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    ccdc417a
    Merged revisions 296167 via svnmerge from · ccdc417a
    Richard Mudgett authored
    https://origsvn.digium.com/svn/asterisk/branches/1.8
    
    ................
      r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
      
      Merged revisions 296166 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.6.2
      
      ................
        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
        
        ........
          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.
        ........
      ................
    ................
    
    
    git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
    ccdc417a
    History
    Merged revisions 296167 via svnmerge from
    Richard Mudgett authored
    https://origsvn.digium.com/svn/asterisk/branches/1.8
    
    ................
      r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
      
      Merged revisions 296166 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.6.2
      
      ................
        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
        
        ........
          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.
        ........
      ................
    ................
    
    
    git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3