Merged revisions 311295 via svnmerge from

Merged revisions 311295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines

  Merged revision 310986 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

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    r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines

    Dial() o option broke when connected line feature added.

    The patch restores the o option behavior and adds the ability to specify
    the CallerID.  The Dial o and f options are complementary to each other.
    The o option stores the CallerID on the outgoing channel as the channel's
    CallerID.  The f option forces the CallerID sent by the outgoing channel.

    o(x) - The argument 'x' is optional.  If not present, then specify that
    the CallerID that was present on the *calling* channel be stored as the
    CallerID on the *called* channel.  This was the behavior of Asterisk 1.0
    and earlier.  If present, then specify the CallerID stored on the *called*
    channel.  Note that o(${CALLERID(all)}) is similar to option o without
    parameters.

    f(x) - The argument 'x' is optional and its presence changes the behavior
    of this option.  If not present, then force the outgoing CallerID on a
    call-forward or deflection to the dialplan extension for this Dial() using
    a dialplan 'hint'.  For example, some PSTNs do not allow CallerID to be
    set to anything other than the numbers assigned to you.  If present, then
    force the outgoing CallerID to 'x'.

    Patches:
  	jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett

    JIRA ABE-2752
    JIRA SWP-3096
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311296 65c4cc65-6c06-0410-ace0-fbb531ad65f3