[asterisk-bugs] [DAHDI-linux 0016644]: On call-waiting events, dropped audio on first call then switch to second call when first call ends without 'flash'

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16644 
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Reported By:                kirkawolff
Assigned To:                
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Project:                    DAHDI-linux
Issue ID:                   16644
Category:                   General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
JIRA:                        
Reviewboard Link:            
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Date Submitted:             2010-01-18 19:07 CST
Last Modified:              2010-06-08 03:35 CDT
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Summary:                    On call-waiting events, dropped audio on first call
then switch to second call when first call ends without 'flash'
Description: 
I'm running asterisk 1.6.2 on ubuntu karmic.  I was running a asterisk on
ubuntu jaunty, and converted configuration from jaunty to karmic.  I have a
Wildcard TDM400P REV I (according to DAHDI_hardware) with three modules,
fxs, fxo, and fxs.  All calls are between DAHDI FXS channel 1 and SIP.  The
fxs channel 1 is configured in /etc/dahdi/system.conf as fxoks and uses the
mg2 echo canceler.

When a phone call is taking place (an incoming call is answered from
DAHDI/1), and a second call comes in.  The callerid tone is sent, and the
person on DAHDI/1 can continue to hear the person on the other end (the SIP
call), but they cannot hear the person on the DAHDI/1 line.  When the first
caller hangs up, the DAHDI/1 line is immediately switched to the new
incoming phone call without any 'flash' event taking place.  The only
information related to DAHDI in the asterisk debug output is the
following:

VERBOSE[12345] chan_dahdi.c:     -- CPE supports Call Waiting Caller*ID. 
Sending 'FIRST LAST/9515554444'

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 (0123095) alecdavis (manager) - 2010-06-08 03:35
 https://issues.asterisk.org/view.php?id=16644#c123095 
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I am experiencing same here with
Asterisk SVN-branch-1.6.2-r268819M
DAHDI SVN-branch-2.2-r8552

initial call between External SIP to internal DAHDI TDM800P FXS port.
2nd call, (call waiting) from PSTN TDM800P FXO port -> FXS port.

After callwaiting tone, symptoms are the same, no audio FXS -> SIP, but
audio SIP -> FXS.

Then SIP end hangs up, and somehow FXS port is then connected to PSTN call
on FXO port (automatically?) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-06-08 03:35 alecdavis      Note Added: 0123095                          
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