Brian, You need to put an include => default in your incoming context.
some samples here http://www.codepipe.com/id25.htm Regards Dave -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Mulligan Sent: 02 March 2004 17:58 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie Voicemenu question Hi I can get the Voicemenu stuff working OK but am unable to switch to a context where the incoming PSTN caller is able to enter SIP number (after promting) and have this forwarded to a proxy.What I am trying to do is give a PSTN caller the choice between voicemail, local extension or remote SIP user. Below is an extract from my extensions.conf, clearly this does not work. Any hint would be most appreciated. Thanks Brian [incoming] exten => s,1,Answer exten => s,2,Background(brian-ivr) exten => 6,1,Voicemail,u5152 exten => 7,1,Goto,pstn-to-sip|s|1 exten => 8,1,Dial,Zap/2 ; [pstn-to-sip] exten =>s,1,Background(pstn-to-sip); ask user to enter sip number exten =>9,_9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) exten =>8,_8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) exten =>7,_7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) _______________________________________________ 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
