I just read:
Certain options to the Dial() statement require that Asterisk is in the
media path, and consequently Asterisk will not let go of it: /t/, ''T",
"h", "H", "w", "W" or "L" (with multiple arguments). Probably there are
more.
I had in my memory that "r", "R", "m" would also prevent a reinvite. Can
anybody say something on that? Below is a list of all options.
o *t*: Allow the /called/ user to transfer the call by hitting #
o *T*: Allow the /calling/ user to transfer the call by hitting #
o *r*: Generate a ringing tone for the calling party, passing
no audio from the called channel(s) until one answers. Use
with care and don't insert this by default into all your
dial statements as you are killing call progress information
for the user. Really, you almost certainly do not want to
use this. Asterisk will generate ring tones automatically
where it is appropriate to do so. "r" makes it go the next
step and additionally generate ring tones where it is
probably not appropriate to do so.
o *R*: Indicate ringing to the calling party when the called
party indicates ringing, pass no audio until answered. This
is available only if you are using kapejod's bristuff
<http://www.voip-info.org/wiki/index.php?page=Asterisk+zaphfc>.
o *m*: Provide Music on Hold to the calling party until the
called channel answers. This is mutually exclusive with
option 'r', obviously. Use m(class) to specify a class for
the music on hold.
o *n*: (Asterisk 1.1 and later) July 2005 bug 752
<http://bugs.digium.com/view.php?id=752> was included in CVS
(Asterisk 1.1) and enhances the privacy manager
considerably. As part of this patch, the 'n' flag to Dial
got changed to be used as part of the privacy features,
instead of being the 'dont jump to +101' flag. That flag is
now 'j'.
o *o*: Restore the Asterisk v1.0 CallerId behaviour (send the
original caller's ID) in Asterisk v1.2 (default: send this
extension's number)
o *j*: Asterisk 1.2 and later: Jump to priority n+101 if all
of the requested channels were busy (just like behaviour in
Asterisk 1.0.x)
o *M(*/x/*)*: Executes the macro (x) upon connect of the call
(i.e. when the called party answers)
o *h*: Allow the callee to hang up by dialing ***
o *H*: Allow the caller to hang up by dialing ***
o *C*: Reset the CDR (Call Detail Record) for this call. This
is like using the NoCDR
<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+NoCDR>
command
o *P(*/x/*)*: Use the PrivacyManager
<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager>,
using /x/ as the database (/x/ is optional)
o *g*: When the called party hangs up, exit to execute more
commands in the current context.
o *G(context^exten^pri)*: If the call is answered, transfer
both parties to the specified context and extension. The
calling party is transferred to priority x, and the called
party to priority x+1. This allows the dialplan to
distinguish between the calling and called legs of the call
(new in v1.2).
o *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party.
o *S(*/n/*)*: Hangup the call /n/ seconds AFTER called party
picks up.
o *d*: This flag trumps the 'H' flag and intercepts any dtmf
while waiting for the call to be answered and returns that
value on the spot. This allows you to dial a 1-digit exit
extension while waiting for the call to be answered - see
also RetryDial
<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+RetryDial>
o *D(*/digits/*)*: After the called party answers, send
/digits/ as a DTMF stream, then connect the call to the
originating channel.
o *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y'
ms are left, repeated every 'z' ms) Only 'x' is required,
'y' and 'z' are optional. The following special variables
are optional for limit calls: (pasted from app_dial.c)
+ *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play
sounds to the caller.
+ *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the
callee.
+ *LIMIT_TIMEOUT_FILE* - File to play when time is up.
+ *LIMIT_CONNECT_FILE* - File to play when call begins.
+ *LIMIT_WARNING_FILE* - File to play as warning if 'y'
is defined. If *LIMIT_WARNING_FILE* is not defined,
then the default behaviour is to announce ("You have
[XX minutes] YY seconds").
o *f*: forces callerid to be set as the extension of the line
making/redirecting the outgoing call. For example, some
PSTNs don't allow callerids from other extensions than the
ones that are assigned to you.
o *w*: Allow the /called/ user to start recording after
pressing *1 or what defined in features.conf
<http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf>
(Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)
o *W*: Allow the /calling/ user to start recording after
pressing *1 or what defined in features.conf
<http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf>
(Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)
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