[asterisk-bugs] [Asterisk 0018129]: [patch] Oneway audio from SIP phone to FXS port after FXS port gets a CallWaiting pip

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18129 
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Reported By:                alecdavis
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   18129
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
JIRA:                       SWP-2367 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/978/ 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 290865 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-10-13 04:31 CDT
Last Modified:              2010-11-24 16:42 CST
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Summary:                    [patch] Oneway audio from SIP phone to FXS port
after FXS port gets a CallWaiting pip
Description: 
Internal SIP phone initiates a call with FXS port on TDM800P.
FXS connected phone has to have FSK CIDCW 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. 

External call comes in on FXO port, and attempts to ring FXS ports.
Call waiting beep is heard at FXS port, also no outbound audio to SIP
phone.

Then FXS port hook flashes, the FXO port is then connected to FXS as
expected.
But SIP device should hear MOH, but has dead air.

If the FXS port hook flashes to the SIP device, the FXO call then does
hear MOH.
Hook flash again back to FXO call, SIP hears nothing.  
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 (0129120) svnbot (reporter) - 2010-11-24 16:42
 https://issues.asterisk.org/view.php?id=18129#c129120 
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Repository: asterisk
Revision: 296166

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_dahdi.c

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r296166 | rmudgett | 2010-11-24 16:42:46 -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 https://issues.asterisk.org/view.php?id=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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http://svn.digium.com/view/asterisk?view=rev&revision=296166 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-24 16:42 svnbot         Checkin                                      
2010-11-24 16:42 svnbot         Note Added: 0129120                          
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