[asterisk-bugs] [Asterisk 0016652]: [patch] SIP CHANNEL(rtpqos, audio, ...) variables missing.
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Feb 17 00:25:18 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16652
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Reported By: kkm
Assigned To:
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Project: Asterisk
Issue ID: 16652
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: ready for testing
Target Version: 1.6.0.24
Asterisk Version: SVN
JIRA: SWP-771
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 240324
Request Review:
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Date Submitted: 2010-01-19 14:19 CST
Last Modified: 2010-02-17 00:25 CST
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Summary: [patch] SIP CHANNEL(rtpqos,audio,...) variables
missing.
Description:
CHANNEL(rtpqos,audio,x) causes the following warnings for some x:
[Jan 14 21:56:19] WARNING[16860] chan_sip.c: Unrecognized argument
'rtpqos,audio,local_maxjitter' to CHANNEL
[Jan 14 21:56:19] WARNING[16860] func_channel.c: Unknown or unavailable
item requested: 'rtpqos,audio,local_maxjitter'
Affected variables (not exhaustive list)
local_maxjitter
local_normdevjitter
remote_maxjitter
remote_normdevjitter
maxrtt
minrtt
normdevrtt
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----------------------------------------------------------------------
(0118139) svnbot (reporter) - 2010-02-17 00:25
https://issues.asterisk.org/view.php?id=16652#c118139
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Repository: asterisk
Revision: 247124
U trunk/channels/Makefile
U trunk/channels/chan_sip.c
A trunk/channels/sip/dialplan_functions.c
U trunk/channels/sip/include/config_parser.h
A trunk/channels/sip/include/dialog.h
A trunk/channels/sip/include/dialplan_functions.h
A trunk/channels/sip/include/globals.h
U trunk/channels/sip/include/sip_utils.h
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r247124 | tilghman | 2010-02-17 00:25:16 -0600 (Wed, 17 Feb 2010) | 13
lines
Make all of the various rtpqos parameters in this branch available from
the CHANNEL function.
Also includes a test for retrieving rtpqos parameters, including a NULL
RTP
driver. Additionally, some further separation of the SIP internal API
into
headers was necessary.
(closes issue https://issues.asterisk.org/view.php?id=16652)
Reported by: kkm
Patches:
20100204__issue16652.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/501/
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http://svn.digium.com/view/asterisk?view=rev&revision=247124
Issue History
Date Modified Username Field Change
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2010-02-17 00:25 svnbot Note Added: 0118139
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