[asterisk-bugs] [Asterisk 0016652]: [patch] SIP CHANNEL(rtpqos, audio, ...) variables missing.

[Asterisk Bug Tracker]

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
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:              2010-02-06 22:32 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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 (0117813) kkm (reporter) - 2010-02-06 22:32
 https://issues.asterisk.org/view.php?id=16652#c117813 
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Sorry, I still could not test that. The machine where I can test it is
currently under production load; also, I am not sure I am finding time for
that. My question is, can the change go into the trunk without me testing
it? Looks straightforward and clean to me. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-06 22:32 kkm            Note Added: 0117813                          
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