[Asterisk-Users] Call Rollover

[Ryan R. Fligg]

Have a question about implementing Call Rollover with my current extensions.conf configuration.   [macro-stdexten] exten => s,1,Dial(${ARG2},20)           ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1})       ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1)          ; If they press #, return to start exten => s,102,Voicemail2(b${ARG1})     ; If busy, send to voicemail w/ busy announce exten => s,103,Goto(default,s,1)        ; If they press #, return to start   [mainmenu] exten => s,1,Zapateller exten => s,2,Wait,3 exten => s,3,Answer exten => s,4,DigitTimeout,5 exten => s,5,ResponeTimeout,10 exten => s,6,Background(/var/lib/asterisk/sounds/vm/100/Work/MainMenu/SDSgreet) exten => 1,1,Goto(salesmenu,s,1) exten => 2,1,Goto(technicalmenu,s,1) exten => 3,1,Directory,default exten => 4,1,Goto,s|6 exten => 5,1,Goto(remoteline,s,1) exten => i,1,Playback(invalid) exten => t,1,Goto(s,7) exten => s,7,Background(/var/lib/asterisk/sounds/vm/100/Work/MainMenu/goodbye) exten => s,8,Hangup include => SDSextensions include => departmentextensions   [salesmenu] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Goto(SDSextensions,202,1) exten => #,1,Goto(mainmenu,s,4)   [technicalmenu] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Goto(SDSextensions,204,1) exten => s,4,Goto(SDSextensions,206,1) exten => s,5,Goto(SDSextensions,207,1) exten => #,1,Goto(mainmenu,s,4)   [SDSextensions] include => macro-stdexten exten => 201,1,Macro(stdexten,${KHVM},${KH}) exten => 202,1,Macro(stdexten,${BOVM},${BO}) exten => 203,1,Macro(stdexten,${GOVM},${GO}) exten => 204,1,Macro(stdexten,${DWVM},${DW}) exten => 205,1,Macro(stdexten,${RFVM},${RF}) exten => 206,1,Macro(stdexten,${BRVM},${BR}) exten => 207,1,Macro(stdexten,${CGVM},${CG}) exten => 208,1,Macro(stdexten,${AGVM},${AG}) exten => 209,1,Macro(stdexten,${MDVM},${MD}) exten => 300,1,Macro(stdexten,${TRVM},${TR})   Conventions used in my Macro stdexten lines: InitialsVM = SIP address initials = extension   Example for user John Doe: JD=SIP/JohnDoe JDVM=400     My issue is that I want any calls coming into my [technicalmenu] context to rollover after a period of time to the next extension, 206, and then to 207.  If none of these extensions pickup I want the call to route back to the initital extensions voicemail.  I am using the stdexten macro to implement the busy and unavailable voicemail automation.  I am using the Goto statements to goto the [SDSextensions] context and use the stdexten macro.  I want to know if there is a way with my current implementation to enter the technical menu -> follow the first extension to my [SDSextensions] context -> execute the stdexten macro and then have that macro timeout without going to voicemail and then execute the other extensions until the last extension is executed and then return to the first extensions voicemail.     System Information: Asterisk CVS version: 12/11/2003-12:22:41   System: Dell Dimension 8300 Pentium 4 2.8GHz 512 Mb memory 120 Gb hard drive   3 X100P Cards 1 DSL Line 3 POTS   6 Snom200 Phones   Thanx in advance, this list has been very helpful, keep it up!   Ryan R. Fligg   Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell:     (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED]