Great thanks
Steven Job wrote:
OK, enough of this...... No reason to bicker about something like
this.....
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dial&diff=57
For those of you that do not have a working web browser or cand find
it with Google here is the text.
Dial macros
Introduced in/for Asterisk 1.2, see bug/patch 2905
You can now add args to the macro by using a '^' char
Dial(Zap/1|60|M(mymacro^cat^dog^bark))
Also, the macro can set the MACRO_RESULT variable to do the following:
ABORT - Hangup both legs of the call
BUSY
CONTINUE - Hangup the called party and continue on in the dialplan
from where you called Dial
GOTO:<context>^<exten>^<priority> - Transfer the call.
Note: If you want the call to be bridged upon completion of the macro,
you should NOT set the MACRO_RESULT variable to anything. IF
MACRO_RESULT is not defined, the thread of execution falls off the end
of the macro and bridges the call. Setting it to CONTINUE causes the
call NOT to be bridged, and execution to resume at n+1 priority in the
calling context. Of course setting it to BUSY or GOTO has the implied
results (and of course the call is not bridged).
(I found this a bit confusing because my mental model expected some
kind of explicit case for "bridge the call", such as setting
MACRO_RESULT to "BRIDGE" or "CONNECT" or something like that. It's the
ABSENCE of any value that causes the call to be bridged.)
Example 2: Dial macro
screen-record: Please record your name press pound when finished.
screen-from: You have a call from
screen-accept: Press 1 to accept this call or any other key to reject.
exten => 890,1,Wait(0.2)
exten => 890,2,Playback(screen-record)
exten => 890,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten => 890,4,Record(${SCREEN_FILE}.gsm|6|25)
exten => 890,5,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
exten => 890,6,Voicemail([EMAIL PROTECTED])
[macro-screen]
exten => s,1,Wait(0.2)
exten => s,2,Playback(screen-from)
exten => s,3,Playback(${ARG1})
exten => s,4,Read(ACCEPT|screen-accept|1)
exten => s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten => s,6,SetVar(MACRO_RESULT=CONTINUE)
exten => s,7,System(/bin/rm ${ARG1})
Notes:
a.. Do not put spaces between the arguments to the Dial command, it
will not work.
b.. When options t, T", "h", "H", "w", "W" or "L" (with multiple
arguments) are applied, Asterisk will remain in the media path, even
if canreinvite=yes'' (a SIP channel option) has been specified.
Return codes
Dial sets DIALSTATUS to indicate its success. However, under some
circumstances, execution will jump to priority n+101 in the current
context. This happens when:
a.. All channels dialed were busy
b.. There is exists something at n+101 in the current context
c.. You are running asterisk 1.0.x, priorityjumping=yes is set in
extensions.conf, or the j option is specificed in the dial command
Note that in Asterisk v1.2+ priorityjumping is considered "off" by
default, which is a change from previous versions.
If the g option is specified, and the called party hangs up before the
calling party, then Dial continues execution at priority n+1.
Note: The bristuff patches change the dial behaviour slightly and jump
to n+201 if the dialed destination isn't connected (e.g. a SIP
softphone is not up and running, or hasn't registered).
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