Merged revisions 296166 via svnmerge from

Merged revisions 296166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

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  r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines

  Merged revisions 296165 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

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    r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines

    Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.

    The FXS connected phone has to have CW/CID support to fail, as it will
    send back a DTMF 'A' or 'D' when it's ready to receive CallerID.  A normal
    phone with no CID never fails.  Also the SIP phone does not hear MOH when
    the CW call is answered.

    The DTMF end frame is suppressed when the phone acknowledges the CW signal
    for CID.  The problem is the DTMF begin frame needs to be suppressed as
    well.  The DTMF begin frame is causing SIP to start sending the DTMF RTP
    frames.  Since the DTMF end frame is suppressed, SIP will not stop sending
    those DTMF RTP packets.

    * Suppress the DTMF begin and end frames when the channel driver is
    looking for DTMF digits.

    * Fixed a couple issues caused by not cleaning up the CID spill if you
    answer the CW call while it is sending the CID spill.

    * Fixed not sending CW/CID spill to the phone when the call is natively
    bridged.  (Fixed by not using native bridge if CW/CID is possible.)

    * Suppress received audio when sending CW/CID spills.  The other parties
    involved do not need to hear the CW/CID spills and may be confused if the
    CW call is for them.

    (closes issue #18129)
    Reported by: alecdavis
    Patches:
          issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
    Tested by: alecdavis, rmudgett

    NOTE:

    * v1.4 does not have the main problem fixed by suppressing the DTMF start
    frames.  The other three items fixed are relevant.

    * If you really must restore native bridging between analog ports, you
    need to disable CW/CID either by configuring chan_dahdi.conf
    callwaitingcallerid=no or dialing *70 before dialing the number to
    temporarily disable CW.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296167 65c4cc65-6c06-0410-ace0-fbb531ad65f3