[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
Mon Nov 22 14:50:47 CST 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:                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 14:50 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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---------------------------------------------------------------------- 
 (0129058) rmudgett (administrator) - 2010-11-22 14:50
 https://issues.asterisk.org/view.php?id=18129#c129058 
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No.  The one-way patch fixed another CW/CID issue as described by the
commit message.

The issue you found is possible on 1.4 and later.  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/replaced, SIP never stops sending those DTMF RTP packets.

I have a patch just about ready for testing once I resolve a related
CW/CID issue I discovered. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-22 14:50 rmudgett       Note Added: 0129058                          
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