[asterisk-bugs] [Asterisk 0015642]: [patch] Fix for Sonus DTMF issues

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 10 18:09:29 CST 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15642 
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Reported By:                jasonshugart
Assigned To:                twilson
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Project:                    Asterisk
Issue ID:                   15642
Category:                   Core/RTP
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Target Version:             1.6.1.13
Asterisk Version:           SVN 
JIRA:                       SWP-406 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-08-03 12:28 CDT
Last Modified:              2009-12-10 18:09 CST
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Summary:                    [patch] Fix for Sonus DTMF issues
Description: 
In some cases when Asterisk sends DTMF to Sonus platforms as RFC2833, Sonus
will not recognize the DTMF.  Several articles have identified the problem
as being the gap in the audio prior to the DTMF packets being sent.  This
patch sends a single G.711 ulaw packet prior to the rfc2833 packets.  In
our testing across with two carriers (Level3 and 360 Networks) this patch
fixed our DTMF issues.  The rtp.c file seems very similar for the 1.4
branch, so minor changes could also be applied there.  I added an option to
the rtp.conf file to enable this fix, called rtpfixdtmf.  Attached are both
the rtp.c fix, and the rtp.conf change.
====================================================================== 

---------------------------------------------------------------------- 
 (0115131) twilson (administrator) - 2009-12-10 18:09
 https://issues.asterisk.org/view.php?id=15642#c115131 
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Is there a way that someone with a connection to a sonus switch could
somehow direct some traffic to one of my test boxes?  I would like to get
this issue fixed, but I am having trouble recreating the "audio gap" with
normal sip phones.  I would really like to come up with a good solution to
this issue, but can't in good conscience modify and commit a patch for an
issue that I don't entirely understand. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-10 18:09 twilson        Note Added: 0115131                          
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