Call is not forwarded to voicemail in below dial plan, why?
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
-- Called SIP/4
-- SIP/4-00000288 is ringing
-- Nobody picked up in 25000 ms
-- Executing [4@extensions:2] GotoIf("IAX2/home_server-6364", "0?line2") in
new stack
-- Executing [4@extensions:3] Dial("IAX2/home_server-6364",
"SIP/54,20,trw") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/54
-- SIP/54-00000289 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-6364'
-- Hungup 'IAX2/home_server-6364'
Extension 4 is not BUSY (just nobody pickup the call) so why isn't call going
to "Voicemail" it shouldn't ring FD_L2 (SIP/54)
Why isn't it going to "Voicemail"?
--
Thelma
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
