Here's your solution [vmtest]
exten => s,1,background(vm-Family,3) exten => s,n,waitexten(3) exten => s,n,Voicemail(${callnum}@default) exten => *,1,VoicemailMain(${callnum}@default) exten => #,1,VoicemailMain(${callnum}@default) exten => i,1,Voicemail(${callnum}@default) exten => t,1,Voicemail(${callnum}@default) vm-family plays when you come in. you have 3 seconds to hit * or #. If not, you go to regular voicemail. If so, you go to admin and get prompted for the password. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Thursday, April 07, 2011 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk login to voicemail Actually the mailbox is 7167435000... in this case the two variables are the same and the mailbox 7167435000 does exist.... I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if they push * during greeting, then i went them to prompted for a PIN for that mailbox (VoiceMailMain) I'm totally lost as to how to get this done... any suggestions? On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas <da...@debsinc.com> wrote: As I see it, callednum and vmbox should not be the same. Vmbox is a "good" mailbox you're going to reach if the user doesn't hit #, callednum is the "fallback" number that you are going to use and should be an established mailbox (3-4 digits) not a full number (10 digits you have indicated). _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Thursday, April 07, 2011 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk login to voicemail Ok, i have this now... [voicemail] exten => s,1,VoiceMail(${vmbox},su) exten => *,1,VoiceMailMain(${callednum}) in AGI i have: $AGI->set_variable("callednum", $options); $AGI->set_variable("vmbox", $options); $AGI->set_context("voicemail"); I'm getting a busy signal and this error.... pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-00000002' sent into invalid extension 'XXXXXXXXXX' in context 'voicemail', but no invalid handler -- _____________________________________________________________________ -- 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