[asterisk-bugs] [Asterisk 0014815]: DTMF Appears to be broken from certain sources on asterisk 1.4.24 - double digit.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14815 
====================================================================== 
Reported By:                geoff2010
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14815
Category:                   Core/RTP
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Target Version:             1.4.25
Asterisk Version:           1.4.24 
Regression:                 Yes 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-04-02 15:38 CDT
Last Modified:              2009-04-28 16:45 CDT
====================================================================== 
Summary:                    DTMF Appears to be broken from certain sources on
asterisk 1.4.24 - double digit.
Description: 
I am having some trouble with 1.4.24.  This appears to be a regression
since 1.4.21.1 as I have just recently upgraded and started to experience
the problem.

When calling my asterisk box from a cell phone the system detects double
DTMF digits.  When calling from a landline or pure SIP device DTMF
detection works fine.  I have determined that the cell phone is sending the
RTP DTMF payload about 2x more than the other devices, but as far as I can
tell it should still be considered a single digit as per the RFC.

Attached are two TXT files of pcap exports which show the dump for a
working DTMF interaction, and one for a failure situation.

Please let me know if I can provide any further information.

Thanks,
Geoff
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014460 Asterisk plays a continuous tone foreve...
====================================================================== 

---------------------------------------------------------------------- 
 (0103897) geoff2010 (reporter) - 2009-04-28 16:45
 http://bugs.digium.com/view.php?id=14815#c103897 
---------------------------------------------------------------------- 
ah that makes perfect sense.  The RTP stream is totally within spec though,
because voice timestamps need to continue to increment even thought the ts
for the RFC2833 packet needs to remain static.

I see what you are saying now, and that definitely explains why my hack of
always setting rtp->dtmfcount=dtmftimeout fixes the issue... 

Now that you see what the issue is, can you suggest a real fix as opposed
by my "lets just try things to see if they work" approach? :)

thanks. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-28 16:45 geoff2010      Note Added: 0103897                          
======================================================================