The right conf must be like this: exten => 2015,1,Dial(SIP/[EMAIL PROTECTED],20,T,t) exten => 2015,2,Voicemail(u${EXTEN}) exten => 2015,102,Voicemail(b${EXTEN}) exten => 2015,103,Hangupv
Chris HARIGA Techselesta Inc. http://www.techselesta.com/ ----- Original Message ----- From: "Chris Clifton" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 10:28 PM Subject: [Asterisk-Users] gs on phone ? > > I have a GS101 connected to * with sip and g729. > > When an incoming call comes in from outside (via pstn for example), and no > one picks up the GS, * reports that 'the user is on the phone'. If no one > answers, I'd expect it to report 'unavailable'. > > Maybe I'm not understanding the call flow ... (should it be u$ at '2', then > b$ at '102' ?) My current config for call flow seems to match others I've > seen on the wiki, etc. > > my extensions.conf for the grandstream at x2015 - > > [incoming] > exten => 2015,1,Dial(SIP/[EMAIL PROTECTED],20,T,t) > exten => 2015,2,Voicemail(b${EXTEN}) > exten => 2015,3,Hangup > exten => 2015,102,Voicemail(u${EXTEN}) > exten => 2015,103,Hangup > > Thanks, > Chris Clifton > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users