[asterisk-bugs] [Asterisk 0015815]: [patch][regression] LIMIT_TIMEOUT_FILE is not functional

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15815 
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Reported By:                adomjan
Assigned To:                jpeeler
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Project:                    Asterisk
Issue ID:                   15815
Category:                   Core/Channels
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.0.27
Asterisk Version:           SVN 
JIRA:                       SWP-764 
Regression:                 Yes 
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:             2009-09-02 07:42 CDT
Last Modified:              2010-03-23 16:17 CDT
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Summary:                    [patch][regression] LIMIT_TIMEOUT_FILE is not
functional
Description: 
I run similar functional bug in 1.6.0.9 but it was fixed there, but in
1.6.0.14 exists, however the fix in 1.6.0.9 is included in 1.6.0.14.

  -- Executing [01231 at sip:1] Set("SIP/11-b7f08060",
"LIMIT_PLAYAUDIO_CALLER=yes") in new stack
    -- Executing [01231 at sip:2] Set("SIP/11-b7f08060",
"LIMIT_PLAYAUDIO_CALLEE=no") in new stack
    -- Executing [01231 at sip:3] Set("SIP/11-b7f08060",
"LIMIT_TIMEOUT_FILE=x") in new stack
    -- Executing [01231 at sip:4] Set("SIP/11-b7f08060",
"LIMIT_CONNECT_FILE=x") in new stack
    -- Executing [01231 at sip:5] Set("SIP/11-b7f08060",
"LIMIT_WARNING_FILE=x") in new stack
    -- Executing [01231 at sip:6] Dial("SIP/11-b7f08060",
"SIP/sipteszt/1231,90,L(15000:5000)") in new stack
    -- Limit Data for this call:
       > timelimit      = 15000
       > play_warning   = 5000
       > play_to_caller = yes
       > play_to_callee = no
       > warning_freq   = 0
       > start_sound    = x
       > warning_sound  = x
       > end_sound      = x
  == Using SIP RTP CoS mark 5
    -- Called sipteszt/1231
    -- SIP/sipteszt-092627d0 answered SIP/11-b7f08060
    -- <SIP/11-b7f08060> Playing 'x.alaw' (language 'en')
    -- Packet2Packet bridging SIP/11-b7f08060 and SIP/sipteszt-092627d0
    -- Packet2Packet bridging SIP/11-b7f08060 and SIP/sipteszt-092627d0
// asterisk should play x file now, but it does not
    -- Packet2Packet bridging SIP/11-b7f08060 and SIP/sipteszt-092627d0
    -- <SIP/11-b7f08060> Playing 'x.alaw' (language 'en')
  == Spawn extension (sip, 01231, 6) exited non-zero on 'SIP/11-b7f08060'

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---------------------------------------------------------------------- 
 (0119768) svnbot (reporter) - 2010-03-23 16:17
 https://issues.asterisk.org/view.php?id=15815#c119768 
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Repository: asterisk
Revision: 254050

U   trunk/main/channel.c

------------------------------------------------------------------------
r254050 | jpeeler | 2010-03-23 16:17:24 -0500 (Tue, 23 Mar 2010) | 14
lines

Exit native bridging early for greater timing accuracy with warnings

This changes native bridging to break one millisecond early so that the
more
accurate timeval calculations done in the generic bridge can be performed
using
the bridge config. Currently the time between exiting native bridging
slightly
late can sometimes cause a large enough discrepancy for warnings to be
missed.
For the record, 1.4 does not attempt to native bridge at all when warnings
are
enabled.

(closes issue https://issues.asterisk.org/view.php?id=15815)
Reported by: adomjan

Review: https://reviewboard.asterisk.org/r/577/

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http://svn.digium.com/view/asterisk?view=rev&revision=254050 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-23 16:17 svnbot         Checkin                                      
2010-03-23 16:17 svnbot         Note Added: 0119768                          
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