asterisk users - Apr 2006 - canreinvite, bandwidth, dial option

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)