[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:                lmadsen
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Project:                    Asterisk
Issue ID:                   15815
Category:                   Core/Channels
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                        
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:              2010-01-19 16:42 CST
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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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---------------------------------------------------------------------- 
 (0116953) lmadsen (administrator) - 2010-01-19 16:42
 https://issues.asterisk.org/view.php?id=15815#c116953 
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Aha! Yep, I was able to reproduce this today! Without the Packet2Packet
bridging, it worked fine. Below is the non-working version.

    -- Called 0004f2040002
    -- SIP/0004f2040002-00000007 is ringing
    -- SIP/0004f2040002-00000007 answered SIP/0004f2040001-00000006
    -- <SIP/0004f2040001-00000006> Playing 'beep.ulaw' (language 'en')
    -- Packet2Packet bridging SIP/0004f2040001-00000006 and
SIP/0004f2040002-00000007
    -- Packet2Packet bridging SIP/0004f2040001-00000006 and
SIP/0004f2040002-00000007

** BEEP SHOULD BE HERE **

    -- Packet2Packet bridging SIP/0004f2040001-00000006 and
SIP/0004f2040002-00000007
    -- <SIP/0004f2040001-00000006> Playing 'beep.ulaw' (language 'en')
  == Spawn extension (phones, 666, 7) exited non-zero on
'SIP/0004f2040001-00000006' 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-19 16:42 lmadsen        Note Added: 0116953                          
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