[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
noreply at bugs.digium.com
Tue Nov 6 14:54:05 CST 2007
The following issue has been ASSIGNED.
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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: 11-06-2007 14:54 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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svnbot - 11-06-07 14:54
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Repository: asterisk
Revision: 89057
U trunk/main/channel.c
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r89057 | file | 2007-11-06 14:54:04 -0600 (Tue, 06 Nov 2007) | 4 lines
Remove native bridging check for DTMF based transfers. Thanks to the last
batch of RTP changes it is no longer required for the media stream to go
through Asterisk if DTMF is going over signalling. It will simply reinvite
back as needed.
(closes issue http://bugs.digium.com/view.php?id=11172)
Reported by: ibc
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Issue History
Date Modified Username Field Change
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11-06-07 14:54 svnbot Checkin
11-06-07 14:54 svnbot Note Added: 0073232
11-06-07 14:54 svnbot Status new => assigned
11-06-07 14:54 svnbot Assigned To => file
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