[asterisk-bugs] [Asterisk 0011172]: Allow direct RTP in "Dial" with "tT" options if phones are "canreinvite=yes" and "dtmfmode=info"

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11172 
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Reported By:                ibc
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11172
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.13  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             11-06-2007 13:04 CST
Last Modified:              11-20-2007 08:12 CST
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Summary:                    Allow direct RTP in "Dial" with "tT" options if
phones are "canreinvite=yes" and "dtmfmode=info"
Description: 
In case two phones have "canreinvite=yes" and "dtmfmode=info" there is no
reason Asterisk to be in the media path when "Dial" with "t" or "T"
option.

"dtmfmode=info" means DTMF in SIP INFO messages, so Asterisk doesn't need
to be in the media path to get phones DTMF for native transfer.

But unfortunatelly Asterisk remains in the media path in the case above.
Could it be possible to consider this in "channels.c"?
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 ibc - 11-20-07 08:12  
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Im' sorry, I selected "reopen" because I went to report a bug about this,
but it was a failure of mine in phone configuration. It works in fact.
Please, close again the bug. Sorry. 

Issue History 
Date Modified   Username       Field                    Change               
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11-20-07 08:12  ibc            Note Added: 0074069                          
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