[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:                     assigned
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:              
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Date Submitted:             2010-10-13 04:31 CDT
Last Modified:              2010-11-22 16:39 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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 (0129064) rmudgett (administrator) - 2010-11-22 16:39
 https://issues.asterisk.org/view.php?id=18129#c129064 
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The issue_18129_v1.8.patch fixes this issue.  It suppresses the DTMF begin
frame as well as the DTMF end frame.  It also fixes a couple cleanup issues
if you answer the call while it is doing the CID spill.

The new problem I found is the CW/CID spill is not sent if the call is
natively bridged to another analog port.  The analog port does not detect
the CW CAS DTMF 'D' response from the phone so the CID spill is not sent. 
DTMF detection is disabled when in a native bridge.  To fix the native
bridge issue is going to need more thought.  Simply enabling DTMF detection
results in doubled digits. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-11-22 16:39 rmudgett       Note Added: 0129064                          
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