[asterisk-commits] samples: Canonicalize app names in extensions.conf.sample. (asterisk[14])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Apr 7 11:27:32 CDT 2017
Anonymous Coward #1000019 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5416 )
Change subject: samples: Canonicalize app names in extensions.conf.sample.
......................................................................
samples: Canonicalize app names in extensions.conf.sample.
This takes care of warnings by ossobv/asterisklint.
Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93
---
M configs/samples/extensions.conf.sample
1 file changed, 22 insertions(+), 23 deletions(-)
Approvals:
George Joseph: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/configs/samples/extensions.conf.sample b/configs/samples/extensions.conf.sample
index 28319b5..fb71092 100644
--- a/configs/samples/extensions.conf.sample
+++ b/configs/samples/extensions.conf.sample
@@ -448,8 +448,8 @@
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
-exten => s-NOANSWER,1,Hangup
-exten => s-BUSY,1,Hangup
+exten => s-NOANSWER,1,Hangup()
+exten => s-BUSY,1,Hangup()
exten => _s-.,1,NoOp
[stdexten]
@@ -478,15 +478,15 @@
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
-exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
-exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
+exten => a,1,VoiceMailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()
[stdPrivacyexten]
@@ -512,11 +512,11 @@
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
-exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
@@ -526,10 +526,10 @@
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
-exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
-exten => a,n,Return
+exten => a,1,VoiceMailMain(${mbx}) ; If they press *, send the user into VoicemailMain
+exten => a,n,Return()
-[macro-page];
+[macro-page]
;
; Paging macro:
;
@@ -538,26 +538,25 @@
; ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
-exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
+exten => s,n,GotoIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
-exten => s,n(fail),Hangup
+exten => s,n(fail),Hangup()
[demo]
-include => stdexten
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait(1) ; Wait a second, just for fun
-exten => s,n,Answer ; Answer the line
+exten => s,n,Answer() ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
-exten => s,n,WaitExten ; Wait for an extension to be dialed.
+exten => s,n,WaitExten() ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
@@ -575,16 +574,16 @@
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
-exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
+exten => 1235,1,VoiceMail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
-exten => 1236,n,Voicemail(1234,b) ; Unless busy
+exten => 1236,n,VoiceMail(1234,b) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
-exten => #,n,Hangup ; Hang them up.
+exten => #,n,Hangup() ; Hang them up.
;
; A timeout and "invalid extension rule"
@@ -596,7 +595,7 @@
; Create an extension, 500, for dialing the
; Asterisk demo.
;
-exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
+exten => 500,1,Playback(demo-abouttotry) ; Let them know what's going on
exten => 500,n,Dial(IAX2/guest at pbx.digium.com/s at default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
@@ -605,7 +604,7 @@
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
-exten => 600,n,Echo ; Do the echo test
+exten => 600,n,Echo() ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
@@ -623,7 +622,7 @@
; Give voicemail at extension 8500
;
-exten => 8500,1,VoicemailMain
+exten => 8500,1,VoiceMailMain()
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
@@ -689,9 +688,9 @@
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
-;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
+;exten => 6245,n,VoiceMail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
-;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
+;exten => 6245,dial+101,VoiceMail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
--
To view, visit https://gerrit.asterisk.org/5416
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Walter Doekes <walter+asterisk at wjd.nu>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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