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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Sep 2 08:15:18 CDT 2009


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:                tilghman
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Project:                    Asterisk
Issue ID:                   15815
Category:                   Core/Channels
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0.14 
Regression:                 Yes 
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:              2009-09-02 08:15 CDT
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Summary:                    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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---------------------------------------------------------------------- 
 (0110012) lmadsen (administrator) - 2009-09-02 08:15
 https://issues.asterisk.org/view.php?id=15815#c110012 
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Just verifying a couple of things:

* x.alaw does exist?
* it exists in /var/lib/asterisk/sounds/en/ ?

I seem to see the x.alaw trying to play -- I don't understand which part
isn't working. I presume the other LIMIT_* files you've setup are working,
but not that one in particular. It would probably be easier to see if you
used different file names.

Assigned to Tilghman for now to review. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-02 08:15 lmadsen        Note Added: 0110012                          
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