Also, a more "fancy" approach [macro-dialvm]
exten => s,1,NoOp("${ExTEN}|${MACRO_EXTEN}|${ARG1}") exten => s,n,Dial(SIP/${ARG1},25,t) exten => s,n,NoOp(${ARG1}) exten => s,n,NoOp(${DIALSTATUS}) exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?BUSY) exten => s,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?NOANSWER) exten => s,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?CHANUNAVAIL) exten => s,n,VoiceMail(${ARG1},a) exten => s,n,MacroExit() exten => s,n(BUSY),Set(CDR(userfield)=DIAL-BUSY) exten => s,n,NoO${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,ResetCDR(w) exten => s,n,VoiceMail(${ARG1},b) exten => s,n,MacroExit() exten => s,n(NOANSWER),Set(CDR(userfield)=DIAL-NOANSWER) exten => s,n,NoOp(${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,ResetCDR(w) exten => s,n,VoiceMail(${ARG1},u) exten => s,n,MacroExit() exten => s,n(CHANUNAVAIL),Set(CDR(userfield)=DIAL-UNAVIL) exten => s,n,NoOp(${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,ResetCDR(w) exten => s,n,VoiceMail(${ARG1},uj) exten => s,n,MacroExit() exten => s,BUSY+101,Set(CDR(userfield)=DIAL-BSY-NOMBX) exten => s,n,NoOp(${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,ResetCDR(w) exten => s,n,NoOp("Mailbox Not found") exten => s,n,Goto(CHANUNAVAIL+101) exten => s,NOANSWER+101,Set(CDR(userfield)=DIAL-NA-NOMBX) exten => s,n,NoOp(${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,NoOp("MailBox Not found") exten => s,n,Goto(CHANUNAVAIL+101) exten => s,CHANUNAVAIL+101,Playback(/home/asterisk/gen/themailbox) exten => s,n,NoOp(${MACRO_EXTEN}) exten => s,n,NoOp(${ARG1}) exten => s,n,SayDigits(${MACRO_EXTEN}) exten => s,n,Playback(/home/asterisk/gen/doesnotexist) exten => s,n,MacroExit() -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Wednesday, December 29, 2010 11:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] call is not going to Voicemail with "1,n" I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten => 1,103,Voicemail(11,b) exten => 1,104,Hangup() exten => 1,n,Voicemail(11,b) ; Right to voicemail exten => 1,n,Hangup() Here is the transcript: -- Executing [...@office-open:1] Playback("SIP/pstn-5665-000000be", "transfer") in new stack -- <SIP/pstn-5665-000000be> Playing 'transfer' (language 'en') -- Executing [...@office-open:2] Dial("SIP/pstn-5665-000000be", "SIP/11&IAX2/iaxy-322|20|jrw") in new stack -- Called 11 -- Called iaxy-322 -- Call accepted by 10.0.0.108 (format ulaw) -- Format for call is ulaw -- IAX2/iaxy-322-8406 is busy -- Hungup 'IAX2/iaxy-322-8406' -- SIP/11-000000bf is ringing -- Nobody picked up in 20000 ms == Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is 'NOANSWER' However, if I number the dial plan in the old fashion way and don't answer the phone it goes to voicemail just fine: exten => 1,1,Playback(transfer) exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten => 1,103,Voicemail(11,b) exten => 1,104,Hangup() exten => 1,3,Voicemail(11,b) ; Right to voicemail exten => 1,4,Hangup() -- Joseph -- _____________________________________________________________________ -- 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