[Asterisk-code-review] configs: Updates to sample configs (asterisk[19])
N A
asteriskteam at digium.com
Mon Dec 13 07:32:19 CST 2021
N A has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/17660 )
Change subject: configs: Updates to sample configs
......................................................................
configs: Updates to sample configs
Includes some minor updates to extensions.conf
and iax.conf. In particular, the demonstration
of macros in extensions.conf is removed, as
Macro is deprecated and will be removed soon.
These examples have been replaced with examples
demonstrating the usage of Gosub instead.
The older exten => ...,n syntax is also mostly
replaced with the same keyword to demonstrate the
newer, more concise way of defining extensions.
IAXTEL no longer exists, so this example is replaced
with something more generic.
Some documentation is also added to extensions.conf
and iax.conf to clarify some of the new expanded
encryption capabilities with IAX2.
ASTERISK-29758 #close
Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46
---
M configs/samples/extensions.conf.sample
M configs/samples/iax.conf.sample
2 files changed, 142 insertions(+), 129 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/60/17660/1
diff --git a/configs/samples/extensions.conf.sample b/configs/samples/extensions.conf.sample
index 8d6c3e8..a371953 100644
--- a/configs/samples/extensions.conf.sample
+++ b/configs/samples/extensions.conf.sample
@@ -223,10 +223,10 @@
;
; so that dialtone remains even after dialing a 9. Please note that ignorepat
; only works with channels which receive dialtone from the PBX, such as DAHDI,
-; Phone, and VPB. Other channels, such as SIP and MGCP, which generate their
-; own dialtone and converse with the PBX only after a number is complete, are
-; generally unaffected by ignorepat (unless DISA or another method is used to
-; generate a dialtone after answering the channel).
+; Phone, and VPB. Other channels, such as IAX2, PJSIP, SIP and MGCP, which generate
+; generate their own dialtone and converse with the PBX only after a number is
+; complete, are generally unaffected by ignorepat (unless DISA or another method
+; is used to generate a dialtone after answering the channel).
;
;
@@ -241,12 +241,15 @@
;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
+;exten => _125642861XX,1,Dial(IAX2/user at example.com/${EXTEN:7})
+;exten => _125642862XX,1,Dial(IAX2/user:secret at example.com/${EXTEN:7})
+;exten => _125642863XX,1,Dial(IAX2/user:secret:[outkey]@example.com/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
-;exten => _12564286000,1,Dial(SIP/customer1)
+;exten => _12564286000,1,Dial(PJSIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
@@ -279,10 +282,10 @@
include => dundi-e164-local
include => dundi-e164-switch
;
-; DUNDi can also be implemented as a Macro instead of using
+; DUNDi can also be implemented as a subroutine instead of using
; the Local channel driver.
;
-[macro-dundi-e164]
+[dundi-e164]
;
; ARG1 is the extension to Dial
;
@@ -300,13 +303,10 @@
include => dundi-e164-lookup
;
-; Here are the entries you need to participate in the IAXTEL
-; call routing system. Most IAXTEL numbers begin with 1-700, but
-; there are exceptions. For more information, and to sign
-; up, please go to www.gnophone.com or www.iaxtel.com
+; Here is how to make a call to another Asterisk server using the IAX2 protocol
;
-[iaxtel700]
-exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
+[iax2-trunk]
+exten => _555XXXX,1,Dial(IAX2/username:secret at example.com/${EXTEN}@some-dialplan-context)
;
; The SWITCH statement permits a server to share the dialplan with
@@ -321,14 +321,15 @@
;
; International long distance through trunk
;
-exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
+exten => _9011.,1,Gosub(dundi-e164,s,1(${EXTEN:4}))
+; same => n,Hangup()
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
[trunkld]
;
; Long distance context accessed through trunk
;
-exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
+exten => _91NXXNXXXXXX,1,Gosub(dundi-e164,s,1(${EXTEN:1}))
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
@@ -421,36 +422,46 @@
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
-same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
-same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
+ same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
+ same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
-same => n,Set(TIMEOUT(absolute)=10800)
-same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
-same => n,GotoIf($["${isnresult}" != ""]?from)
-same => n,Set(DIALSTATUS=CONGESTION)
-same => n,Goto(fn-CONGESTION,1)
-same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
-same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
-same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
-same => n(dial),Dial(SIP/${isnresult},40)
-same => n,Goto(fn-${DIALSTATUS},1)
+ same => n,Set(TIMEOUT(absolute)=10800)
+ same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
+ same => n,GotoIf($["${isnresult}" != ""]?from)
+ same => n,Set(DIALSTATUS=CONGESTION)
+ same => n,Goto(fn-CONGESTION,1)
+ same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
+ same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
+ same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
+ same => n(dial),Dial(SIP/${isnresult},40)
+ same => n,Goto(fn-${DIALSTATUS},1)
exten => fn-BUSY,1,Busy()
exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()
-[macro-trunkdial]
+[trunkdial]
;
-; Standard trunk dial macro (hangs up on a dialstatus that should
+; Standard trunk dial subroutine (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
;
-exten => s,1,Dial(${ARG1})
+exten => s,1,StackPop()
+ same => n,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup()
-exten => s-BUSY,1,Hangup()
-exten => _s-.,1,NoOp
+exten => s-BUSY,1,PlayTones(busy)
+ same => n,Wait(60)
+ same => n,Hangup()
+exten => s-CHANUNAVAIL,1,PlayTones(congestion)
+ same => n,Wait(60)
+ same => n,Hangup()
+exten => s-CONGESTION,1,PlayTones(congestion)
+ same => n,Wait(60)
+ same => n,Hangup()
+exten => _s-.,1,NoOp(${DIALSTATUS})
+ same => n,Hangup()
[stdexten]
;
@@ -471,18 +482,18 @@
; previous value (before being declared as LOCAL()) upon Return.
;
exten => _X.,50000(stdexten),NoOp(Start stdexten)
-exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
-exten => _X.,n,Set(LOCAL(dev)=${ARG1})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
-exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
-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)
+ same => n,Set(LOCAL(ext)=${EXTEN})
+ same => n,Set(LOCAL(dev)=${ARG1})
+ same => n,Set(LOCAL(cntx)=${ARG2})
+ same => n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
+ same => n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
+ same => 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,n,Return() ; If they press #, return to start
+ same => n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,Return() ; If they press #, return to start
+ same => n,Return() ; If they press #, return to start
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
@@ -501,24 +512,24 @@
; See above note in stdexten about priority handling on exit.
;
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
-exten => _X.,n,Set(LOCAL(ext)=${ARG1})
-exten => _X.,n,Set(LOCAL(dev)=${ARG2})
-exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
-exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
+ same => n,Set(LOCAL(ext)=${ARG1})
+ same => n,Set(LOCAL(dev)=${ARG2})
+ same => n,Set(LOCAL(dontcntx)=${ARG3})
+ same => n,Set(LOCAL(tortcntx)=${ARG4})
+ same => n,Set(LOCAL(cntx)=${ARG5})
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
-exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
+ same => n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.screening)
-exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+ same => 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,n,NoOp(Finish stdPrivacyexten NOANSWER)
-exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
+ same => n,NoOp(Finish stdPrivacyexten NOANSWER)
+ same => n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
-exten => stdexten-BUSY,n,Return() ; If they press #, return to start
+ same => n,NoOp(Finish stdPrivacyexten BUSY)
+ same => n,Return() ; If they press #, return to start
exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
@@ -527,24 +538,22 @@
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()
+ same => n,Return()
-[macro-page]
+[sub-page]
;
-; Paging macro:
+; Paging subroutine:
;
; Check to see if SIP device is in use and DO NOT PAGE if they are
;
; ${ARG1} - Device to page
-
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
-exten => s,n,GotoIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
-exten => s,n(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(${ARG1})
-exten => s,n(fail),Hangup()
-
+ same => n,GotoIf($["${AVAILSTATUS}" = "1"]?autoanswer:fail)
+ same => n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
+ same => n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
+ same => n,NoOp() ; Add others here and Post on the Wiki!!!!
+ same => n,Dial(${ARG1})
+ same => n(fail),Hangup()
[demo]
include => stdexten
@@ -552,18 +561,18 @@
; 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,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.
+ same => n,Answer() ; Answer the line
+ same => n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
+ same => n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
+ same => n(restart),BackGround(demo-congrats) ; Play a congratulatory message
+ same => n(instruct),BackGround(demo-instruct) ; Play some instructions
+ same => 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)
exten => 3,1,Set(CHANNEL(language)=fr) ; Set language to french
-exten => 3,n,Goto(s,restart) ; Start with the congratulations
+ same => n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
@@ -572,19 +581,19 @@
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
-exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
-exten => 1234,n,Goto(default,s,1) ; exited Voicemail
+ same => n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
+ same => n,Goto(default,s,1) ; exited 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
+ same => 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.
+ same => n,Hangup() ; Hang them up.
;
; A timeout and "invalid extension rule"
@@ -597,64 +606,65 @@
; Asterisk demo.
;
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.
+ same => n,Dial(IAX2/guest at pbx.digium.com/s at default) ; Call the Asterisk demo
+ same => n,Playback(demo-nogo) ; Couldn't connect to the demo site
+ same => n,Goto(s,6) ; Return to the start over message.
;
; 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,Playback(demo-echodone) ; Let them know it's over
-exten => 600,n,Goto(s,6) ; Start over
+ same => n,Echo() ; Do the echo test
+ same => n,Playback(demo-echodone) ; Let them know it's over
+ same => n,Goto(s,6) ; Start over
;
-; You can use the Macro Page to intercom a individual user
-exten => 76245,1,Macro(page,SIP/Grandstream1)
+; You can use the page subroutine to intercom a individual user
+exten => 76245,1,Gosub(page,s,1(SIP/Grandstream1))
; or if your peernames are the same as extensions
-exten => _7XXX,1,Macro(page,SIP/${EXTEN})
+exten => _7XXX,1,Gosub(page,s,1(SIP/${EXTEN}))
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
-exten => 7999,2,Page(Local/Grandstream1 at page&Local/Xlite1 at page&Local/1234 at page/n,d)
+ same => n,Page(Local/Grandstream1 at page&Local/Xlite1 at page&Local/1234 at page/n,d)
; Give voicemail at extension 8500
;
exten => 8500,1,VoiceMailMain()
-exten => 8500,n,Goto(s,6)
+ same => n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
-;exten => 1265,n,Goto(s,5)
+; same => n,Goto(s,5)
;
-; The page context calls up the page macro that sets variables needed for auto-answer
+; The page context calls up the sub-page subroutine that sets variables needed for auto-answer
; It is in is own context to make calling it from the Page() application as simple as
; Local/{peername}@page
;
[page]
-exten => _X.,1,Macro(page,SIP/${EXTEN})
+exten => _X.,1,Gosub(sub-page,s,1(SIP/${EXTEN}))
+ same => n,Hangup()
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
-;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
-;exten => s,n,WaitExten
+; same => n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
+; same => n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
-;exten => s,n,Wait,2
-;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
-;exten => s,n,WaitExten
+; same => n,Wait,2
+; same => n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
+; same => n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
@@ -704,7 +714,7 @@
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
; Ditto for wil
-;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
+; same => n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)
;If you want to subscribe to the status of a parking space, this is
@@ -727,7 +737,7 @@
; voicemailmain
;
;exten => 8500,1,VoicemailMain
-;exten => 8500,n,Hangup
+; same => n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
@@ -743,17 +753,18 @@
; this is the context that your incoming IAX/SIP trunk dumps you in...
;[acme-incoming]
;exten => s,1,Wait(1)
-;exten => s,n,Answer()
-;exten => s,n(menu),Playback(acme/vm-brief-menu)
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
+; same => n,Answer() ; "same" allows you to continue an extension more
+; concisely without having to repeat the extension or pattern
+; same => n(menu),Playback(acme/vm-brief-menu)
+; same => n(exten),Background(vm-enter-num-to-call)
+; same => n,WaitExten(5)
+; same => n(goodbye),Playback(vm-goodbye)
+; same => n(end),Hangup()
;
;include => acme-extens
;
;exten => i,1,Playback(vm-invalid)
-;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
+; same => n,Goto(s,exten) ; optionally, transfer to operator
;
;exten => t,1,Goto(s,goodbye)
;
@@ -761,10 +772,10 @@
;
;[acme-internal]
;exten => s,1,Answer()
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
+; same => n(exten),Background(vm-enter-num-to-call)
+; same => n,WaitExten(5)
+; same => n(goodbye),Playback(vm-goodbye)
+; same => n(end),Hangup()
;
;include => trunkint
;include => trunkld
@@ -780,10 +791,10 @@
;[acme-extens]
;include => stdexten
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
-;exten => 111,n,Goto(s,exten)
+; same => n,Goto(s,exten)
;
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
-;exten => 112,n,Goto(s,end)
+; same => n,Goto(s,end)
;
; end of acme example
@@ -823,20 +834,20 @@
[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
+ same => n,Wait(0.25)
+ same => n,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages if there's no pause before the synchronizing beep.
-exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
-exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
-exten => _X.,n,SayPhonetic(z)
+ same => n,Set(FUTURETIME=$[${EPOCH} + 12])
+ same => n,SayUnixTime(${FUTURETIME},Zulu,HNS)
+ same => n,SayPhonetic(z)
; use the timezone associated with the extension (sip only), or system-wide
; default if one hasn't been set.
-exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
-exten => _X.,n,Playback(spy-local)
-exten => _X.,n,WaitUntil(${FUTURETIME})
-exten => _X.,n,Playback(beep)
-exten => _X.,n,Return()
+ same => n,SayUnixTime(${FUTURETIME},${timezone},HNS)
+ same => n,Playback(spy-local)
+ same => n,WaitUntil(${FUTURETIME})
+ same => n,Playback(beep)
+ same => n,Return()
;
; ANI context: use in the same way as "time" above
@@ -844,13 +855,13 @@
[ani]
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
-exten => _X.,n,Playback(vm-from)
-exten => _X.,n,SayDigits(${CALLERID(ani)})
-exten => _X.,n,Wait(1.25)
-exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
-exten => _X.,n,Return()
+ same => n,Wait(0.25)
+ same => n,Answer()
+ same => n,Playback(vm-from)
+ same => n,SayDigits(${CALLERID(ani)})
+ same => n,Wait(1.25)
+ same => n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
+ same => n,Return()
; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
diff --git a/configs/samples/iax.conf.sample b/configs/samples/iax.conf.sample
index cbce423..1d1c136 100644
--- a/configs/samples/iax.conf.sample
+++ b/configs/samples/iax.conf.sample
@@ -206,7 +206,8 @@
; across the net. "md5" uses a challenge/response md5 sum arrangement, but
; still requires both ends have plain text access to the secret. "rsa" allows
; unidirectional secret knowledge through public/private keys. There is no
-; default unless set here in the [general] section.
+; default unless set here in the [general] section. Only md5 and rsa support
+; media encryption.
;
;auth=md5
@@ -215,10 +216,10 @@
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
-; Enable IAX2 encryption. The default is no.
+; Enable IAX2 encryption. The default is no. This option does not force
+; encryption for calls, it merely allows it to be used for calls.
;
-;encryption=yes
-;
+encryption=yes
;
; Force encryption insures no connection is established unless both sides
@@ -542,7 +543,8 @@
; the ":" character. "outkey" is a single, private key to use to authenticate
; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub
; while private keys are named /var/lib/asterisk/keys/<name>.key. Private
-; keys should always be 3DES encrypted.
+; keys should always be 3DES encrypted. If encryption is used (applicable to
+; md5 and rsa only), a secret must be provided.
;
;
; NOTE: All hostnames and IP addresses in this file are for example purposes
--
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Gerrit-Project: asterisk
Gerrit-Branch: 19
Gerrit-Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46
Gerrit-Change-Number: 17660
Gerrit-PatchSet: 1
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