[asterisk-bugs] [Asterisk 0016714]: [patch] [regression] DTMF Relaying appers broken

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16714 
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Reported By:                greenfieldtech
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16714
Category:                   General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.21 
JIRA:                       SWP-818 
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:             2010-01-27 05:11 CST
Last Modified:              2010-02-04 02:30 CST
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Summary:                    [patch] [regression] DTMF Relaying appers broken
Description: 
When accepting calls from a SIP provider (VoxBone) and then terminating to
another provider (any SIP termination), DTMF signals will not pass
utilizing RFC2833 configuration.


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 (0117696) wdoekes (reporter) - 2010-02-04 02:30
 https://issues.asterisk.org/view.php?id=16714#c117696 
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ast14271-dtmf-filtered-and-selected.pcap.bz2 is a pcap of RTP traffic
between 10.0.0.35 (a phone) and 10.0.0.3 (asterisk 1.4.27.1) and between
10.0.0.3 and 95.215.204.68 (IVR on destination 1.4.24.1).

It contains the RTP DTMF events along with surrounding RTP voice.

(I had to trim it to the selected RTP portion only, as the PHP-upload
didn't like my 6MB file. You may need to decode-as-rtp some UDP packets
when using wireshark.)
(I don't have time to plough through the FreePBX sip and extensions
configs right now, but they should be pretty standard. Console logs are
unavailable but I saw nothing unusual.) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-04 02:30 wdoekes        Note Added: 0117696                          
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