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Summary:ASTERISK-13446: Sound overlapping if Read() called again very soon
Reporter:dmartin (dmartin)Labels:
Date Opened:2009-01-25 13:09:56.000-0600Date Closed:2011-06-07 14:03:16
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_read
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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.

****** ADDITIONAL INFORMATION ******

temporary open for the cause: sip:290@sip.davidm.es
Comments:By: dmartin (dmartin) 2009-01-25 13:15:28.000-0600

Output from the Asterisk console while testing condition:

== Using SIP RTP CoS mark 5
 == Using UDPTL CoS mark 5
   -- Executing [290@users:1] Answer("SIP/201-103a7420", "") in new stack
   -- Executing [290@users:2] Read("SIP/201-103a7420", "exta,/pool/asterisk/menu,1,s,2,0") in new stack
   -- Accepting a maximum of 1 digits.
   -- <SIP/201-103a7420> Playing '/pool/asterisk/menu.alaw' (language 'en')
   -- User entered '2'
   -- Executing [290@users:3] Read("SIP/201-103a7420", "extb,/pool/asterisk/menu,1,s,2,0") in new stack
   -- Accepting a maximum of 1 digits.
   -- <SIP/201-103a7420> Playing '/pool/asterisk/menu.alaw' (language 'en')
   -- User disconnected
By: Joshua C. Colp (jcolp) 2009-02-11 12:35:57.000-0600

I've tried numerous combinations with the latest trunk to try to reproduce this and failed, unless it is indeed happening and I just don't realize it. Can you attach console output with debug enabled (debug set to console in logger.conf and core set debug 2) so I can see when the DTMF is being received? Plus an actual recording of this happening would be great, and even a better explanation of what you mean.
By: dmartin (dmartin) 2009-02-12 12:58:44.000-0600

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@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@users:1] Answer("SIP/201-103acce8", "") in new stack
   -- Executing [290@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@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>
By: Joshua C. Colp (jcolp) 2009-02-12 13:03:39.000-0600

A Monitor recording would be extremely useful.
By: Joshua C. Colp (jcolp) 2009-02-12 13:05:13.000-0600

As for trying to listen via the SIP URI provided that actually yields a 480 Temporarily unavailable.
By: dmartin (dmartin) 2009-02-12 13:22:30.000-0600

I was recording and uploading the files. Sorry about the inconvenience with the SIP uri, I can't test it right now through Internet.

http://sip.davidm.es/out-full-message.wav  - This is the message played once, without pushing any number

http://sip.davidm.es/out-glitch.wav  - This is when I push a number.

This is the extension I used for the test:

290 => {
Monitor(wav,/tmp/out,m);
Answer();
Read(exta,/pool/asterisk/menu,1,s,2,0);
Read(extb,/pool/asterisk/menu,1,s,2,0);
Hangup();
};
By: Joshua C. Colp (jcolp) 2009-02-12 14:08:03.000-0600

Very interesting... can you attach the output of "module show" and the startup output plus the output of "timing test"?
By: Mark Michelson (mmichelson) 2009-02-12 14:18:44.000-0600

I think I know what's up here. Try using a revision of trunk after rev 173354. I think my fix on issue ASTERISK-13495 will fix this, too.
By: dmartin (dmartin) 2009-02-12 14:44:57.000-0600

Just compiled latest revision (r175255) and this problem is gone. Maybe the issue that you mention caused the problem with Read().

Fixed for me. Thank you!
By: Mark Michelson (mmichelson) 2009-02-12 14:46:35.000-0600

Yep, I did some tests with the revision you reported and the latest trunk and got the same results. Thanks for reporting back so quickly!
By: Joshua C. Colp (jcolp) 2009-02-12 14:46:39.000-0600

Closed per reporter, fixed by the wonderful putnopvut as a result of 14384.