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

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Oct 19 05:08:00 CDT 2010


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:                
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Project:                    Asterisk
Issue ID:                   18129
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-2367 
Regression:                 No 
Reviewboard Link:            
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:              2010-10-19 05:08 CDT
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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.

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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---------------------------------------------------------------------- 
 (0128187) alecdavis (manager) - 2010-10-19 05:08
 https://issues.asterisk.org/view.php?id=18129#c128187 
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Uploaded bug18129.diff.txt

The issue is as the callwaiting spill to sent to the FXS port, the DSP is
triggers on a DTMF.

Then the RTP engine sends DTMF packets continuously to the SIP device,
hence the continuous tone.

This patch disabled the DTMF detection, when the callwait is started.

This can be simulated by
establish a call from FXS port -> SIP phone
then from CLI
  console dial 89 at phones 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-19 05:08 alecdavis      Note Added: 0128187                          
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