diff --git a/apps/app_dial.c b/apps/app_dial.c
index 855945b244..dbce62dcf8 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -143,11 +143,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
-
- Many SIP and ISDN phones cannot send DTMF digits until the call is
- connected. If you wish to use this option with these phones, you
- can use the Answer application before dialing.
-
+ NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
+ connected. If you wish to use this option with these phones, you
+ can use the Answer application before dialing.
When the caller hangs up, transfer the called party
to the specified destination and start execution at that location.
-
- Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').
-
+ NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+ prefixed with one or two underbars ('_').
When the caller hangs up, transfer the called party to the next priority of the current extension
and start execution at that location.
-
- Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').
-
-
- Using this option from a Macro() or GoSub() might not make sense as there would be no return points.
-
+ NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+ prefixed with one or two underbars ('_').
+ NOTE: Using this option from a Macro() or GoSub() might not make sense as there would be no return points.
Proceed with dialplan execution at the next priority in the current extension if the
@@ -205,9 +197,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
If the call is answered, transfer the calling party to
the specified priority and the called party to the specified
priority plus one.
-
- You cannot use any additional action post answer options in conjunction with this option.
-
+ NOTE: You cannot use any additional action post answer options in conjunction with this option.
Allow the called party to hang up by sending the DTMF sequence
@@ -216,12 +206,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
Allow the calling party to hang up by sending the DTMF sequence
defined for disconnect in features.conf .
-
- Many SIP and ISDN phones cannot send DTMF digits until the call is
- connected. If you wish to allow DTMF disconnect before the dialed
- party answers with these phones, you can use the Answer
- application before dialing.
-
+ NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
+ connected. If you wish to allow DTMF disconnect before the dialed
+ party answers with these phones, you can use the Answer
+ application before dialing.
Asterisk will ignore any forwarding requests it may receive on this dial attempt.
@@ -317,14 +305,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
- You cannot use any additional action post answer options in conjunction
- with this option. Also, pbx services are run on the peer (called) channel,
- so you will not be able to set timeouts via the TIMEOUT() function in this macro.
-
- Be aware of the limitations that macros have, specifically with regards to use of
+ NOTE: You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this macro.
+ WARNING: Be aware of the limitations that macros have, specifically with regards to use of
the WaitExten application. For more information, see the documentation for
- Macro() . Macro() .
@@ -389,10 +375,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
to send no cause. See the causes.h file for the
full list of valid causes and names.
-
- chan_sip does not support setting the cause on a CANCEL to anything
- other than ANSWERED_ELSEWHERE.
-
+ NOTE: chan_sip does not support setting the cause on a CANCEL to anything
+ other than ANSWERED_ELSEWHERE.
Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
@@ -427,7 +411,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
- Name of the subroutine to execute via Gosub
+ Name of the subroutine context to execute via Gosub .
+ The subroutine execution starts in the named context at the s exten and priority 1.
Arguments for the Gosub routine
@@ -456,11 +441,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
- You cannot use any additional action post answer options in conjunction
- with this option. Also, pbx services are run on the peer (called) channel,
- so you will not be able to set timeouts via the TIMEOUT() function in this routine.
-
+ NOTE: You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are run on the called channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this routine.
@@ -545,27 +528,29 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
[default]
- exten => callee_channel,1,NoOp()
+ exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
same => n,Return()
- exten => called_channel,1,NoOp()
+ exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
same => n,Return()
exten => _X.,1,NoOp()
- same => n,Dial(PJSIP/alice,,b(default^called_channel^1)B(default^callee_channel^1))
+ same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
same => n,Hangup()
- [default]
+ [my_gosub_routine]
- exten => called_channel,1,NoOp()
+ exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
same => n,Playback(hello)
same => n,Return()
+ [default]
+
exten => _X.,1,NoOp()
- same => n,Dial(PJSIP/alice,,U(default^called_channel^1))
+ same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
same => n,Hangup()
diff --git a/apps/app_queue.c b/apps/app_queue.c
index eb85c51b0e..641debffdf 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -152,21 +152,15 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
When the caller hangs up, transfer the called member
to the specified destination and start execution at that location.
-
- Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').
-
+ NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+ prefixed with one or two underbars ('_').
When the caller hangs up, transfer the called member to the next priority of
the current extension and start execution at that location.
-
- Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').
-
-
- Using this option from a Macro() or GoSub() might not make sense as there would be no return points.
-
+ NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+ prefixed with one or two underbars ('_').
+ NOTE: Using this option from a Macro() or GoSub() might not make sense as there would be no return points.
Allow callee to hang up by pressing * .
@@ -241,7 +235,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
Will run a macro on the called party's channel (the queue member) once the parties are connected.
- Will run a gosub on the called party's channel (the queue member) once the parties are connected.
+ Will run a gosub on the called party's channel (the queue member)
+ once the parties are connected. The subroutine execution starts in the
+ named context at the s exten and priority 1.
Will cause the queue's defaultrule to be overridden by the rule specified.