[asterisk-bugs] [Asterisk 0014328]: Sound overlapping if Read() called again very soon

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14328 
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Reported By:                dmartin
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14328
Category:                   Applications/app_read
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 170980 
Request Review:              
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Date Submitted:             2009-01-25 13:09 CST
Last Modified:              2009-02-12 12:58 CST
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Summary:                    Sound overlapping if Read() called again very soon
Description: 
Having this dummy extension scenario:

290 => {
	Answer();
	Read(exta,/pool/asterisk/menu,1,s,2,0);
	Read(extb,/pool/asterisk/menu,1,s,2,0);
	Hangup();
};

If you dial a number before the first message played at all, the second
one starts overlapping the audio of the firsrt one.

I've tested this in another (useful) extension, and tried lots of values
for parameters, always with the same result.
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---------------------------------------------------------------------- 
 (0100023) dmartin (reporter) - 2009-02-12 12:58
 http://bugs.digium.com/view.php?id=14328#c100023 
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Hello,

This is the output you requested.

I'll try to explain this:

1. I call the extension and the message starts to play normally
2. In the middle of the message, I push a number (any)
3. The next message starts playing. At the same time, the previous message
that didn't ended when I push the number, is still playing.
4. Both messages are playing at the same time with glitches. When the
first ends, the second continues without glitches until it ends.

Can I help you recording the thing directly from the phone, or using
Monitor() ?

I remind you that this extension is available at sip:290 at sip.davidm.es (no
need to register).

I've tested with:

-Linksys PAP2
-Siphon
-Twinkle
-XTen Lite
-XMeeting

Always with same results.



frankie*CLI> core set debug 2
Core debug was 0 and is now 2
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [290 at users:1] Answer("SIP/201-103acce8", "") in new
stack
    -- Executing [290 at users:2] Read("SIP/201-103acce8",
"exta,/pool/asterisk/menu,1,s,2,0") in new stack
    -- Accepting a maximum of 1 digits.
    -- <SIP/201-103acce8> Playing '/pool/asterisk/menu.alaw' (language
'en')
    -- User entered '2'
    -- Executing [290 at users:3] Read("SIP/201-103acce8",
"extb,/pool/asterisk/menu,1,s,2,0") in new stack
    -- Accepting a maximum of 1 digits.
    -- <SIP/201-103acce8> Playing '/pool/asterisk/menu.alaw' (language
'en')
    -- User disconnected
frankie*CLI> 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-12 12:58 dmartin        Note Added: 0100023                          
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