191306)") in new stack -- Executing [s@noanswer:2] System("Local/123456@main_incoming-0268;2", "echo "Verpasster Anruf vom 035 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
Re: [asterisk-users] Notifying missed calls
On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote: > Really, I can't understand what you mean... I'm feeling really > dumb... No need to feel dumb. I'm not an expert and when I look to my extensions.conf... well... countless pulling my hairs out, head banging on the keyboard,,, :-) 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. Take a look to the example I posted. It's very basic, but it does the job. -- _ -- 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
Re: [asterisk-users] Notifying missed calls
Am 06.11.2021 um 14:43 schrieb Frank Vanoni: Hi Frank > On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > >> 1) The E-Mails will be sent "double" > > It sends the first mail by executing "noanswer,2" and a second mail > because because of "main-incoming,h,2" Really, I can't understand what you mean... I'm feeling really dumb... >> 2) The E-Mails will be sent for outgoing unanswered calls, too. > > Use the "h" extension only in the context for incoming calls I have just one "h" extension: [main_incoming] exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done) exten => h,n,Goto(noanswer,s,1) exten => h,n(done),NoOp() Could you explain what you mean? 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
Re: [asterisk-users] Notifying missed calls
On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > 1) The E-Mails will be sent "double" It sends the first mail by executing "noanswer,2" and a second mail because because of "main-incoming,h,2" > 2) The E-Mails will be sent for outgoing unanswered calls, too. Use the "h" extension only in the context for incoming calls > -- _ -- 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
Re: [asterisk-users] Notifying missed calls
Here my configuration: [incoming] ; Incoming from Swisscom exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)}) same => n,Dial(SIP/deskphone,120) same => n,Hangup() exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done) exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s "Missed Call from ${CALLERID(num)}" my-em...@address.here) exten => h,n(done),NoOp() -- _ -- 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